ASoC: msm: Add Audio drivers for MSM targets

Add snapshot for audio drivers for MSM targets. The code is
migrated from msm-3.18 kernel at the below commit/AU level -

AU_LINUX_ANDROID_LA.HB.1.3.1.06.00.00.187.056
(e70ad0cd5efdd9dc91a77dcdac31d6132e1315c1)
(Promotion of kernel.lnx.3.18-151201.)

Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>

52 files changed, 17030 insertions, 0 deletions

diff --git a/include/linux/msm_audio_ion.h b/include/linux/msm_audio_ion.h
new file mode 100644
index 000000000000..1dda6fd32dad
--- /dev/null
+++ b/include/linux/msm_audio_ion.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef _LINUX_MSM_AUDIO_ION_H
+#define _LINUX_MSM_AUDIO_ION_H
+#include <sound/q6asm-v2.h>
+#include <sound/pcm.h>
+#include <linux/msm_ion.h>
+
+
+int msm_audio_ion_alloc(const char *name, struct ion_client **client,
+			struct ion_handle **handle, size_t bufsz,
+			ion_phys_addr_t *paddr, size_t *pa_len, void **vaddr);
+
+int msm_audio_ion_import(const char *name, struct ion_client **client,
+			struct ion_handle **handle, int fd,
+			unsigned long *ionflag, size_t bufsz,
+			ion_phys_addr_t *paddr, size_t *pa_len, void **vaddr);
+int msm_audio_ion_free(struct ion_client *client, struct ion_handle *handle);
+int msm_audio_ion_mmap(struct audio_buffer *substream,
+		       struct vm_area_struct *vma);
+
+bool msm_audio_ion_is_smmu_available(void);
+int msm_audio_ion_cache_operations(struct audio_buffer *abuff, int cache_op);
+
+struct ion_client *msm_audio_ion_client_create(const char *name);
+void msm_audio_ion_client_destroy(struct ion_client *client);
+int msm_audio_ion_import_legacy(const char *name, struct ion_client *client,
+			struct ion_handle **handle, int fd,
+			unsigned long *ionflag, size_t bufsz,
+			ion_phys_addr_t *paddr, size_t *pa_len, void **vaddr);
+int msm_audio_ion_free_legacy(struct ion_client *client,
+			struct ion_handle *handle);
+u32 populate_upper_32_bits(ion_phys_addr_t pa);
+#endif /* _LINUX_MSM_AUDIO_ION_H */
diff --git a/include/linux/qdsp6v2/apr.h b/include/linux/qdsp6v2/apr.h
new file mode 100644
index 000000000000..f73f2e1eb5b3
--- /dev/null
+++ b/include/linux/qdsp6v2/apr.h
@@ -0,0 +1,179 @@
+/* Copyright (c) 2010-2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef __APR_H_
+#define __APR_H_
+
+#include <linux/mutex.h>
+#include <soc/qcom/subsystem_notif.h>
+
+enum apr_subsys_state {
+	APR_SUBSYS_DOWN,
+	APR_SUBSYS_UP,
+	APR_SUBSYS_LOADED,
+};
+
+struct apr_q6 {
+	void *pil;
+	atomic_t q6_state;
+	atomic_t modem_state;
+	struct mutex lock;
+};
+
+struct apr_hdr {
+	uint16_t hdr_field;
+	uint16_t pkt_size;
+	uint8_t src_svc;
+	uint8_t src_domain;
+	uint16_t src_port;
+	uint8_t dest_svc;
+	uint8_t dest_domain;
+	uint16_t dest_port;
+	uint32_t token;
+	uint32_t opcode;
+};
+
+#define APR_HDR_LEN(hdr_len) ((hdr_len)/4)
+#define APR_PKT_SIZE(hdr_len, payload_len) ((hdr_len) + (payload_len))
+#define APR_HDR_FIELD(msg_type, hdr_len, ver)\
+	(((msg_type & 0x3) << 8) | ((hdr_len & 0xF) << 4) | (ver & 0xF))
+
+#define APR_HDR_SIZE sizeof(struct apr_hdr)
+
+/* Version */
+#define APR_PKT_VER		0x0
+
+/* Command and Response Types */
+#define APR_MSG_TYPE_EVENT	0x0
+#define APR_MSG_TYPE_CMD_RSP	0x1
+#define APR_MSG_TYPE_SEQ_CMD	0x2
+#define APR_MSG_TYPE_NSEQ_CMD	0x3
+#define APR_MSG_TYPE_MAX	0x04
+
+/* APR Basic Response Message */
+#define APR_BASIC_RSP_RESULT 0x000110E8
+#define APR_RSP_ACCEPTED     0x000100BE
+
+/* Domain IDs */
+#define APR_DOMAIN_SIM	0x1
+#define APR_DOMAIN_PC		0x2
+#define APR_DOMAIN_MODEM	0x3
+#define APR_DOMAIN_ADSP	0x4
+#define APR_DOMAIN_APPS	0x5
+#define APR_DOMAIN_MAX	0x6
+
+/* ADSP service IDs */
+#define APR_SVC_TEST_CLIENT     0x2
+#define APR_SVC_ADSP_CORE	0x3
+#define APR_SVC_AFE		0x4
+#define APR_SVC_VSM		0x5
+#define APR_SVC_VPM		0x6
+#define APR_SVC_ASM		0x7
+#define APR_SVC_ADM		0x8
+#define APR_SVC_ADSP_MVM	0x09
+#define APR_SVC_ADSP_CVS	0x0A
+#define APR_SVC_ADSP_CVP	0x0B
+#define APR_SVC_USM		0x0C
+#define APR_SVC_LSM		0x0D
+#define APR_SVC_VIDC		0x16
+#define APR_SVC_MAX		0x17
+
+/* Modem Service IDs */
+#define APR_SVC_MVS		0x3
+#define APR_SVC_MVM		0x4
+#define APR_SVC_CVS		0x5
+#define APR_SVC_CVP		0x6
+#define APR_SVC_SRD		0x7
+
+/* APR Port IDs */
+#define APR_MAX_PORTS		0x80
+
+#define APR_NAME_MAX		0x40
+
+#define RESET_EVENTS		0x000130D7
+
+#define LPASS_RESTART_EVENT	0x1000
+#define LPASS_RESTART_READY	0x1001
+
+struct apr_client_data {
+	uint16_t reset_event;
+	uint16_t reset_proc;
+	uint16_t payload_size;
+	uint16_t hdr_len;
+	uint16_t msg_type;
+	uint16_t src;
+	uint16_t dest_svc;
+	uint16_t src_port;
+	uint16_t dest_port;
+	uint32_t token;
+	uint32_t opcode;
+	void *payload;
+};
+
+typedef int32_t (*apr_fn)(struct apr_client_data *data, void *priv);
+
+struct apr_svc {
+	uint16_t id;
+	uint16_t dest_id;
+	uint16_t client_id;
+	uint16_t dest_domain;
+	uint8_t rvd;
+	uint8_t port_cnt;
+	uint8_t svc_cnt;
+	uint8_t need_reset;
+	apr_fn port_fn[APR_MAX_PORTS];
+	void *port_priv[APR_MAX_PORTS];
+	apr_fn fn;
+	void *priv;
+	struct mutex m_lock;
+	spinlock_t w_lock;
+};
+
+struct apr_client {
+	uint8_t id;
+	uint8_t svc_cnt;
+	uint8_t rvd;
+	struct mutex m_lock;
+	struct apr_svc_ch_dev *handle;
+	struct apr_svc svc[APR_SVC_MAX];
+};
+
+int apr_load_adsp_image(void);
+struct apr_client *apr_get_client(int dest_id, int client_id);
+int apr_wait_for_device_up(int dest_id);
+int apr_get_svc(const char *svc_name, int dest_id, int *client_id,
+		int *svc_idx, int *svc_id);
+void apr_cb_func(void *buf, int len, void *priv);
+struct apr_svc *apr_register(char *dest, char *svc_name, apr_fn svc_fn,
+					uint32_t src_port, void *priv);
+inline int apr_fill_hdr(void *handle, uint32_t *buf, uint16_t src_port,
+			uint16_t msg_type, uint16_t dest_port,
+			uint32_t token, uint32_t opcode, uint16_t len);
+
+int apr_send_pkt(void *handle, uint32_t *buf);
+int apr_deregister(void *handle);
+void subsys_notif_register(struct notifier_block *mod_notif,
+				struct notifier_block *lp_notif);
+int apr_get_dest_id(char *dest);
+uint16_t apr_get_data_src(struct apr_hdr *hdr);
+void change_q6_state(int state);
+void q6audio_dsp_not_responding(void);
+void apr_reset(void *handle);
+enum apr_subsys_state apr_get_modem_state(void);
+void apr_set_modem_state(enum apr_subsys_state state);
+enum apr_subsys_state apr_get_q6_state(void);
+int apr_set_q6_state(enum apr_subsys_state state);
+void apr_set_subsys_state(void);
+const char *apr_get_lpass_subsys_name(void);
+bool apr_register_voice_svc(void);
+uint16_t apr_get_reset_domain(uint16_t proc);
+#endif
diff --git a/include/linux/qdsp6v2/apr_tal.h b/include/linux/qdsp6v2/apr_tal.h
new file mode 100644
index 000000000000..dd6442462e3a
--- /dev/null
+++ b/include/linux/qdsp6v2/apr_tal.h
@@ -0,0 +1,55 @@
+/* Copyright (c) 2010-2011, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef __APR_TAL_H_
+#define __APR_TAL_H_
+
+#include <linux/kernel.h>
+#include <linux/kthread.h>
+#include <linux/uaccess.h>
+
+/* APR Client IDs */
+#define APR_CLIENT_AUDIO	0x0
+#define APR_CLIENT_VOICE	0x1
+#define APR_CLIENT_MAX	0x2
+
+#define APR_DL_SMD    0
+#define APR_DL_MAX    1
+
+#define APR_DEST_MODEM 0
+#define APR_DEST_QDSP6 1
+#define APR_DEST_MAX   2
+
+#define APR_MAX_BUF   8192
+
+#define APR_OPEN_TIMEOUT_MS 5000
+
+typedef void (*apr_svc_cb_fn)(void *buf, int len, void *priv);
+struct apr_svc_ch_dev *apr_tal_open(uint32_t svc, uint32_t dest,
+			uint32_t dl, apr_svc_cb_fn func, void *priv);
+int apr_tal_write(struct apr_svc_ch_dev *apr_ch, void *data, int len);
+int apr_tal_close(struct apr_svc_ch_dev *apr_ch);
+struct apr_svc_ch_dev {
+	struct smd_channel *ch;
+	spinlock_t         lock;
+	spinlock_t         w_lock;
+	struct mutex       m_lock;
+	apr_svc_cb_fn      func;
+	char               data[APR_MAX_BUF];
+	wait_queue_head_t  wait;
+	void               *priv;
+	uint32_t           smd_state;
+	wait_queue_head_t  dest;
+	uint32_t           dest_state;
+};
+
+#endif
diff --git a/include/linux/qdsp6v2/apr_us.h b/include/linux/qdsp6v2/apr_us.h
new file mode 100644
index 000000000000..9a6804a4d634
--- /dev/null
+++ b/include/linux/qdsp6v2/apr_us.h
@@ -0,0 +1,193 @@
+/* Copyright (c) 2011-2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef __APR_US_H__
+#define __APR_US_H__
+
+#include <linux/qdsp6v2/apr.h>
+
+/* ======================================================================= */
+/*  Session Level commands */
+
+#define USM_SESSION_CMD_RUN				0x00012306
+struct usm_stream_cmd_run {
+	struct apr_hdr hdr;
+	u32            flags;
+	u32            msw_ts;
+	u32            lsw_ts;
+} __packed;
+
+/* Stream level commands */
+#define USM_STREAM_CMD_OPEN_READ			0x00012309
+struct usm_stream_cmd_open_read {
+	struct apr_hdr hdr;
+	u32            uMode;
+	u32            src_endpoint;
+	u32            pre_proc_top;
+	u32            format;
+} __packed;
+
+#define USM_STREAM_CMD_OPEN_WRITE			0x00011271
+struct usm_stream_cmd_open_write {
+	struct apr_hdr hdr;
+	u32            format;
+} __packed;
+
+
+#define USM_STREAM_CMD_CLOSE				0x0001230A
+
+#define USM_STREAM_CMD_SET_PARAM			0x00012731
+struct usm_stream_cmd_set_param {
+	struct apr_hdr hdr;
+	u32            buf_addr_lsw;
+	u32            buf_addr_msw;
+	u32            mem_map_handle;
+	u32            buf_size;
+	u32            module_id;
+	u32            param_id;
+} __packed;
+
+#define USM_STREAM_CMD_GET_PARAM			0x00012732
+struct usm_stream_cmd_get_param {
+	struct apr_hdr hdr;
+	u32            buf_addr_lsw;
+	u32            buf_addr_msw;
+	u32            mem_map_handle;
+	u32            buf_size;
+	u32            module_id;
+	u32            param_id;
+} __packed;
+
+/* Encoder configuration definitions */
+#define USM_STREAM_CMD_SET_ENC_PARAM			0x0001230B
+/* Decoder configuration definitions */
+#define USM_DATA_CMD_MEDIA_FORMAT_UPDATE		0x00011272
+
+/* Encoder/decoder configuration block */
+#define USM_PARAM_ID_ENCDEC_ENC_CFG_BLK			0x0001230D
+
+/* Max number of static located ports (bytes) */
+#define USM_MAX_PORT_NUMBER 8
+
+/* Max number of static located transparent data (bytes) */
+#define USM_MAX_CFG_DATA_SIZE 100
+
+/* Parameter structures used in  USM_STREAM_CMD_SET_ENCDEC_PARAM command */
+/* common declarations */
+struct usm_cfg_common {
+	u16 ch_cfg;
+	u16 bits_per_sample;
+	u32 sample_rate;
+	u32 dev_id;
+	u8 data_map[USM_MAX_PORT_NUMBER];
+} __packed;
+
+struct us_encdec_cfg {
+	u32 format_id;
+	struct usm_cfg_common cfg_common;
+	u16 params_size;
+	u8 *params;
+} __packed;
+
+/* Start/stop US signal detection */
+#define USM_SESSION_CMD_SIGNAL_DETECT_MODE		0x00012719
+
+struct usm_session_cmd_detect_info {
+	struct apr_hdr hdr;
+	u32 detect_mode;
+	u32 skip_interval;
+	u32 algorithm_cfg_size;
+} __packed;
+
+/* US signal detection result */
+#define USM_SESSION_EVENT_SIGNAL_DETECT_RESULT		0x00012720
+
+/* ======================================================================= */
+/*  Session Level commands */
+#define USM_CMD_SHARED_MEM_MAP_REGION		0x00012728
+struct usm_cmd_memory_map_region {
+	struct apr_hdr hdr;
+	u16            mempool_id;
+	u16            num_regions;
+	u32            flags;
+	u32            shm_addr_lsw;
+	u32            shm_addr_msw;
+	u32            mem_size_bytes;
+} __packed;
+
+#define USM_CMDRSP_SHARED_MEM_MAP_REGION	0x00012729
+struct usm_cmdrsp_memory_map_region {
+	u32            mem_map_handle;
+} __packed;
+
+#define USM_CMD_SHARED_MEM_UNMAP_REGION         0x0001272A
+struct usm_cmd_memory_unmap_region {
+	struct apr_hdr hdr;
+	u32            mem_map_handle;
+} __packed;
+
+#define USM_DATA_CMD_READ			0x00012724
+struct usm_stream_cmd_read {
+	struct apr_hdr hdr;
+	u32            buf_addr_lsw;
+	u32            buf_addr_msw;
+	u32            mem_map_handle;
+	u32            buf_size;
+	u32            seq_id;
+	u32            counter;
+} __packed;
+
+#define USM_DATA_EVENT_READ_DONE		0x00012725
+
+#define USM_DATA_CMD_WRITE			0x00012726
+struct usm_stream_cmd_write {
+	struct apr_hdr hdr;
+	u32            buf_addr_lsw;
+	u32            buf_addr_msw;
+	u32            mem_map_handle;
+	u32            buf_size;
+	u32            seq_id;
+	u32            res0;
+	u32            res1;
+	u32            res2;
+} __packed;
+
+#define USM_DATA_EVENT_WRITE_DONE		0x00012727
+
+struct usm_stream_media_format_update {
+	struct apr_hdr hdr;
+	u32 format_id;
+	/* <cfg_size> = sizeof(usm_cfg_common)+|transp_data| */
+	u32 cfg_size;
+	struct usm_cfg_common cfg_common;
+	/* Transparent configuration data for specific encoder */
+	u8  transp_data[USM_MAX_CFG_DATA_SIZE];
+} __packed;
+
+struct usm_encode_cfg_blk {
+	u32 frames_per_buf;
+	u32 format_id;
+	/* <cfg_size> = sizeof(usm_cfg_common)+|transp_data| */
+	u32 cfg_size;
+	struct usm_cfg_common cfg_common;
+	/* Transparent configuration data for specific encoder */
+	u8  transp_data[USM_MAX_CFG_DATA_SIZE];
+} __packed;
+
+struct usm_stream_cmd_encdec_cfg_blk {
+	struct apr_hdr hdr;
+	u32 param_id;
+	u32 param_size;
+	struct usm_encode_cfg_blk enc_blk;
+} __packed;
+
+#endif /* __APR_US_H__ */
diff --git a/include/linux/qdsp6v2/dsp_debug.h b/include/linux/qdsp6v2/dsp_debug.h
new file mode 100644
index 000000000000..bc1cd9ec8743
--- /dev/null
+++ b/include/linux/qdsp6v2/dsp_debug.h
@@ -0,0 +1,22 @@
+/* Copyright (c) 2010, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef __DSP_DEBUG_H_
+#define __DSP_DEBUG_H_
+
+typedef int (*dsp_state_cb)(int state);
+int dsp_debug_register(dsp_state_cb ptr);
+
+#define DSP_STATE_CRASHED         0x0
+#define DSP_STATE_CRASH_DUMP_DONE 0x1
+
+#endif
diff --git a/include/linux/qdsp6v2/rtac.h b/include/linux/qdsp6v2/rtac.h
new file mode 100644
index 000000000000..3e5433b23a51
--- /dev/null
+++ b/include/linux/qdsp6v2/rtac.h
@@ -0,0 +1,98 @@
+/* Copyright (c) 2011, 2013-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+
+#ifndef __RTAC_H__
+#define __RTAC_H__
+
+#include <sound/apr_audio-v2.h>
+
+/* Voice Modes */
+#define RTAC_CVP		0
+#define RTAC_CVS		1
+#define RTAC_VOICE_MODES	2
+
+#define RTAC_MAX_ACTIVE_DEVICES		4
+#define RTAC_MAX_ACTIVE_POPP		8
+
+#define DEFAULT_APP_TYPE	0x00011130
+
+enum {
+	ADM_RTAC_CAL,
+	ASM_RTAC_CAL,
+	VOICE_RTAC_CAL,
+	AFE_RTAC_CAL,
+	MAX_RTAC_BLOCKS
+};
+
+struct rtac_cal_mem_map_data {
+	uint32_t		map_size;
+	uint32_t		map_handle;
+	struct ion_client	*ion_client;
+	struct ion_handle	*ion_handle;
+};
+
+struct rtac_cal_data {
+	size_t			size;
+	void			*kvaddr;
+	phys_addr_t		paddr;
+};
+
+struct rtac_cal_block_data {
+	struct rtac_cal_mem_map_data	map_data;
+	struct rtac_cal_data		cal_data;
+};
+
+struct rtac_popp_data {
+	uint32_t	popp;
+	uint32_t	popp_topology;
+	uint32_t	app_type;
+};
+
+struct rtac_adm_data {
+	uint32_t		topology_id;
+	uint32_t		afe_topology;
+	uint32_t		afe_port;
+	uint32_t		copp;
+	uint32_t		num_of_popp;
+	uint32_t		app_type;
+	uint32_t		acdb_dev_id;
+	struct rtac_popp_data	popp[RTAC_MAX_ACTIVE_POPP];
+};
+
+struct rtac_adm {
+	uint32_t			num_of_dev;
+	struct rtac_adm_data		device[RTAC_MAX_ACTIVE_DEVICES];
+};
+
+void rtac_add_adm_device(u32 port_id, u32 copp_id, u32 path_id, u32 popp_id,
+			u32 app_type, u32 acdb_dev_id);
+void rtac_remove_adm_device(u32 port_id, u32 copp_id);
+void rtac_remove_popp_from_adm_devices(u32 popp_id);
+void rtac_add_voice(u32 cvs_handle, u32 cvp_handle, u32 rx_afe_port,
+	u32 tx_afe_port, u32 rx_acdb_id, u32 tx_acdb_id, u32 session_id);
+void rtac_remove_voice(u32 cvs_handle);
+void rtac_set_adm_handle(void *handle);
+bool rtac_make_adm_callback(uint32_t *payload, u32 payload_size);
+void rtac_copy_adm_payload_to_user(void *payload, u32 payload_size);
+void rtac_set_asm_handle(u32 session_id, void *handle);
+bool rtac_make_asm_callback(u32 session_id, uint32_t *payload,
+	u32 payload_size);
+void rtac_copy_asm_payload_to_user(void *payload, u32 payload_size);
+void rtac_set_voice_handle(u32 mode, void *handle);
+bool rtac_make_voice_callback(u32 mode, uint32_t *payload, u32 payload_size);
+void rtac_copy_voice_payload_to_user(void *payload, u32 payload_size);
+int rtac_clear_mapping(uint32_t cal_type);
+bool rtac_make_afe_callback(uint32_t *payload, u32 payload_size);
+void rtac_set_afe_handle(void *handle);
+void get_rtac_adm_data(struct rtac_adm *adm_data);
+#endif
diff --git a/include/linux/qdsp6v2/usf.h b/include/linux/qdsp6v2/usf.h
new file mode 100644
index 000000000000..544b624c2cda
--- /dev/null
+++ b/include/linux/qdsp6v2/usf.h
@@ -0,0 +1,298 @@
+/* Copyright (c) 2011-2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+
+#ifndef __USF_H__
+#define __USF_H__
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define USF_IOCTL_MAGIC 'U'
+
+#define US_SET_TX_INFO   _IOW(USF_IOCTL_MAGIC, 0, \
+				struct us_tx_info_type)
+#define US_START_TX      _IO(USF_IOCTL_MAGIC, 1)
+#define US_GET_TX_UPDATE _IOWR(USF_IOCTL_MAGIC, 2, \
+				struct us_tx_update_info_type)
+#define US_SET_RX_INFO   _IOW(USF_IOCTL_MAGIC, 3, \
+				struct us_rx_info_type)
+#define US_SET_RX_UPDATE _IOWR(USF_IOCTL_MAGIC, 4, \
+				struct us_rx_update_info_type)
+#define US_START_RX      _IO(USF_IOCTL_MAGIC, 5)
+
+#define US_STOP_TX      _IO(USF_IOCTL_MAGIC, 6)
+#define US_STOP_RX      _IO(USF_IOCTL_MAGIC, 7)
+
+#define US_SET_DETECTION _IOWR(USF_IOCTL_MAGIC, 8, \
+				struct us_detect_info_type)
+
+#define US_GET_VERSION  _IOWR(USF_IOCTL_MAGIC, 9, \
+				struct us_version_info_type)
+
+#define US_SET_TX_STREAM_PARAM   _IOW(USF_IOCTL_MAGIC, 10, \
+				struct us_stream_param_type)
+#define US_GET_TX_STREAM_PARAM  _IOWR(USF_IOCTL_MAGIC, 11, \
+				struct us_stream_param_type)
+#define US_SET_RX_STREAM_PARAM   _IOW(USF_IOCTL_MAGIC, 12, \
+				struct us_stream_param_type)
+#define US_GET_RX_STREAM_PARAM  _IOWR(USF_IOCTL_MAGIC, 13, \
+				struct us_stream_param_type)
+
+/* Special timeout values */
+#define USF_NO_WAIT_TIMEOUT	0x00000000
+/* Infinitive */
+#define USF_INFINITIVE_TIMEOUT	0xffffffff
+/* Default value, used by the driver */
+#define USF_DEFAULT_TIMEOUT	0xfffffffe
+
+/* US detection place (HW|FW) */
+enum us_detect_place_enum {
+/* US is detected in HW */
+	US_DETECT_HW,
+/* US is detected in FW */
+	US_DETECT_FW
+};
+
+/* US detection mode */
+enum us_detect_mode_enum {
+/* US detection is disabled */
+	US_DETECT_DISABLED_MODE,
+/* US detection is enabled in continue mode */
+	US_DETECT_CONTINUE_MODE,
+/* US detection is enabled in one shot mode */
+	US_DETECT_SHOT_MODE
+};
+
+/* Encoder (TX), decoder (RX) supported US data formats */
+#define USF_POINT_EPOS_FORMAT	0
+#define USF_RAW_FORMAT		1
+
+/* Indexes of event types, produced by the calculators */
+#define USF_TSC_EVENT_IND      0
+#define USF_TSC_PTR_EVENT_IND  1
+#define USF_MOUSE_EVENT_IND    2
+#define USF_KEYBOARD_EVENT_IND 3
+#define USF_TSC_EXT_EVENT_IND  4
+#define USF_MAX_EVENT_IND      5
+
+/* Types of events, produced by the calculators */
+#define USF_NO_EVENT 0
+#define USF_TSC_EVENT      (1 << USF_TSC_EVENT_IND)
+#define USF_TSC_PTR_EVENT  (1 << USF_TSC_PTR_EVENT_IND)
+#define USF_MOUSE_EVENT    (1 << USF_MOUSE_EVENT_IND)
+#define USF_KEYBOARD_EVENT (1 << USF_KEYBOARD_EVENT_IND)
+#define USF_TSC_EXT_EVENT  (1 << USF_TSC_EXT_EVENT_IND)
+#define USF_ALL_EVENTS         (USF_TSC_EVENT |\
+				USF_TSC_PTR_EVENT |\
+				USF_MOUSE_EVENT |\
+				USF_KEYBOARD_EVENT |\
+				USF_TSC_EXT_EVENT)
+
+/* min, max array dimension */
+#define MIN_MAX_DIM 2
+
+/* coordinates (x,y,z) array dimension */
+#define COORDINATES_DIM 3
+
+/* tilts (x,y) array dimension */
+#define TILTS_DIM 2
+
+/* Max size of the client name */
+#define USF_MAX_CLIENT_NAME_SIZE	20
+
+/* Max number of the ports (mics/speakers) */
+#define USF_MAX_PORT_NUM                8
+
+/* Info structure common for TX and RX */
+struct us_xx_info_type {
+/* Input:  general info */
+/* Name of the client - event calculator */
+	const char __user *client_name;
+/* Selected device identification, accepted in the kernel's CAD */
+	uint32_t dev_id;
+/* 0 - point_epos type; (e.g. 1 - gr_mmrd) */
+	uint32_t stream_format;
+/* Required sample rate in Hz */
+	uint32_t sample_rate;
+/* Size of a buffer (bytes) for US data transfer between the module and USF */
+	uint32_t buf_size;
+/* Number of the buffers for the US data transfer */
+	uint16_t buf_num;
+/* Number of the microphones (TX) or speakers(RX) */
+	uint16_t port_cnt;
+/* Microphones(TX) or speakers(RX) indexes in their enumeration */
+	uint8_t  port_id[USF_MAX_PORT_NUM];
+/* Bits per sample 16 or 32 */
+	uint16_t bits_per_sample;
+/* Input:  Transparent info for encoder in the LPASS */
+/* Parameters data size in bytes */
+	uint16_t params_data_size;
+/* Pointer to the parameters */
+	uint8_t __user *params_data;
+/* Max size of buffer for get and set parameter */
+	uint32_t max_get_set_param_buf_size;
+};
+
+struct us_input_info_type {
+	/* Touch screen dimensions: min & max;for input module */
+	int tsc_x_dim[MIN_MAX_DIM];
+	int tsc_y_dim[MIN_MAX_DIM];
+	int tsc_z_dim[MIN_MAX_DIM];
+	/* Touch screen tilt dimensions: min & max;for input module */
+	int tsc_x_tilt[MIN_MAX_DIM];
+	int tsc_y_tilt[MIN_MAX_DIM];
+	/* Touch screen pressure limits: min & max; for input module */
+	int tsc_pressure[MIN_MAX_DIM];
+	/* The requested buttons bitmap */
+	uint16_t req_buttons_bitmap;
+	/* Bitmap of types of events (USF_X_EVENT), produced by calculator */
+	uint16_t event_types;
+	/* Bitmap of types of events from devs, conflicting with USF */
+	uint16_t conflicting_event_types;
+};
+
+struct us_tx_info_type {
+	/* Common info */
+	struct us_xx_info_type us_xx_info;
+	/* Info specific for TX*/
+	struct us_input_info_type input_info;
+};
+
+struct us_rx_info_type {
+	/* Common info */
+	struct us_xx_info_type us_xx_info;
+	/* Info specific for RX*/
+};
+
+struct point_event_type {
+/* Pen coordinates (x, y, z) in units, defined by <coordinates_type>  */
+	int coordinates[COORDINATES_DIM];
+	/* {x;y}  in transparent units */
+	int inclinations[TILTS_DIM];
+/* [0-1023] (10bits); 0 - pen up */
+	uint32_t pressure;
+/* Bitmap for button state. 1 - down, 0 - up */
+	uint16_t buttons_state_bitmap;
+};
+
+/* Mouse buttons, supported by USF */
+#define USF_BUTTON_LEFT_MASK   1
+#define USF_BUTTON_MIDDLE_MASK 2
+#define USF_BUTTON_RIGHT_MASK  4
+struct mouse_event_type {
+/* The mouse relative movement (dX, dY, dZ) */
+	int rels[COORDINATES_DIM];
+/* Bitmap of mouse buttons states: 1 - down, 0 - up; */
+	uint16_t buttons_states;
+};
+
+struct key_event_type {
+/*  Calculated MS key- see input.h. */
+	uint32_t key;
+/* Keyboard's key state: 1 - down, 0 - up; */
+	uint8_t key_state;
+};
+
+struct usf_event_type {
+/* Event sequence number */
+	uint32_t seq_num;
+/* Event generation system time */
+	uint32_t timestamp;
+/* Destination input event type index (e.g. touch screen, mouse, key) */
+	uint16_t event_type_ind;
+	union {
+		struct point_event_type point_event;
+		struct mouse_event_type mouse_event;
+		struct key_event_type   key_event;
+	} event_data;
+};
+
+struct us_tx_update_info_type {
+/* Input  general: */
+/* Number of calculated events */
+	uint16_t event_counter;
+/* Calculated events or NULL */
+	struct usf_event_type __user *event;
+/* Pointer (read index) to the end of available region */
+/* in the shared US data memory */
+	uint32_t free_region;
+/* Time (sec) to wait for data or special values: */
+/* USF_NO_WAIT_TIMEOUT, USF_INFINITIVE_TIMEOUT, USF_DEFAULT_TIMEOUT */
+	uint32_t timeout;
+/* Events (from conflicting devs) to be disabled/enabled */
+	uint16_t event_filters;
+
+/* Input  transparent data: */
+/* Parameters size */
+	uint16_t params_data_size;
+/* Pointer to the parameters */
+	uint8_t __user *params_data;
+/* Output parameters: */
+/* Pointer (write index) to the end of ready US data region */
+/* in the shared memory */
+	uint32_t ready_region;
+};
+
+struct us_rx_update_info_type {
+/* Input  general: */
+/* Pointer (write index) to the end of ready US data region */
+/* in the shared memory */
+	uint32_t ready_region;
+/* Input  transparent data: */
+/* Parameters size */
+	uint16_t params_data_size;
+/* pPointer to the parameters */
+	uint8_t __user *params_data;
+/* Output parameters: */
+/* Pointer (read index) to the end of available region */
+/* in the shared US data memory */
+	uint32_t free_region;
+};
+
+struct us_detect_info_type {
+/* US detection place (HW|FW) */
+/* NA in the Active and OFF states */
+	enum us_detect_place_enum us_detector;
+/* US detection mode */
+	enum us_detect_mode_enum  us_detect_mode;
+/* US data dropped during this time (msec) */
+	uint32_t skip_time;
+/* Transparent data size */
+	uint16_t params_data_size;
+/* Pointer to the transparent data */
+	uint8_t __user *params_data;
+/* Time (sec) to wait for US presence event */
+	uint32_t detect_timeout;
+/* Out parameter: US presence */
+	bool is_us;
+};
+
+struct us_version_info_type {
+/* Size of memory for the version string */
+	uint16_t buf_size;
+/* Pointer to the memory for the version string */
+	char __user *pbuf;
+};
+
+struct us_stream_param_type {
+/* Id of module */
+	uint32_t module_id;
+/* Id of parameter */
+	uint32_t param_id;
+/* Size of memory of the parameter buffer */
+	uint32_t buf_size;
+/* Pointer to the memory of the parameter buffer */
+	uint8_t __user *pbuf;
+};
+
+#endif /* __USF_H__ */
diff --git a/include/soc/qcom/liquid_dock.h b/include/soc/qcom/liquid_dock.h
new file mode 100644
index 000000000000..3668224f1938
--- /dev/null
+++ b/include/soc/qcom/liquid_dock.h
@@ -0,0 +1,21 @@
+/* Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#include <linux/notifier.h>
+
+#if IS_ENABLED(CONFIG_QCOM_LIQUID_DOCK)
+void register_liquid_dock_notify(struct notifier_block *nb);
+void unregister_liquid_dock_notify(struct notifier_block *nb);
+#else
+static inline void register_liquid_dock_notify(struct notifier_block *nb) { }
+static inline void unregister_liquid_dock_notify(struct notifier_block *nb) { }
+#endif
diff --git a/include/sound/adsp_err.h b/include/sound/adsp_err.h
new file mode 100644
index 000000000000..68fd61e59633
--- /dev/null
+++ b/include/sound/adsp_err.h
@@ -0,0 +1,116 @@
+/*
+ * Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __ADSP_ERR__
+#define __ADSP_ERR__
+
+#include <linux/errno.h>
+#include <sound/apr_audio-v2.h>
+
+
+/* ERROR STRING */
+/* Success. The operation completed with no errors. */
+#define ADSP_EOK_STR          "ADSP_EOK"
+/* General failure. */
+#define ADSP_EFAILED_STR      "ADSP_EFAILED"
+/* Bad operation parameter. */
+#define ADSP_EBADPARAM_STR    "ADSP_EBADPARAM"
+/* Unsupported routine or operation. */
+#define ADSP_EUNSUPPORTED_STR "ADSP_EUNSUPPORTED"
+/* Unsupported version. */
+#define ADSP_EVERSION_STR     "ADSP_EVERSION"
+/* Unexpected problem encountered. */
+#define ADSP_EUNEXPECTED_STR  "ADSP_EUNEXPECTED"
+/* Unhandled problem occurred. */
+#define ADSP_EPANIC_STR       "ADSP_EPANIC"
+/* Unable to allocate resource. */
+#define ADSP_ENORESOURCE_STR  "ADSP_ENORESOURCE"
+/* Invalid handle. */
+#define ADSP_EHANDLE_STR      "ADSP_EHANDLE"
+/* Operation is already processed. */
+#define ADSP_EALREADY_STR     "ADSP_EALREADY"
+/* Operation is not ready to be processed. */
+#define ADSP_ENOTREADY_STR    "ADSP_ENOTREADY"
+/* Operation is pending completion. */
+#define ADSP_EPENDING_STR     "ADSP_EPENDING"
+/* Operation could not be accepted or processed. */
+#define ADSP_EBUSY_STR        "ADSP_EBUSY"
+/* Operation aborted due to an error. */
+#define ADSP_EABORTED_STR     "ADSP_EABORTED"
+/* Operation preempted by a higher priority. */
+#define ADSP_EPREEMPTED_STR   "ADSP_EPREEMPTED"
+/* Operation requests intervention to complete. */
+#define ADSP_ECONTINUE_STR    "ADSP_ECONTINUE"
+/* Operation requests immediate intervention to complete. */
+#define ADSP_EIMMEDIATE_STR   "ADSP_EIMMEDIATE"
+/* Operation is not implemented. */
+#define ADSP_ENOTIMPL_STR     "ADSP_ENOTIMPL"
+/* Operation needs more data or resources. */
+#define ADSP_ENEEDMORE_STR    "ADSP_ENEEDMORE"
+/* Operation does not have memory. */
+#define ADSP_ENOMEMORY_STR    "ADSP_ENOMEMORY"
+/* Item does not exist. */
+#define ADSP_ENOTEXIST_STR    "ADSP_ENOTEXIST"
+/* Unexpected error code. */
+#define ADSP_ERR_MAX_STR      "ADSP_ERR_MAX"
+
+
+struct adsp_err_code {
+	int		lnx_err_code;
+	char	*adsp_err_str;
+};
+
+
+static struct adsp_err_code adsp_err_code_info[ADSP_ERR_MAX+1] = {
+	{ 0, ADSP_EOK_STR},
+	{ -ENOTRECOVERABLE, ADSP_EFAILED_STR},
+	{ -EINVAL, ADSP_EBADPARAM_STR},
+	{ -ENOSYS, ADSP_EUNSUPPORTED_STR},
+	{ -ENOPROTOOPT, ADSP_EVERSION_STR},
+	{ -ENOTRECOVERABLE, ADSP_EUNEXPECTED_STR},
+	{ -ENOTRECOVERABLE, ADSP_EPANIC_STR},
+	{ -ENOSPC, ADSP_ENORESOURCE_STR},
+	{ -EBADR, ADSP_EHANDLE_STR},
+	{ -EALREADY, ADSP_EALREADY_STR},
+	{ -EPERM, ADSP_ENOTREADY_STR},
+	{ -EINPROGRESS, ADSP_EPENDING_STR},
+	{ -EBUSY, ADSP_EBUSY_STR},
+	{ -ECANCELED, ADSP_EABORTED_STR},
+	{ -EAGAIN, ADSP_EPREEMPTED_STR},
+	{ -EAGAIN, ADSP_ECONTINUE_STR},
+	{ -EAGAIN, ADSP_EIMMEDIATE_STR},
+	{ -EAGAIN, ADSP_ENOTIMPL_STR},
+	{ -ENODATA, ADSP_ENEEDMORE_STR},
+	{ -EADV, ADSP_ERR_MAX_STR},
+	{ -ENOMEM, ADSP_ENOMEMORY_STR},
+	{ -ENODEV, ADSP_ENOTEXIST_STR},
+	{ -EADV, ADSP_ERR_MAX_STR},
+};
+
+static inline int adsp_err_get_lnx_err_code(u32 adsp_error)
+{
+	if (adsp_error > ADSP_ERR_MAX)
+		return adsp_err_code_info[ADSP_ERR_MAX].lnx_err_code;
+	else
+		return adsp_err_code_info[adsp_error].lnx_err_code;
+}
+
+static inline char *adsp_err_get_err_str(u32 adsp_error)
+{
+	if (adsp_error > ADSP_ERR_MAX)
+		return adsp_err_code_info[ADSP_ERR_MAX].adsp_err_str;
+	else
+		return adsp_err_code_info[adsp_error].adsp_err_str;
+}
+
+#endif
diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
new file mode 100644
index 000000000000..1add15a7323e
--- /dev/null
+++ b/include/sound/apr_audio-v2.h
@@ -0,0 +1,8841 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License version 2 and
+* only 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.
+*/
+
+
+#ifndef _APR_AUDIO_V2_H_
+#define _APR_AUDIO_V2_H_
+
+#include <linux/qdsp6v2/apr.h>
+
+/* size of header needed for passing data out of band */
+#define APR_CMD_OB_HDR_SZ  12
+
+/* size of header needed for getting data */
+#define APR_CMD_GET_HDR_SZ 16
+
+struct param_outband {
+	size_t       size;
+	void        *kvaddr;
+	phys_addr_t  paddr;
+};
+
+#define ADSP_ADM_VERSION    0x00070000
+
+#define ADM_CMD_SHARED_MEM_MAP_REGIONS    0x00010322
+#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323
+#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324
+
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D
+/* Enumeration for an audio Rx matrix ID.*/
+#define ADM_MATRIX_ID_AUDIO_RX              0
+
+#define ADM_MATRIX_ID_AUDIO_TX              1
+
+#define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX   2
+/* Enumeration for an audio Tx matrix ID.*/
+#define ADM_MATRIX_ID_AUDIOX              1
+
+#define ADM_MAX_COPPS 5
+
+/* make sure this matches with msm_audio_calibration */
+#define SP_V2_NUM_MAX_SPKR 2
+
+/*  Session map node structure.
+*	Immediately following this structure are num_copps
+*	entries of COPP IDs. The COPP IDs are 16 bits, so
+*	there might be a padding 16-bit field if num_copps
+*	is odd.
+*/
+struct adm_session_map_node_v5 {
+	u16                  session_id;
+/* Handle of the ASM session to be routed. Supported values: 1
+* to 8.
+*/
+
+
+	u16                  num_copps;
+	/* Number of COPPs to which this session is to be routed.
+			Supported values: 0 < num_copps <= ADM_MAX_COPPS.
+	*/
+} __packed;
+
+/*  Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command.
+*	Immediately following this structure are num_sessions of the session map
+*	node payload (adm_session_map_node_v5).
+*/
+
+struct adm_cmd_matrix_map_routings_v5 {
+	struct apr_hdr	hdr;
+
+	u32                  matrix_id;
+/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx
+* (1). Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
+* macros to set this field.
+*/
+	u32                  num_sessions;
+	/* Number of sessions being updated by this command (optional).*/
+} __packed;
+
+/* This command allows a client to open a COPP/Voice Proc. TX module
+*	and sets up	the device session: Matrix -> COPP -> AFE on the RX
+*	and AFE -> COPP -> Matrix on the TX. This enables PCM data to
+*	be transferred to/from the endpoint (AFEPortID).
+*
+*	@return
+*	#ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and
+*	COPP ID.
+*/
+#define ADM_CMD_DEVICE_OPEN_V5                          0x00010326
+
+/* Definition for a low latency stream session. */
+#define ADM_LOW_LATENCY_DEVICE_SESSION			0x2000
+
+/* Definition for a ultra low latency stream session. */
+#define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION		0x4000
+
+/* Definition for a ultra low latency with Post Processing stream session. */
+#define ADM_ULL_POST_PROCESSING_DEVICE_SESSION		0x8000
+
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION                                      0
+
+/* Indicates that endpoint_id_2 is to be ignored.*/
+#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE				0xFFFF
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP		 1
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP		 2
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP	 3
+
+/* Indicates that an audio COPP is to send/receive a mono PCM
+ * stream to/from
+ *	END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO		1
+
+/* Indicates that an audio COPP is to send/receive a
+ *	stereo PCM stream to/from END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO		2
+
+/* Sample rate is 8000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000
+
+/* Sample rate is 16000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000
+
+/* Sample rate is 48000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000
+
+/* Definition for a COPP live input flag bitmask.*/
+#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U)
+
+/* Definition for a COPP live shift value bitmask.*/
+#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG	 0
+
+/* Definition for the COPP ID bitmask.*/
+#define ADM_BIT_MASK_COPP_ID  (0x0000FFFFUL)
+
+/* Definition for the COPP ID shift value.*/
+#define ADM_SHIFT_COPP_ID	0
+
+/* Definition for the service ID bitmask.*/
+#define ADM_BIT_MASK_SERVICE_ID  (0x00FF0000UL)
+
+/* Definition for the service ID shift value.*/
+#define ADM_SHIFT_SERVICE_ID	16
+
+/* Definition for the domain ID bitmask.*/
+#define ADM_BIT_MASK_DOMAIN_ID    (0xFF000000UL)
+
+/* Definition for the domain ID shift value.*/
+#define ADM_SHIFT_DOMAIN_ID	24
+
+/*  ADM device open command payload of the
+	#ADM_CMD_DEVICE_OPEN_V5 command.
+*/
+struct adm_cmd_device_open_v5 {
+	struct apr_hdr		hdr;
+	u16                  flags;
+/* Reserved for future use. Clients must set this field
+ * to zero.
+ */
+
+	u16                  mode_of_operation;
+/* Specifies whether the COPP must be opened on the Tx or Rx
+ * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
+ * supported values and interpretation.
+ * Supported values:
+ * - 0x1 -- Rx path COPP
+ * - 0x2 -- Tx path live COPP
+ * - 0x3 -- Tx path nonlive COPP
+ * Live connections cause sample discarding in the Tx device
+ * matrix if the destination output ports do not pull them
+ * fast enough. Nonlive connections queue the samples
+ * indefinitely.
+ */
+
+	u16                  endpoint_id_1;
+/* Logical and physical endpoint ID of the audio path.
+ * If the ID is a voice processor Tx block, it receives near
+ * samples.	Supported values: Any pseudoport, AFE Rx port,
+ * or AFE Tx port For a list of valid IDs, refer to
+ * @xhyperref{Q4,[Q4]}.
+ * Q4 = Hexagon Multimedia: AFE Interface Specification
+ */
+
+	u16                  endpoint_id_2;
+/* Logical and physical endpoint ID 2 for a voice processor
+ * Tx block.
+ * This is not applicable to audio COPP.
+ * Supported values:
+ * - AFE Rx port
+ * - 0xFFFF -- Endpoint 2 is unavailable and the voice
+ * processor Tx
+ * block ignores this endpoint
+ * When the voice processor Tx block is created on the audio
+ * record path,
+ * it can receive far-end samples from an AFE Rx port if the
+ * voice call
+ * is active. The ID of the AFE port is provided in this
+ * field.
+ * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
+ */
+
+	u32                  topology_id;
+	/* Audio COPP topology ID; 32-bit GUID. */
+
+	u16                  dev_num_channel;
+/* Number of channels the audio COPP sends to/receives from
+ * the endpoint.
+ * Supported values: 1 to 8.
+ * The value is ignored for the voice processor Tx block,
+ * where channel
+ * configuration is derived from the topology ID.
+ */
+
+	u16                  bit_width;
+/* Bit width (in bits) that the audio COPP sends to/receives
+ * from the
+ * endpoint. The value is ignored for the voice processing
+ * Tx block,
+ * where the PCM width is 16 bits.
+ */
+
+	u32                  sample_rate;
+/* Sampling rate at which the audio COPP/voice processor
+ * Tx block
+ * interfaces with the endpoint.
+ * Supported values for voice processor Tx: 8000, 16000,
+ * 48000 Hz
+ * Supported values for audio COPP: >0 and <=192 kHz
+ */
+
+	u8                   dev_channel_mapping[8];
+/* Array of channel mapping of buffers that the audio COPP
+ * sends to the endpoint. Channel[i] mapping describes channel
+ * I inside the buffer, where 0 < i < dev_num_channel.
+ * This value is relevent only for an audio Rx COPP.
+ * For the voice processor block and Tx audio block, this field
+ * is set to zero and is ignored.
+ */
+} __packed;
+
+/*
+ *	This command allows the client to close a COPP and disconnect
+ *	the device session.
+ */
+#define ADM_CMD_DEVICE_CLOSE_V5                         0x00010327
+
+/* Sets one or more parameters to a COPP.
+*/
+#define ADM_CMD_SET_PP_PARAMS_V5                        0x00010328
+
+/*  Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command.
+ *	If the data_payload_addr_lsw and data_payload_addr_msw element
+ *	are NULL, a series of adm_param_datastructures immediately
+ *	follows, whose total size is data_payload_size bytes.
+ */
+struct adm_cmd_set_pp_params_v5 {
+	struct apr_hdr hdr;
+	u32		payload_addr_lsw;
+	/* LSW of parameter data payload address.*/
+	u32		payload_addr_msw;
+	/* MSW of parameter data payload address.*/
+
+	u32		mem_map_handle;
+/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * command */
+/* If mem_map_handle is zero implies the message is in
+ * the payload */
+
+	u32		payload_size;
+/* Size in bytes of the variable payload accompanying this
+ * message or
+ * in shared memory. This is used for parsing the parameter
+ * payload.
+ */
+} __packed;
+
+/*  Payload format for COPP parameter data.
+ * Immediately following this structure are param_size bytes
+ * of parameter
+ * data.
+ */
+struct adm_param_data_v5 {
+	u32                  module_id;
+	/* Unique ID of the module. */
+	u32                  param_id;
+	/* Unique ID of the parameter. */
+	u16                  param_size;
+	/* Data size of the param_id/module_id combination.
+	This value is a
+		multiple of 4 bytes. */
+	u16                  reserved;
+	/* Reserved for future enhancements.
+	 * This field must be set to zero.
+	 */
+} __packed;
+
+/* set customized mixing on matrix mixer */
+#define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5                        0x00010344
+struct adm_cmd_set_pspd_mtmx_strtr_params_v5 {
+	struct apr_hdr hdr;
+	/* LSW of parameter data payload address.*/
+	u32		payload_addr_lsw;
+	/* MSW of parameter data payload address.*/
+	u32		payload_addr_msw;
+	/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */
+	/* command. If mem_map_handle is zero implies the message is in */
+	/* the payload */
+	u32		mem_map_handle;
+	/* Size in bytes of the variable payload accompanying this */
+	/* message or in shared memory. This is used for parsing the */
+	/* parameter payload. */
+	u32		payload_size;
+	u16		direction;
+	u16		sessionid;
+	u16		deviceid;
+	u16		reserved;
+} __packed;
+
+/* Defined specifically for in-band use, includes params */
+struct adm_cmd_set_pp_params_inband_v5 {
+	struct apr_hdr hdr;
+	/* LSW of parameter data payload address.*/
+	u32             payload_addr_lsw;
+	/* MSW of parameter data payload address.*/
+	u32             payload_addr_msw;
+	/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */
+	/* command. If mem_map_handle is zero implies the message is in */
+	/* the payload */
+	u32             mem_map_handle;
+	/* Size in bytes of the variable payload accompanying this */
+	/* message or in shared memory. This is used for parsing the */
+	/* parameter payload. */
+	u32             payload_size;
+	/* Parameters passed for in band payload */
+	struct adm_param_data_v5        params;
+} __packed;
+
+/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+#define ADM_CMDRSP_DEVICE_OPEN_V5                      0x00010329
+
+/*  Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message,
+ *	which returns the
+ *	status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+struct adm_cmd_rsp_device_open_v5 {
+	u32                  status;
+	/* Status message (error code).*/
+
+	u16                  copp_id;
+	/* COPP ID:  Supported values: 0 <= copp_id < ADM_MAX_COPPS*/
+
+	u16                  reserved;
+	/* Reserved. This field must be set to zero.*/
+} __packed;
+
+/* This command allows a query of one COPP parameter.
+*/
+#define ADM_CMD_GET_PP_PARAMS_V5                                0x0001032A
+
+/*  Payload an #ADM_CMD_GET_PP_PARAMS_V5 command.
+*/
+struct adm_cmd_get_pp_params_v5 {
+	struct apr_hdr hdr;
+	u32                  data_payload_addr_lsw;
+	/* LSW of parameter data payload address.*/
+
+	u32                  data_payload_addr_msw;
+	/* MSW of parameter data payload address.*/
+
+	/* If the mem_map_handle is non zero,
+	 * on ACK, the ParamData payloads begin at
+	 * the address specified (out-of-band).
+	 */
+
+	u32                  mem_map_handle;
+	/* Memory map handle returned
+	 * by ADM_CMD_SHARED_MEM_MAP_REGIONS command.
+	 * If the mem_map_handle is 0, it implies that
+	 * the ACK's payload will contain the ParamData (in-band).
+	 */
+
+	u32                  module_id;
+	/* Unique ID of the module. */
+
+	u32                  param_id;
+	/* Unique ID of the parameter. */
+
+	u16                  param_max_size;
+	/* Maximum data size of the parameter
+	 *ID/module ID combination. This
+	 * field is a multiple of 4 bytes.
+	 */
+	u16                  reserved;
+	/* Reserved for future enhancements.
+	 * This field must be set to zero.
+	 */
+} __packed;
+
+/* Returns parameter values
+ *	in response to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ */
+#define ADM_CMDRSP_GET_PP_PARAMS_V5		0x0001032B
+
+/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message,
+ * which returns parameter values in response
+ * to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ * Immediately following this
+ * structure is the adm_param_data_v5
+ * structure containing the pre/postprocessing
+ * parameter data. For an in-band
+ * scenario, the variable payload depends
+ * on the size of the parameter.
+*/
+struct adm_cmd_rsp_get_pp_params_v5 {
+	u32                  status;
+	/* Status message (error code).*/
+} __packed;
+
+/* Structure for holding soft stepping volume parameters. */
+
+/*
+ * Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+
+struct audproc_softvolume_params {
+	u32 period;
+	u32 step;
+	u32 rampingcurve;
+} __packed;
+
+struct audproc_volume_ctrl_master_gain {
+	struct adm_cmd_set_pp_params_v5 params;
+	struct adm_param_data_v5 data;
+	/* Linear gain in Q13 format. */
+	uint16_t                  master_gain;
+	/* Clients must set this field to zero. */
+	uint16_t                  reserved;
+} __packed;
+
+struct audproc_soft_step_volume_params {
+	struct adm_cmd_set_pp_params_v5 params;
+	struct adm_param_data_v5 data;
+/*
+ * Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+	uint32_t                  period;
+/*
+ * Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+	uint32_t                  step;
+/*
+ * Ramping curve type.
+ * Supported values:
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+	uint32_t                  ramping_curve;
+} __packed;
+
+struct audproc_enable_param_t {
+	struct adm_cmd_set_pp_params_inband_v5 pp_params;
+	/*
+	 * Specifies whether the Audio processing module is enabled.
+	 * This parameter is generic/common parameter to configure or
+	 * determine the state of any audio processing module.
+
+	 * @values 0 : Disable 1: Enable
+	 */
+	uint32_t                  enable;
+};
+
+/*
+ * Allows a client to control the gains on various session-to-COPP paths.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_V5                                 0x0001032C
+
+/* Indicates that the target gain in the
+ *	current adm_session_copp_gain_v5
+ *	structure is to be applied to all
+ *	the session-to-COPP paths that exist for
+ *	the specified session.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS     0xFFFF
+
+/* Indicates that the target gain is
+ * to be immediately applied to the
+ * specified session-to-COPP path,
+ * without a ramping fashion.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE         0x0000
+
+/* Enumeration for a linear ramping curve.*/
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR               0x0000
+
+/*  Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
+ * Immediately following this structure are num_gains of the
+ * adm_session_copp_gain_v5structure.
+ */
+struct adm_cmd_matrix_ramp_gains_v5 {
+	u32                  matrix_id;
+/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
+ * Use the ADM_MATRIX_ID_AUDIO_RX or  ADM_MATRIX_ID_AUDIOX
+ * macros to set this field.
+*/
+
+	u16                  num_gains;
+	/* Number of gains being applied. */
+
+	u16                  reserved_for_align;
+	/* Reserved. This field must be set to zero.*/
+} __packed;
+
+/*  Session-to-COPP path gain structure, used by the
+ *	#ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
+ *	This structure specifies the target
+ *	gain (per channel) that must be applied
+ *	to a particular session-to-COPP path in
+ *	the audio matrix. The structure can
+ *	also be used to apply the gain globally
+ *	to all session-to-COPP paths that
+ *	exist for the given session.
+ *	The aDSP uses device channel mapping to
+ *	determine which channel gains to
+ *	use from this command. For example,
+ *	if the device is configured as stereo,
+ *	the aDSP uses only target_gain_ch_1 and
+ *	target_gain_ch_2, and it ignores
+ *	the others.
+ */
+struct adm_session_copp_gain_v5 {
+	u16                  session_id;
+/* Handle of the ASM session.
+ *	Supported values: 1 to 8.
+ */
+
+	u16                  copp_id;
+/* Handle of the COPP. Gain will be applied on the Session ID
+ * COPP ID path.
+ */
+
+	u16                  ramp_duration;
+/* Duration (in milliseconds) of the ramp over
+ * which target gains are
+ * to be applied. Use
+ * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE
+ * to indicate that gain must be applied immediately.
+ */
+
+	u16                  step_duration;
+/* Duration (in milliseconds) of each step in the ramp.
+ * This parameter is ignored if ramp_duration is equal to
+ * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE.
+ * Supported value: 1
+ */
+
+	u16                  ramp_curve;
+/* Type of ramping curve.
+ * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR
+ */
+
+	u16                  reserved_for_align;
+	/* Reserved. This field must be set to zero. */
+
+	u16                  target_gain_ch_1;
+	/* Target linear gain for channel 1 in Q13 format; */
+
+	u16                  target_gain_ch_2;
+	/* Target linear gain for channel 2 in Q13 format; */
+
+	u16                  target_gain_ch_3;
+	/* Target linear gain for channel 3 in Q13 format; */
+
+	u16                  target_gain_ch_4;
+	/* Target linear gain for channel 4 in Q13 format; */
+
+	u16                  target_gain_ch_5;
+	/* Target linear gain for channel 5 in Q13 format; */
+
+	u16                  target_gain_ch_6;
+	/* Target linear gain for channel 6 in Q13 format; */
+
+	u16                  target_gain_ch_7;
+	/* Target linear gain for channel 7 in Q13 format; */
+
+	u16                  target_gain_ch_8;
+	/* Target linear gain for channel 8 in Q13 format; */
+} __packed;
+
+/* Allows to set mute/unmute on various session-to-COPP paths.
+ *	For every session-to-COPP path (stream-device interconnection),
+ *	mute/unmute can be set individually on the output channels.
+ */
+#define ADM_CMD_MATRIX_MUTE_V5                                0x0001032D
+
+/* Indicates that mute/unmute in the
+ *	current adm_session_copp_mute_v5structure
+ *	is to be applied to all the session-to-COPP
+ *	paths that exist for the specified session.
+ */
+#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS     0xFFFF
+
+/*  Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/
+struct adm_cmd_matrix_mute_v5 {
+	u32                  matrix_id;
+/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
+ * Use the ADM_MATRIX_ID_AUDIO_RX or  ADM_MATRIX_ID_AUDIOX
+ * macros to set this field.
+ */
+
+	u16                  session_id;
+/* Handle of the ASM session.
+ * Supported values: 1 to 8.
+ */
+
+	u16                  copp_id;
+/* Handle of the COPP.
+ * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS
+ * to indicate that mute/unmute must be applied to
+ * all the COPPs connected to session_id.
+ * Supported values:
+ * - 0xFFFF -- Apply mute/unmute to all connected COPPs
+ * - Other values -- Valid COPP ID
+ */
+
+	u8                  mute_flag_ch_1;
+	/* Mute flag for channel 1 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_2;
+	/* Mute flag for channel 2 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_3;
+	/* Mute flag for channel 3 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_4;
+	/* Mute flag for channel 4 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_5;
+	/* Mute flag for channel 5 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_6;
+	/* Mute flag for channel 6 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_7;
+	/* Mute flag for channel 7 is set to unmute (0) or mute (1). */
+
+	u8                  mute_flag_ch_8;
+	/* Mute flag for channel 8 is set to unmute (0) or mute (1). */
+
+	u16                 ramp_duration;
+/* Period (in milliseconds) over which the soft mute/unmute will be
+ * applied.
+ * Supported values: 0 (Default) to 0xFFFF
+ * The default of 0 means mute/unmute will be applied immediately.
+ */
+
+	u16                 reserved_for_align;
+	/* Clients must set this field to zero.*/
+} __packed;
+
+#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8)
+
+struct asm_aac_stereo_mix_coeff_selection_param_v2 {
+	struct apr_hdr          hdr;
+	u32                     param_id;
+	u32                     param_size;
+	u32                     aac_stereo_mix_coeff_flag;
+} __packed;
+
+/* Allows a client to connect the desired stream to
+ * the desired AFE port through the stream router
+ *
+ * This command allows the client to connect specified session to
+ * specified AFE port. This is used for compressed streams only
+ * opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command.
+ *
+ * @prerequisites
+ * Session ID and AFE Port ID must be valid.
+ * #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED
+ * must have been called on this session.
+ */
+
+#define ADM_CMD_CONNECT_AFE_PORT_V5	0x0001032E
+#define ADM_CMD_DISCONNECT_AFE_PORT_V5	0x0001032F
+/* Enumeration for the Rx stream router ID.*/
+#define ADM_STRTR_ID_RX                    0
+/* Enumeration for the Tx stream router ID.*/
+#define ADM_STRTR_IDX                    1
+
+/*  Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/
+struct adm_cmd_connect_afe_port_v5 {
+	struct apr_hdr     hdr;
+	u8                  mode;
+/* ID of the stream router (RX/TX). Use the
+ * ADM_STRTR_ID_RX or ADM_STRTR_IDX macros
+ * to set this field.
+ */
+
+	u8                  session_id;
+	/* Session ID of the stream to connect */
+
+	u16                 afe_port_id;
+	/* Port ID of the AFE port to connect to.*/
+	u32                 num_channels;
+/* Number of device channels
+ * Supported values: 2(Audio Sample Packet),
+ * 8 (HBR Audio Stream Sample Packet)
+ */
+
+	u32                 sampling_rate;
+/* Device sampling rate
+* Supported values: Any
+*/
+} __packed;
+
+
+/* adsp_adm_api.h */
+
+
+/* Port ID. Update afe_get_port_index
+ *	when a new port is added here. */
+#define PRIMARY_I2S_RX 0
+#define PRIMARY_I2S_TX 1
+#define SECONDARY_I2S_RX 4
+#define SECONDARY_I2S_TX 5
+#define MI2S_RX 6
+#define MI2S_TX 7
+#define HDMI_RX 8
+#define RSVD_2 9
+#define RSVD_3 10
+#define DIGI_MIC_TX 11
+#define VOICE2_PLAYBACK_TX 0x8002
+#define VOICE_RECORD_RX 0x8003
+#define VOICE_RECORD_TX 0x8004
+#define VOICE_PLAYBACK_TX 0x8005
+
+/* Slimbus Multi channel port id pool  */
+#define SLIMBUS_0_RX		0x4000
+#define SLIMBUS_0_TX		0x4001
+#define SLIMBUS_1_RX		0x4002
+#define SLIMBUS_1_TX		0x4003
+#define SLIMBUS_2_RX		0x4004
+#define SLIMBUS_2_TX		0x4005
+#define SLIMBUS_3_RX		0x4006
+#define SLIMBUS_3_TX		0x4007
+#define SLIMBUS_4_RX		0x4008
+#define SLIMBUS_4_TX		0x4009
+#define SLIMBUS_5_RX		0x400a
+#define SLIMBUS_5_TX		0x400b
+#define SLIMBUS_6_RX		0x400c
+#define SLIMBUS_6_TX		0x400d
+#define SLIMBUS_PORT_LAST	SLIMBUS_6_TX
+#define INT_BT_SCO_RX 0x3000
+#define INT_BT_SCO_TX 0x3001
+#define INT_BT_A2DP_RX 0x3002
+#define INT_FM_RX 0x3004
+#define INT_FM_TX 0x3005
+#define RT_PROXY_PORT_001_RX	0x2000
+#define RT_PROXY_PORT_001_TX	0x2001
+
+#define AFE_PORT_INVALID 0xFFFF
+#define SLIMBUS_INVALID AFE_PORT_INVALID
+
+#define AFE_PORT_CMD_START 0x000100ca
+
+#define AFE_EVENT_RTPORT_START 0
+#define AFE_EVENT_RTPORT_STOP 1
+#define AFE_EVENT_RTPORT_LOW_WM 2
+#define AFE_EVENT_RTPORT_HI_WM 3
+
+#define ADSP_AFE_VERSION    0x00200000
+
+/* Size of the range of port IDs for the audio interface. */
+#define  AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE	0xF
+
+/* Size of the range of port IDs for internal BT-FM ports. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE	0x6
+
+/* Size of the range of port IDs for SLIMbus<sup>&reg;
+ * </sup> multichannel
+ * ports.
+ */
+#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE	0xA
+
+/* Size of the range of port IDs for real-time proxy ports. */
+#define  AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE	0x2
+
+/* Size of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE	0x5
+
+/* Start of the range of port IDs for the audio interface. */
+#define  AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START	0x1000
+
+/* End of the range of port IDs for the audio interface. */
+#define  AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \
+	(AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\
+	AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1)
+
+/* Start of the range of port IDs for real-time proxy ports. */
+#define  AFE_PORT_ID_RT_PROXY_PORT_RANGE_START	0x2000
+
+/* End of the range of port IDs for real-time proxy ports. */
+#define  AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \
+	(AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\
+	AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START	0x3000
+
+/* End of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \
+	(AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\
+	AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1)
+
+/*	Start of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_START	0x4000
+
+/*	End of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_END \
+	(AFE_PORT_ID_SLIMBUS_RANGE_START +\
+	AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_START	0x8001
+
+/* End of the range of port IDs for pseudoports.  */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \
+	(AFE_PORT_ID_PSEUDOPORT_RANGE_START +\
+	AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for TDM devices. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_START	0x9000
+
+/* End of the range of port IDs for TDM devices. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_END \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START+0x40-1)
+
+/* Size of the range of port IDs for TDM ports. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_SIZE \
+	(AFE_PORT_ID_TDM_PORT_RANGE_END - \
+	AFE_PORT_ID_TDM_PORT_RANGE_START+1)
+
+#define AFE_PORT_ID_PRIMARY_MI2S_RX         0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX         0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX       0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX       0x1003
+#define AFE_PORT_ID_TERTIARY_MI2S_RX        0x1004
+#define AFE_PORT_ID_TERTIARY_MI2S_TX        0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX      0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX      0x1007
+#define AUDIO_PORT_ID_I2S_RX				0x1008
+#define AFE_PORT_ID_DIGITAL_MIC_TX          0x1009
+#define AFE_PORT_ID_PRIMARY_PCM_RX          0x100A
+#define AFE_PORT_ID_PRIMARY_PCM_TX          0x100B
+#define AFE_PORT_ID_SECONDARY_PCM_RX        0x100C
+#define AFE_PORT_ID_SECONDARY_PCM_TX        0x100D
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX       0x100E
+#define AFE_PORT_ID_SECONDARY_MI2S_RX_SD1	0x1010
+#define AFE_PORT_ID_QUINARY_MI2S_RX		0x1016
+#define AFE_PORT_ID_QUINARY_MI2S_TX		0x1017
+/* ID of the senary MI2S Rx port. */
+#define AFE_PORT_ID_SENARY_MI2S_RX		0x1018
+/* ID of the senary MI2S Tx port. */
+#define AFE_PORT_ID_SENARY_MI2S_TX		0x1019
+#define AFE_PORT_ID_SPDIF_RX                0x5000
+#define  AFE_PORT_ID_RT_PROXY_PORT_001_RX   0x2000
+#define  AFE_PORT_ID_RT_PROXY_PORT_001_TX   0x2001
+#define AFE_PORT_ID_INTERNAL_BT_SCO_RX      0x3000
+#define AFE_PORT_ID_INTERNAL_BT_SCO_TX      0x3001
+#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX     0x3002
+#define AFE_PORT_ID_INTERNAL_FM_RX          0x3004
+#define AFE_PORT_ID_INTERNAL_FM_TX          0x3005
+/* SLIMbus Rx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX      0x4000
+/* SLIMbus Tx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX      0x4001
+/* SLIMbus Rx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX      0x4002
+/* SLIMbus Tx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX      0x4003
+/* SLIMbus Rx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX      0x4004
+/* SLIMbus Tx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX      0x4005
+/* SLIMbus Rx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX      0x4006
+/* SLIMbus Tx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX      0x4007
+/* SLIMbus Rx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX      0x4008
+/* SLIMbus Tx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX      0x4009
+/* SLIMbus Rx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX      0x400a
+/* SLIMbus Tx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX      0x400b
+/* SLIMbus Rx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX      0x400c
+/* SLIMbus Tx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX      0x400d
+
+/* Generic pseudoport 1. */
+#define AFE_PORT_ID_PSEUDOPORT_01      0x8001
+/* Generic pseudoport 2. */
+#define AFE_PORT_ID_PSEUDOPORT_02      0x8002
+
+/* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx}
+	Primary Aux PCM Tx port ID.
+*/
+#define AFE_PORT_ID_PRIMARY_PCM_TX      0x100B
+/* Pseudoport that corresponds to the voice Rx path.
+ * For recording, the voice Rx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_RX	0x8003
+
+/* Pseudoport that corresponds to the voice Tx path.
+ * For recording, the voice Tx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_TX	0x8004
+/* Pseudoport that corresponds to in-call voice delivery samples.
+ * During in-call audio delivery, the audio path delivers samples
+ * to this port from where the voice path delivers them on the
+ * Rx path.
+ */
+#define AFE_PORT_ID_VOICE2_PLAYBACK_TX  0x8002
+#define AFE_PORT_ID_VOICE_PLAYBACK_TX   0x8005
+
+#define AFE_PORT_ID_PRIMARY_TDM_RX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x00)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_1 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_2 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_3 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_4 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_5 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_6 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_7 \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_PRIMARY_TDM_TX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x01)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_1 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_2 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_3 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_4 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_5 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_6 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_7 \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_SECONDARY_TDM_RX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x10)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_1 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_2 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_3 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_4 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_5 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_6 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_7 \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_SECONDARY_TDM_TX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x11)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_1 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_2 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_3 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_4 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_5 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_6 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_7 \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_TERTIARY_TDM_RX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x20)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_1 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_2 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_3 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_4 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_5 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_6 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_7 \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_TERTIARY_TDM_TX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x21)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_1 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_2 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_3 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_4 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_5 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_6 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_7 \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_QUATERNARY_TDM_RX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x30)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_1 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_2 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_3 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_4 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_5 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_6 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_7 \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_QUATERNARY_TDM_TX \
+	(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x31)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_1 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_2 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_3 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_4 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_5 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_6 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_7 \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_INVALID             0xFFFF
+
+#define AAC_ENC_MODE_AAC_LC 0x02
+#define AAC_ENC_MODE_AAC_P 0x05
+#define AAC_ENC_MODE_EAAC_P 0x1D
+
+#define AFE_PSEUDOPORT_CMD_START 0x000100cf
+struct afe_pseudoport_start_command {
+	struct apr_hdr hdr;
+	u16 port_id;		/* Pseudo Port 1 = 0x8000 */
+				/* Pseudo Port 2 = 0x8001 */
+				/* Pseudo Port 3 = 0x8002 */
+	u16 timing;		/* FTRT = 0 , AVTimer = 1, */
+} __packed;
+
+#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
+struct afe_pseudoport_stop_command {
+	struct apr_hdr hdr;
+	u16 port_id;		/* Pseudo Port 1 = 0x8000 */
+				/* Pseudo Port 2 = 0x8001 */
+				/* Pseudo Port 3 = 0x8002 */
+	u16 reserved;
+} __packed;
+
+
+#define AFE_MODULE_SIDETONE_IIR_FILTER	0x00010202
+#define AFE_PARAM_ID_ENABLE	0x00010203
+
+/*  Payload of the #AFE_PARAM_ID_ENABLE
+ * parameter, which enables or
+ * disables any module.
+ * The fixed size of this structure is four bytes.
+ */
+
+struct afe_mod_enable_param {
+	u16                  enable;
+	/* Enables (1) or disables (0) the module. */
+
+	u16                  reserved;
+	/* This field must be set to zero.
+		*/
+} __packed;
+
+/* ID of the configuration parameter used by the
+ * #AFE_MODULE_SIDETONE_IIR_FILTER module.
+ */
+#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG	0x00010204
+
+struct afe_sidetone_iir_filter_config_params {
+	u16                  num_biquad_stages;
+/* Number of stages.
+ * Supported values: Minimum of 5 and maximum of 10
+ */
+
+	u16                  pregain;
+/* Pregain for the compensating filter response.
+ * Supported values: Any number in Q13 format
+ */
+} __packed;
+
+#define AFE_MODULE_LOOPBACK	0x00010205
+#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH	0x00010206
+
+/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter,
+ * which gets/sets loopback gain of a port to an Rx port.
+ * The Tx port ID of the loopback is part of the set_param command.
+ */
+
+/*  Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's
+ * configuration/calibration settings for the AFE port.
+ */
+struct afe_port_cmd_set_param_v2 {
+	u16 port_id;
+/* Port interface and direction (Rx or Tx) to start.
+ */
+
+	u16 payload_size;
+/* Actual size of the payload in bytes.
+ * This is used for parsing the parameter payload.
+ * Supported values: > 0
+ */
+
+u32 payload_address_lsw;
+/* LSW of 64 bit Payload address.
+ * Address should be 32-byte,
+ * 4kbyte aligned and must be contiguous memory.
+ */
+
+u32 payload_address_msw;
+/* MSW of 64 bit Payload address.
+ * In case of 32-bit shared memory address,
+ * this field must be set to zero.
+ * In case of 36-bit shared memory address,
+ * bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned
+ * and must be contiguous memory.
+ */
+
+u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values:
+ * - NULL -- Message. The parameter data is in-band.
+ * - Non-NULL -- The parameter data is Out-band.Pointer to
+ * the physical address
+ * in shared memory of the payload data.
+ * An optional field is available if parameter
+ * data is in-band:
+ * afe_param_data_v2 param_data[...].
+ * For detailed payload content, see the
+ * afe_port_param_data_v2 structure.
+ */
+} __packed;
+
+#define AFE_PORT_CMD_SET_PARAM_V2	0x000100EF
+
+struct afe_port_param_data_v2 {
+	u32 module_id;
+/* ID of the module to be configured.
+ * Supported values: Valid module ID
+ */
+
+u32 param_id;
+/* ID of the parameter corresponding to the supported parameters
+ * for the module ID.
+ * Supported values: Valid parameter ID
+ */
+
+u16 param_size;
+/* Actual size of the data for the
+ * module_id/param_id pair. The size is a
+ * multiple of four bytes.
+ * Supported values: > 0
+ */
+
+u16 reserved;
+/* This field must be set to zero.
+ */
+} __packed;
+
+struct afe_loopback_gain_per_path_param {
+	struct apr_hdr	hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	u16                  rx_port_id;
+/* Rx port of the loopback. */
+
+u16                  gain;
+/* Loopback gain per path of the port.
+ * Supported values: Any number in Q13 format
+ */
+} __packed;
+
+/* Parameter ID used to configure and enable/disable the
+ * loopback path. The difference with respect to the existing
+ * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be
+ * configured as source port in loopback path. Port-id in
+ * AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be
+ * Tx or Rx port. In addition, we can configure the type of
+ * routing mode to handle different use cases.
+ */
+#define AFE_PARAM_ID_LOOPBACK_CONFIG	0x0001020B
+#define AFE_API_VERSION_LOOPBACK_CONFIG	0x1
+
+enum afe_loopback_routing_mode {
+	LB_MODE_DEFAULT = 1,
+	/* Regular loopback from source to destination port */
+	LB_MODE_SIDETONE,
+	/* Sidetone feed from Tx source to Rx destination port */
+	LB_MODE_EC_REF_VOICE_AUDIO,
+	/* Echo canceller reference, voice + audio + DTMF */
+	LB_MODE_EC_REF_VOICE
+	/* Echo canceller reference, voice alone */
+} __packed;
+
+/*  Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG ,
+ * which enables/disables one AFE loopback.
+ */
+struct afe_loopback_cfg_v1 {
+	struct apr_hdr	hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	u32		loopback_cfg_minor_version;
+/* Minor version used for tracking the version of the RMC module
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
+ */
+	u16                  dst_port_id;
+	/* Destination Port Id. */
+	u16                  routing_mode;
+/* Specifies data path type from src to dest port.
+ * Supported values:
+ * #LB_MODE_DEFAULT
+ * #LB_MODE_SIDETONE
+ * #LB_MODE_EC_REF_VOICE_AUDIO
+ * #LB_MODE_EC_REF_VOICE_A
+ * #LB_MODE_EC_REF_VOICE
+ */
+
+	u16                  enable;
+/* Specifies whether to enable (1) or
+ * disable (0) an AFE loopback.
+ */
+	u16                  reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.
+ */
+
+} __packed;
+
+#define AFE_MODULE_SPEAKER_PROTECTION	0x00010209
+#define AFE_PARAM_ID_SPKR_PROT_CONFIG	0x0001020a
+#define AFE_API_VERSION_SPKR_PROT_CONFIG	0x1
+#define AFE_SPKR_PROT_EXCURSIONF_LEN	512
+struct afe_spkr_prot_cfg_param_v1 {
+	u32       spkr_prot_minor_version;
+/*
+ * Minor version used for tracking the version of the
+ * speaker protection module configuration interface.
+ * Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG
+ */
+
+int16_t        win_size;
+/* Analysis and synthesis window size (nWinSize).
+ * Supported values: 1024, 512, 256 samples
+ */
+
+int16_t        margin;
+/* Allowable margin for excursion prediction,
+ * in L16Q15 format. This is a
+ * control parameter to allow
+ * for overestimation of peak excursion.
+ */
+
+int16_t        spkr_exc_limit;
+/* Speaker excursion limit, in L16Q15 format.*/
+
+int16_t        spkr_resonance_freq;
+/* Resonance frequency of the speaker; used
+ * to define a frequency range
+ * for signal modification.
+ *
+ * Supported values: 0 to 2000 Hz */
+
+int16_t        limhresh;
+/* Threshold of the hard limiter; used to
+ * prevent overshooting beyond a
+ * signal level that was set by the limiter
+ * prior to speaker protection.
+ * Supported values: 0 to 32767
+ */
+
+int16_t        hpf_cut_off_freq;
+/* High pass filter cutoff frequency.
+ * Supported values: 100, 200, 300 Hz
+ */
+
+int16_t        hpf_enable;
+/* Specifies whether the high pass filter
+ * is enabled (0) or disabled (1).
+ */
+
+int16_t        reserved;
+/* This field must be set to zero. */
+
+int32_t        amp_gain;
+/* Amplifier gain in L32Q15 format.
+ * This is the RMS voltage at the
+ * loudspeaker when a 0dBFS tone
+ * is played in the digital domain.
+ */
+
+int16_t        excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN];
+/* Array of the excursion transfer function.
+ * The peak excursion of the
+ * loudspeaker diaphragm is
+ * measured in millimeters for 1 Vrms Sine
+ * tone at all FFT bin frequencies.
+ * Supported values: Q15 format
+ */
+} __packed;
+
+
+#define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER	0x000100E0
+
+/*  Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER
+ * command, which registers a real-time port driver
+ * with the AFE service.
+ */
+struct afe_service_cmd_register_rt_port_driver {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Port ID with which the real-time driver exchanges data
+ * (registers for events).
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+	u16                  reserved;
+	/* This field must be set to zero. */
+} __packed;
+
+#define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER	0x000100E1
+
+/*  Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER
+ * command, which unregisters a real-time port driver from
+ * the AFE service.
+ */
+struct afe_service_cmd_unregister_rt_port_driver {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Port ID from which the real-time
+ * driver unregisters for events.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+	u16                  reserved;
+	/* This field must be set to zero.	*/
+} __packed;
+
+#define AFE_EVENT_RT_PROXY_PORT_STATUS	0x00010105
+#define AFE_EVENTYPE_RT_PROXY_PORT_START	0
+#define AFE_EVENTYPE_RT_PROXY_PORT_STOP	1
+#define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK	2
+#define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK	3
+#define AFE_EVENTYPE_RT_PROXY_PORT_INVALID	0xFFFF
+
+/*  Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * message, which sends an event from the AFE service
+ * to a registered client.
+ */
+struct afe_event_rt_proxy_port_status {
+	u16                  port_id;
+/* Port ID to which the event is sent.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+	u16                  eventype;
+/* Type of event.
+ * Supported values:
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_START
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_STOP
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED
+
+struct afe_port_data_cmd_rt_proxy_port_write_v2 {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Tx (mic) proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+	u16                  reserved;
+	/* This field must be set to zero. */
+
+	u32                  buffer_address_lsw;
+/* LSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+	u32                  buffer_address_msw;
+/* MSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+	u32					mem_map_handle;
+/* A memory map handle encapsulating shared memory
+ * attributes is returned if
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS
+ * command is successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+	u32                  available_bytes;
+/* Number of valid bytes available
+ * in the buffer (including all
+ * channels: number of bytes per
+ * channel = availableBytesumChannels).
+ * Supported values: > 0
+ *
+ * This field must be equal to the frame
+ * size specified in the #AFE_PORT_AUDIO_IF_CONFIG
+ * command that was sent to configure this
+ * port.
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2	0x000100EE
+
+/*  Payload of the
+ * #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which
+ * delivers an empty buffer to the AFE service. On
+ * acknowledgment, data is filled in the buffer.
+ */
+struct afe_port_data_cmd_rt_proxy_port_read_v2 {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Rx proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ * (This must be an Rx (speaker) port.)
+ */
+
+	u16                  reserved;
+	/* This field must be set to zero. */
+
+	u32                  buffer_address_lsw;
+/* LSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+
+	u32                  buffer_address_msw;
+/* MSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+		u32				mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+	u32                  available_bytes;
+/* Number of valid bytes available in the buffer (including all
+ * channels).
+ * Supported values: > 0
+ * This field must be equal to the frame size specified in the
+ * #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure
+ * this port.
+ */
+} __packed;
+
+/* This module ID is related to device configuring like I2S,PCM,
+ * HDMI, SLIMBus etc. This module supports follwing parameter ids.
+ * - #AFE_PARAM_ID_I2S_CONFIG
+ * - #AFE_PARAM_ID_PCM_CONFIG
+ * - #AFE_PARAM_ID_DIGI_MIC_CONFIG
+ * - #AFE_PARAM_ID_HDMI_CONFIG
+ * - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * - #AFE_PARAM_ID_SLIMBUS_CONFIG
+ * - #AFE_PARAM_ID_RT_PROXY_CONFIG
+ */
+
+#define AFE_MODULE_AUDIO_DEV_INTERFACE    0x0001020C
+#define AFE_PORT_SAMPLE_RATE_8K           8000
+#define AFE_PORT_SAMPLE_RATE_16K          16000
+#define AFE_PORT_SAMPLE_RATE_48K          48000
+#define AFE_PORT_SAMPLE_RATE_96K          96000
+#define AFE_PORT_SAMPLE_RATE_192K         192000
+#define AFE_LINEAR_PCM_DATA				0x0
+#define AFE_NON_LINEAR_DATA				0x1
+#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2
+#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3
+
+/* This param id is used to configure I2S interface */
+#define AFE_PARAM_ID_I2S_CONFIG	0x0001020D
+#define AFE_API_VERSION_I2S_CONFIG	0x1
+/*	Enumeration for setting the I2S configuration
+ * channel_mode parameter to
+ * serial data wire number 1-3 (SD3).
+ */
+#define AFE_PORT_I2S_SD0                     0x1
+#define AFE_PORT_I2S_SD1                     0x2
+#define AFE_PORT_I2S_SD2                     0x3
+#define AFE_PORT_I2S_SD3                     0x4
+#define AFE_PORT_I2S_QUAD01                  0x5
+#define AFE_PORT_I2S_QUAD23                  0x6
+#define AFE_PORT_I2S_6CHS                    0x7
+#define AFE_PORT_I2S_8CHS                    0x8
+#define AFE_PORT_I2S_MONO                    0x0
+#define AFE_PORT_I2S_STEREO                  0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL  0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL  0x1
+
+/*  Payload of the #AFE_PARAM_ID_I2S_CONFIG
+ * command's (I2S configuration
+ * parameter).
+ */
+struct afe_param_id_i2s_cfg {
+	u32                  i2s_cfg_minor_version;
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+	u16                  channel_mode;
+/* I2S lines and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_I2S_SD0
+ * - #AFE_PORT_I2S_SD1
+ * - #AFE_PORT_I2S_SD2
+ * - #AFE_PORT_I2S_SD3
+ * - #AFE_PORT_I2S_QUAD01
+ * - #AFE_PORT_I2S_QUAD23
+ * - #AFE_PORT_I2S_6CHS
+ * - #AFE_PORT_I2S_8CHS
+ */
+
+	u16                  mono_stereo;
+/* Specifies mono or stereo. This applies only when
+ * a single I2S line is used.
+ * Supported values:
+ * - #AFE_PORT_I2S_MONO
+ * - #AFE_PORT_I2S_STEREO
+ */
+
+	u16                  ws_src;
+/* Word select source: internal or external.
+ * Supported values:
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL
+ */
+
+	u32                  sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+
+	u16					data_format;
+/* data format
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+		u16                  reserved;
+	/* This field must be set to zero. */
+} __packed;
+
+/*
+ * This param id is used to configure PCM interface
+ */
+
+#define AFE_API_VERSION_SPDIF_CONFIG 0x1
+#define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1
+#define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1
+#define AFE_CH_STATUS_A 1
+#define AFE_CH_STATUS_B 2
+
+#define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244
+#define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245
+#define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246
+
+#define AFE_PORT_CLK_ROOT_LPAPLL 0x3
+#define AFE_PORT_CLK_ROOT_LPAQ6PLL   0x4
+
+struct afe_param_id_spdif_cfg {
+/* Minor version used for tracking the version of the SPDIF
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SPDIF_CONFIG
+ */
+	u32	spdif_cfg_minor_version;
+
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_22_05K
+ * - #AFE_PORT_SAMPLE_RATE_32K
+ * - #AFE_PORT_SAMPLE_RATE_44_1K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_176_4K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+	u32	sample_rate;
+
+/* data format
+ * Supported values:
+ * - #AFE_LINEAR_PCM_DATA
+ * - #AFE_NON_LINEAR_DATA
+ */
+	u16	data_format;
+/* Number of channels supported by the port
+ * - PCM - 1, Compressed Case - 2
+ */
+	u16	num_channels;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+	u16	bit_width;
+/* This field must be set to zero. */
+	u16	reserved;
+} __packed;
+
+struct afe_param_id_spdif_ch_status_cfg {
+	u32 ch_status_cfg_minor_version;
+/* Minor version used for tracking the version of channel
+ * status configuration. Current supported version is 1
+ */
+
+	u32 status_type;
+/* Indicate if the channel status is for channel A or B
+ * Supported values:
+ * - #AFE_CH_STATUS_A
+ * - #AFE_CH_STATUS_B
+ */
+
+	u8 status_bits[24];
+/* Channel status - 192 bits for channel
+ * Byte ordering as defined by IEC60958-3
+ */
+
+	u8 status_mask[24];
+/* Channel status with mask bits 1 will be applied.
+ * Byte ordering as defined by IEC60958-3
+ */
+} __packed;
+
+struct afe_param_id_spdif_clk_cfg {
+	u32 clk_cfg_minor_version;
+/* Minor version used for tracking the version of SPDIF
+ * interface clock configuration. Current supported version
+ * is 1
+ */
+
+	u32 clk_value;
+/* Specifies the clock frequency in Hz to set
+ * Supported values:
+ * 0 - Disable the clock
+ * 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2
+ * (channels A and B)
+ */
+
+	u32 clk_root;
+/* Specifies SPDIF root clk source
+ * Supported Values:
+ * - #AFE_PORT_CLK_ROOT_LPAPLL
+ * - #AFE_PORT_CLK_ROOT_LPAQ6PLL
+ */
+} __packed;
+
+struct afe_spdif_clk_config_command {
+	struct apr_hdr                    hdr;
+	struct afe_port_cmd_set_param_v2  param;
+	struct afe_port_param_data_v2     pdata;
+	struct afe_param_id_spdif_clk_cfg clk_cfg;
+} __packed;
+
+struct afe_spdif_chstatus_config_command {
+	struct apr_hdr                    hdr;
+	struct afe_port_cmd_set_param_v2  param;
+	struct afe_port_param_data_v2     pdata;
+	struct afe_param_id_spdif_ch_status_cfg ch_status;
+} __packed;
+
+struct afe_spdif_port_config {
+	struct afe_param_id_spdif_cfg            cfg;
+	struct afe_param_id_spdif_ch_status_cfg  ch_status;
+} __packed;
+
+#define AFE_PARAM_ID_PCM_CONFIG        0x0001020E
+#define AFE_API_VERSION_PCM_CONFIG	0x1
+/* Enumeration for the auxiliary PCM synchronization signal
+ * provided by an external source.
+ */
+
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0
+/*	Enumeration for the auxiliary PCM synchronization signal
+ * provided by an internal source.
+ */
+#define AFE_PORT_PCM_SYNC_SRC_INTERNAL  0x1
+/*	Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use
+ * short synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_PCM  0x0
+/*
+ * Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use long
+ * synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_AUX    0x1
+/*
+ * Enumeration for setting the PCM configuration frame to 8.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_8  0x0
+/*
+ * Enumeration for setting the PCM configuration frame to 16.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_16   0x1
+
+/*	Enumeration for setting the PCM configuration frame to 32.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2
+
+/*	Enumeration for setting the PCM configuration frame to 64.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_64   0x3
+
+/*	Enumeration for setting the PCM configuration frame to 128.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4
+
+/*	Enumeration for setting the PCM configuration frame to 256.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5
+
+/*	Enumeration for setting the PCM configuration
+ * quantype parameter to A-law with no padding.
+ */
+#define AFE_PORT_PCM_ALAW_NOPADDING 0x0
+
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with no padding.
+ */
+#define AFE_PORT_PCM_MULAW_NOPADDING 0x1
+/*	Enumeration for setting the PCM configuration quantype
+ * parameter to linear with no padding.
+ */
+#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2
+/*	Enumeration for setting the PCM configuration quantype
+ * parameter to A-law with padding.
+ */
+#define AFE_PORT_PCM_ALAW_PADDING  0x3
+/*	Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with padding.
+ */
+#define AFE_PORT_PCM_MULAW_PADDING 0x4
+/*	Enumeration for setting the PCM configuration quantype
+ * parameter to linear with padding.
+ */
+#define AFE_PORT_PCM_LINEAR_PADDING 0x5
+/*	Enumeration for disabling the PCM configuration
+ * ctrl_data_out_enable parameter.
+ * The PCM block is the only master.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0
+/*
+ * Enumeration for enabling the PCM configuration
+ * ctrl_data_out_enable parameter. The PCM block shares
+ * the signal with other masters.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE  0x1
+
+/*  Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
+ * (PCM configuration parameter).
+ */
+
+struct afe_param_id_pcm_cfg {
+	u32                  pcm_cfg_minor_version;
+/* Minor version used for tracking the version of the AUX PCM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_PCM_CONFIG
+ */
+
+	u16                  aux_mode;
+/* PCM synchronization setting.
+ * Supported values:
+ * - #AFE_PORT_PCM_AUX_MODE_PCM
+ * - #AFE_PORT_PCM_AUX_MODE_AUX
+ */
+
+	u16                  sync_src;
+/* Synchronization source.
+ * Supported values:
+ * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL
+ * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL
+ */
+
+	u16                  frame_setting;
+/* Number of bits per frame.
+ * Supported values:
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_8
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_16
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_32
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_64
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_128
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_256
+ */
+
+	u16                  quantype;
+/* PCM quantization type.
+ * Supported values:
+ * - #AFE_PORT_PCM_ALAW_NOPADDING
+ * - #AFE_PORT_PCM_MULAW_NOPADDING
+ * - #AFE_PORT_PCM_LINEAR_NOPADDING
+ * - #AFE_PORT_PCM_ALAW_PADDING
+ * - #AFE_PORT_PCM_MULAW_PADDING
+ * - #AFE_PORT_PCM_LINEAR_PADDING
+ */
+
+	u16                  ctrl_data_out_enable;
+/* Specifies whether the PCM block shares the data-out
+ * signal to the drive with other masters.
+ * Supported values:
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE
+ */
+		u16                  reserved;
+	/* This field must be set to zero. */
+
+	u32                  sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+	u16                  num_channels;
+/* Number of channels.
+ * Supported values: 1 to 4
+ */
+
+	u16                  slot_number_mapping[4];
+/* Specifies the slot number for the each channel in
+ * multi channel scenario.
+ * Supported values: 1 to 32
+ */
+} __packed;
+
+/*
+ * This param id is used to configure DIGI MIC interface
+ */
+#define AFE_PARAM_ID_DIGI_MIC_CONFIG	0x0001020F
+/*  This version information is used to handle the new
+ *   additions to the config interface in future in backward
+ *   compatible manner.
+ */
+#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 0.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT0  0x1
+
+/*Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 0.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT0  0x2
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT1  0x3
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 0.
+ */
+#define AFE_PORT_DIGI_MIC_MODE_STEREO0  0x5
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 1.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_STEREO1    0x6
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to quad.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_QUAD     0x7
+
+/*  Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's
+ * (DIGI MIC configuration
+ * parameter).
+ */
+struct afe_param_id_digi_mic_cfg {
+	u32                  digi_mic_cfg_minor_version;
+/* Minor version used for tracking the version of the DIGI Mic
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+	u16                  channel_mode;
+/* Digital mic and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT0
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT0
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT1
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT1
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO0
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO1
+ * - #AFE_PORT_DIGI_MIC_MODE_QUAD
+ */
+
+	u32                  sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+} __packed;
+
+/*
+* This param id is used to configure HDMI interface
+*/
+#define AFE_PARAM_ID_HDMI_CONFIG     0x00010210
+
+/*  This version information is used to handle the new
+*   additions to the config interface in future in backward
+*   compatible manner.
+*/
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command,
+ * which configures a multichannel HDMI audio interface.
+ */
+struct afe_param_id_hdmi_multi_chan_audio_cfg {
+	u32                  hdmi_cfg_minor_version;
+/* Minor version used for tracking the version of the HDMI
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_HDMI_CONFIG
+ */
+
+u16                  datatype;
+/* data type
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+
+u16                  channel_allocation;
+/* HDMI channel allocation information for programming an HDMI
+ * frame. The default is 0 (Stereo).
+ *
+ * This information is defined in the HDMI standard, CEA 861-D
+ * (refer to @xhyperref{S1,[S1]}). The number of channels is also
+ * inferred from this parameter.
+*/
+
+
+u32                  sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - 22050, 44100, 176400 for compressed streams
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+		u16                  reserved;
+	/* This field must be set to zero. */
+} __packed;
+
+/*
+* This param id is used to configure BT or FM(RIVA) interface
+*/
+#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG  0x00010211
+
+/*  This version information is used to handle the new
+*   additions to the config interface in future in backward
+*   compatible manner.
+*/
+#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG	0x1
+
+/*  Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * command's BT voice/BT audio/FM configuration parameter.
+ */
+struct afe_param_id_internal_bt_fm_cfg {
+	u32                  bt_fm_cfg_minor_version;
+/* Minor version used for tracking the version of the BT and FM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG
+ */
+
+	u16                  num_channels;
+/* Number of channels.
+ * Supported values: 1 to 2
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+	u32                  sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP)
+ */
+} __packed;
+
+/* This param id is used to configure SLIMBUS interface using
+ * shared channel approach.
+ */
+
+
+#define AFE_PARAM_ID_SLIMBUS_CONFIG    0x00010212
+
+/*  This version information is used to handle the new
+*   additions to the config interface in future in backward
+*   compatible manner.
+*/
+#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+
+/*	Enumeration for setting SLIMbus device ID 1.
+*/
+#define AFE_SLIMBUS_DEVICE_1           0x0
+
+/*	Enumeration for setting SLIMbus device ID 2.
+*/
+#define AFE_SLIMBUS_DEVICE_2          0x1
+
+/*	Enumeration for setting the SLIMbus data formats.
+*/
+#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0
+
+/* Enumeration for setting the maximum number of streams per
+ * device.
+ */
+
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT	0x8
+
+/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus
+ * port configuration parameter.
+ */
+
+struct afe_param_id_slimbus_cfg {
+	u32                  sb_cfg_minor_version;
+/* Minor version used for tracking the version of the SLIMBUS
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
+ */
+
+	u16                  slimbus_dev_id;
+/* SLIMbus hardware device ID, which is required to handle
+ * multiple SLIMbus hardware blocks.
+ * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
+ */
+
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+	u16                  data_format;
+/* Data format supported by the SLIMbus hardware. The default is
+ * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
+ * hardware does not perform any format conversions before the data
+ * transfer.
+ */
+
+
+	u16                  num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+	u8  shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+/* Mapping of shared channel IDs (128 to 255) to which the
+ * master port is to be connected.
+ * Shared_channel_mapping[i] represents the shared channel assigned
+ * for audio channel i in multichannel audio data.
+ */
+
+	u32              sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+} __packed;
+
+/*
+* This param id is used to configure Real Time Proxy interface.
+*/
+#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213
+
+/*  This version information is used to handle the new
+*   additions to the config interface in future in backward
+*   compatible manner.
+*/
+#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1
+
+/*  Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG
+ * command (real-time proxy port configuration parameter).
+ */
+struct afe_param_id_rt_proxy_port_cfg {
+	u32                  rt_proxy_cfg_minor_version;
+/* Minor version used for tracking the version of rt-proxy
+ * config interface.
+ */
+
+	u16                  bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+	u16                  interleaved;
+/* Specifies whether the data exchanged between the AFE
+ * interface and real-time port is interleaved.
+ * Supported values: - 0 -- Non-interleaved (samples from each
+ * channel are contiguous in the buffer) - 1 -- Interleaved
+ * (corresponding samples from each input channel are interleaved
+ * within the buffer)
+ */
+
+
+	u16                  frame_size;
+ /* Size of the frames that are used for PCM exchanges with this
+ * port.
+ * Supported values: > 0, in bytes
+ * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples
+ * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320
+ * bytes.
+ */
+	u16                  jitter_allowance;
+/* Configures the amount of jitter that the port will allow.
+ * Supported values: > 0
+ * For example, if +/-10 ms of jitter is anticipated in the timing
+ * of sending frames to the port, and the configuration is 16 kHz
+ * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2
+ * bytes/sample = 320.
+ */
+
+	u16                  low_water_mark;
+/* Low watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any low watermark events
+ * - > 0 -- Low watermark for triggering an event
+ * If the number of bytes in an internal circular buffer is lower
+ * than this low_water_mark parameter, a LOW_WATER_MARK event is
+ * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * Use of watermark events is optional for debugging purposes.
+ */
+
+	u16                  high_water_mark;
+/* High watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any high watermark events
+ * - > 0 -- High watermark for triggering an event
+ * If the number of bytes in an internal circular buffer exceeds
+ * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event
+ * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * The use of watermark events is optional and for debugging
+ * purposes.
+ */
+
+
+	u32					sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+
+	u16                  num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+	u16                  reserved;
+	/* For 32 bit alignment. */
+} __packed;
+
+
+/* This param id is used to configure the Pseudoport interface */
+
+#define AFE_PARAM_ID_PSEUDO_PORT_CONFIG	0x00010219
+
+/* Version information used to handle future additions to the configuration
+ * interface (for backward compatibility).
+ */
+#define AFE_API_VERSION_PSEUDO_PORT_CONFIG                          0x1
+
+/* Enumeration for setting the timing_mode parameter to faster than real
+ * time.
+ */
+#define AFE_PSEUDOPORT_TIMING_MODE_FTRT                             0x0
+
+/* Enumeration for setting the timing_mode parameter to real time using
+ * timers.
+ */
+#define AFE_PSEUDOPORT_TIMING_MODE_TIMER                            0x1
+
+/* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by
+    AFE_MODULE_AUDIO_DEV_INTERFACE.
+*/
+struct afe_param_id_pseudo_port_cfg {
+	u32                  pseud_port_cfg_minor_version;
+	/*
+	 * Minor version used for tracking the version of the pseudoport
+	 * configuration interface.
+	 */
+
+	u16                  bit_width;
+	/* Bit width of the sample at values 16, 24 */
+
+	u16                  num_channels;
+	/* Number of channels at values  1 to 8 */
+
+	u16                  data_format;
+	/* Non-linear data format supported by the pseudoport (for future use).
+	 * At values #AFE_LINEAR_PCM_DATA
+	 */
+
+	u16                  timing_mode;
+	/* Indicates whether the pseudoport synchronizes to the clock or
+	 * operates faster than real time.
+	 * at values
+	 * - #AFE_PSEUDOPORT_TIMING_MODE_FTRT
+	 * - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend
+	 */
+
+	u32                  sample_rate;
+	/* Sample rate at which the pseudoport will run.
+	 * at values
+	 * - #AFE_PORT_SAMPLE_RATE_8K
+	 * - #AFE_PORT_SAMPLE_RATE_32K
+	 * - #AFE_PORT_SAMPLE_RATE_48K
+	 * - #AFE_PORT_SAMPLE_RATE_96K
+	 * - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend
+	 */
+} __packed;
+
+#define AFE_PARAM_ID_TDM_CONFIG		0x0001029D
+
+#define AFE_API_VERSION_TDM_CONFIG              1
+
+#define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE        0
+#define AFE_PORT_TDM_LONG_SYNC_MODE             1
+#define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE       2
+
+#define AFE_PORT_TDM_SYNC_SRC_EXTERNAL          0
+#define AFE_PORT_TDM_SYNC_SRC_INTERNAL          1
+
+#define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE       0
+#define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE        1
+
+#define AFE_PORT_TDM_SYNC_NORMAL                0
+#define AFE_PORT_TDM_SYNC_INVERT                1
+
+#define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE    0
+#define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE    1
+#define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE    2
+
+/* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by
+    AFE_MODULE_AUDIO_DEV_INTERFACE.
+*/
+struct afe_param_id_tdm_cfg {
+	u32	tdm_cfg_minor_version;
+	/**< Minor version used to track TDM configuration.
+	@values #AFE_API_VERSION_TDM_CONFIG */
+
+	u32	num_channels;
+	/**< Number of enabled slots for TDM frame.
+	@values 1 to 8 */
+
+	u32	sample_rate;
+	/**< Sampling rate of the port.
+	@values
+	- #AFE_PORT_SAMPLE_RATE_8K
+	- #AFE_PORT_SAMPLE_RATE_16K
+	- #AFE_PORT_SAMPLE_RATE_24K
+	- #AFE_PORT_SAMPLE_RATE_32K
+	- #AFE_PORT_SAMPLE_RATE_48K @tablebulletend */
+
+	u32	bit_width;
+	/**< Bit width of the sample.
+	@values 16, 24 */
+
+	u16	data_format;
+	/**< Data format: linear and compressed
+
+	@values
+	- #AFE_LINEAR_PCM_DATA
+	- #AFE_NON_LINEAR_DATA @tablebulletend */
+
+	u16	sync_mode;
+	/**< TDM synchronization setting.
+	@values (short, long, slot) sync mode
+	- #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE
+	- #AFE_PORT_TDM_LONG_SYNC_MODE
+	- #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend */
+
+	u16	sync_src;
+	/**< Synchronization source.
+	@values
+	- #AFE_PORT_TDM_SYNC_SRC_EXTERNAL
+	- #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend */
+
+	u16	nslots_per_frame;
+	/**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
+	@values 1 - 32 */
+
+	u16	ctrl_data_out_enable;
+	/**< Specifies whether the TDM block shares the data-out signal to the
+	drive with other masters.
+	@values
+	- #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE
+	- #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend */
+
+	u16	ctrl_invert_sync_pulse;
+	/**< Specifies whether to invert the sync or not.
+	@values
+	- #AFE_PORT_TDM_SYNC_NORMAL
+	- #AFE_PORT_TDM_SYNC_INVERT @tablebulletend */
+
+	u16	ctrl_sync_data_delay;
+	/**< Specifies the number of bit clock to delay data with respect to
+	sync edge.
+	@values
+	- #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE
+	- #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE
+	- #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend */
+
+	u16	slot_width;
+	/**< Slot width of the slot in a TDM frame.  (slot_width >= bit_width)
+	have to be satisfied.
+	@values 16, 24, 32 */
+
+	u32	slot_mask;
+	/**< Position of active slots.  When that bit is set,
+	that paricular slot is active.
+	Number of active slots can be inferred by number of
+	bits set in the mask.  Only 8 individual bits can be enabled.
+	Bits 0..31 corresponding to slot 0..31
+	@values 1 to 2^32 - 1 */
+} __packed;
+
+/** ID of Time Divsion Multiplexing (TDM) module,
+	which is used for configuring the AFE TDM.
+
+	This module supports following parameter IDs:
+	- #AFE_PORT_TDM_SLOT_CONFIG
+
+	To configure the TDM interface, the client must use the
+	#AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the
+	respective parameter IDs as listed above.
+*/
+
+#define AFE_MODULE_TDM		0x0001028A
+
+/** ID of the parameter used by #AFE_MODULE_TDM to configure
+	the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
+*/
+#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG	0x00010297
+
+/** Version information used to handle future additions to slot mapping
+	configuration (for backward compatibility).
+*/
+#define AFE_API_VERSION_SLOT_MAPPING_CONFIG	0x1
+
+/** Data align type  */
+#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB		0
+#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB		1
+
+#define AFE_SLOT_MAPPING_OFFSET_INVALID		0xFFFF
+
+/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG
+	command's TDM configuration parameter.
+*/
+struct afe_param_id_slot_mapping_cfg {
+	u32	minor_version;
+	/**< Minor version used for tracking TDM slot configuration.
+	  @values #AFE_API_VERSION_TDM_SLOT_CONFIG */
+
+	u16	num_channel;
+	/**< number of channel of the audio sample.
+	@values 1, 2, 4, 6, 8 @tablebulletend */
+
+	u16	bitwidth;
+	/**< Slot bit width for each channel
+	@values 16, 24, 32 */
+
+	u32	data_align_type;
+	/**< indicate how data packed from slot_offset for 32 slot bit width
+	in case of sample bit width is 24.
+	@values
+	#AFE_SLOT_MAPPING_DATA_ALIGN_MSB
+	#AFE_SLOT_MAPPING_DATA_ALIGN_LSB  */
+
+	u16	offset[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+	/**< Array of the slot mapping start offset in bytes for this frame.
+	The bytes is counted from 0. The 0 is mapped to the 1st byte
+	in or out of the digital serial data line this sub-frame belong to.
+	slot_offset[] setting is per-channel based.
+	The max num of channel supported is 8.
+	The valid offset value must always be continuly placed in from index 0.
+	Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays.
+	If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24,
+	"data_align_type" is used to indicate how 24 bit sample data in aligning
+	with 32 bit slot width per-channel.
+	@values, in byte*/
+} __packed;
+
+/** ID of the parameter used by #AFE_MODULE_TDM to configure
+    the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
+*/
+#define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG		0x00010298
+
+/** Version information used to handle future additions to custom TDM header
+	configuration (for backward compatibility).
+*/
+#define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG	0x1
+
+#define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID		0x0
+#define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT		0x1
+#define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST	0x2
+
+#define AFE_CUSTOM_TDM_HEADER_MAX_CNT	0x8
+
+/** Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID
+*/
+struct afe_param_id_custom_tdm_header_cfg {
+	u32	minor_version;
+	/**< Minor version used for tracking custom TDM header configuration.
+	@values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG */
+
+	u16	start_offset;
+	/**< the slot mapping start offset in bytes from this sub-frame
+	The bytes is counted from 0. The 0 is mapped to the 1st byte in or out of
+	the digital serial data line this sub-frame belong to.
+	@values, in byte,
+	supported values are 0, 4, 8,    */
+
+	u16	header_width;
+	/**< the header width per-frame followed.
+	2 bytes for MOST/TDM case
+	@values, in byte
+	supported value is 2 */
+
+	u16	header_type;
+	/**< Indicate what kind of custom TDM header it is.
+	@values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0
+	#AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1  (for AAN channel per MOST)
+	#AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2
+	(for entertainment channel, which will overwrite
+	AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST) */
+
+	u16	num_frame_repeat;
+	/**< num of header followed.
+	@values, supported value is 8*/
+	u16	header[AFE_CUSTOM_TDM_HEADER_MAX_CNT];
+	/** < SAD header for MOST/TDM case is followed as payload as below.
+	The size of followed SAD header in bytes is num_of_frame_repeat * header_width_per_frame
+	which is 2 * 8 = 16 bytes here.
+	the supported payload format is in uint16_t as below
+	uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00
+	uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218
+	uint16_t header2; DTCP Info     Info[6] - unused -> 0x0
+	uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0
+	uint16_t header4; Reserved Info[0] - Num of bytes following  -> 0x7
+	uint16_t header5; Reserved Info[1] - Media Type -> 0x0
+	uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0
+	uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low  Byte -> 0x0 */
+} __packed;
+
+struct afe_slot_mapping_config_command {
+	struct apr_hdr	hdr;
+	struct afe_port_cmd_set_param_v2	param;
+	struct afe_port_param_data_v2	pdata;
+	struct afe_param_id_slot_mapping_cfg	slot_mapping;
+} __packed;
+
+struct afe_custom_tdm_header_config_command {
+	struct apr_hdr	hdr;
+	struct afe_port_cmd_set_param_v2	param;
+	struct afe_port_param_data_v2	pdata;
+	struct afe_param_id_custom_tdm_header_cfg	custom_tdm_header;
+} __packed;
+
+struct afe_tdm_port_config {
+	struct afe_param_id_tdm_cfg				tdm;
+	struct afe_param_id_slot_mapping_cfg		slot_mapping;
+	struct afe_param_id_custom_tdm_header_cfg	custom_tdm_header;
+} __packed;
+
+#define AFE_PARAM_ID_DEVICE_HW_DELAY     0x00010243
+#define AFE_API_VERSION_DEVICE_HW_DELAY  0x1
+
+struct afe_param_id_device_hw_delay_cfg {
+	uint32_t    device_hw_delay_minor_version;
+	uint32_t    delay_in_us;
+} __packed;
+
+#define AFE_PARAM_ID_SET_TOPOLOGY    0x0001025A
+#define AFE_API_VERSION_TOPOLOGY_V1 0x1
+
+struct afe_param_id_set_topology_cfg {
+	/*
+	 * Minor version used for tracking afe topology id configuration.
+	 * @values #AFE_API_VERSION_TOPOLOGY_V1
+	 */
+	u32		minor_version;
+	/*
+	 * Id of the topology for the afe session.
+	 * @values Any valid AFE topology ID
+	 */
+	u32		topology_id;
+} __packed;
+
+union afe_port_config {
+	struct afe_param_id_pcm_cfg               pcm;
+	struct afe_param_id_i2s_cfg               i2s;
+	struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
+	struct afe_param_id_slimbus_cfg           slim_sch;
+	struct afe_param_id_rt_proxy_port_cfg     rtproxy;
+	struct afe_param_id_internal_bt_fm_cfg    int_bt_fm;
+	struct afe_param_id_pseudo_port_cfg       pseudo_port;
+	struct afe_param_id_device_hw_delay_cfg   hw_delay;
+	struct afe_param_id_spdif_cfg             spdif;
+	struct afe_param_id_set_topology_cfg      topology;
+	struct afe_param_id_tdm_cfg               tdm;
+} __packed;
+
+struct afe_audioif_config_command_no_payload {
+	struct apr_hdr			hdr;
+	struct afe_port_cmd_set_param_v2 param;
+} __packed;
+
+struct afe_audioif_config_command {
+	struct apr_hdr			hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	union afe_port_config            port;
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+
+/*  Payload of the #AFE_PORT_CMD_DEVICE_START.*/
+struct afe_port_cmd_device_start {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+
+	u16                  reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_STOP  0x000100E6
+
+/*  Payload of the #AFE_PORT_CMD_DEVICE_STOP.
+*/
+struct afe_port_cmd_device_stop {
+	struct apr_hdr hdr;
+	u16                  port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+	u16                  reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+} __packed;
+
+#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA
+
+/*  Memory map regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS .
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of afe_service_shared_map_region_payload.
+ */
+struct afe_service_cmd_shared_mem_map_regions {
+	struct apr_hdr hdr;
+u16                  mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ * Supported values:
+ * - #ADSP_MEMORY_MAP_EBI_POOL
+ * - #ADSP_MEMORY_MAP_SMI_POOL
+ * - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL
+ * - Other values are reserved
+ *
+ * The memory pool ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory
+ * ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory
+ * ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte
+ * addressable, and 4 KB aligned.
+ */
+
+
+	u16                  num_regions;
+/* Number of regions to map.
+ * Supported values:
+ * - Any value greater than zero
+ */
+
+	u32                  property_flag;
+/* Configures one common property for all the regions in the
+ * payload.
+ *
+ * Supported values: - 0x00000000 to 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 Shared memory
+ * address provided in afe_service_shared_map_region_payloadis a
+ * physical address. The shared memory needs to be mapped( hardware
+ * TLB entry) and a software entry needs to be added for internal
+ * book keeping.
+ *
+ * 1 Shared memory address provided in
+ * afe_service_shared_map_region_payloadis a virtual address. The
+ * shared memory must not be mapped (since hardware TLB entry is
+ * already available) but a software entry needs to be added for
+ * internal book keeping. This can be useful if two services with in
+ * ADSP is communicating via APR. They can now directly communicate
+ * via the Virtual address instead of Physical address. The virtual
+ * regions must be contiguous. num_regions must be 1 in this case.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+
+} __packed;
+/*  Map region payload used by the
+ * afe_service_shared_map_region_payloadstructure.
+ */
+struct afe_service_shared_map_region_payload {
+	u32                  shm_addr_lsw;
+/* least significant word of starting address in the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ * Supported values: - Any 32 bit value
+ */
+
+
+	u32                  shm_addr_msw;
+/* most significant word of startng address in the memory region
+ * to map. For 32 bit shared memory address, this field must be set
+ * to zero. For 36 bit shared memory address, bit31 to bit 4 must be
+ * set to zero
+ *
+ * Supported values: - For 32 bit shared memory address, this field
+ * must be set to zero. - For 36 bit shared memory address, bit31 to
+ * bit 4 must be set to zero - For 64 bit shared memory address, any
+ * 32 bit value
+ */
+
+
+	u32                  mem_size_bytes;
+/* Number of bytes in the region. The aDSP will always map the
+ * regions as virtual contiguous memory, but the memory size must be
+ * in multiples of 4 KB to avoid gaps in the virtually contiguous
+ * mapped memory.
+ *
+ * Supported values: - multiples of 4KB
+ */
+
+} __packed;
+
+#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB
+struct afe_service_cmdrsp_shared_mem_map_regions {
+	u32                  mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful. In the case of failure , a generic APR error response
+ * is returned to the client.
+ *
+ * Supported Values: - Any 32 bit value
+ */
+
+} __packed;
+#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC
+/* Memory unmap regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS
+ *
+ * This structure allows clients to unmap multiple shared memory
+ * regions in a single command.
+ */
+
+
+struct afe_service_cmd_shared_mem_unmap_regions {
+	struct apr_hdr hdr;
+u32                  mem_map_handle;
+/* memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands
+ *
+ * Supported Values:
+ * - Any 32 bit value
+ */
+} __packed;
+
+#define  AFE_PORT_CMD_GET_PARAM_V2 0x000100F0
+
+/*  Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command,
+ * which queries for one post/preprocessing parameter of a
+ * stream.
+ */
+struct afe_port_cmd_get_param_v2 {
+	u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. */
+
+	u16 payload_size;
+/* Maximum data size of the parameter ID/module ID combination.
+ * This is a multiple of four bytes
+ * Supported values: > 0
+ */
+
+	u32 payload_address_lsw;
+/* LSW of 64 bit Payload address. Address should be 32-byte,
+ * 4kbyte aligned and must be contig memory.
+ */
+
+
+	u32 payload_address_msw;
+/* MSW of 64 bit Payload address. In case of 32-bit shared
+ * memory address, this field must be set to zero. In case of 36-bit
+ * shared memory address, bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned and must be contiguous
+ * memory.
+ */
+
+	u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values: - NULL -- Message. The parameter data is
+ * in-band. - Non-NULL -- The parameter data is Out-band.Pointer to
+ * - the physical address in shared memory of the payload data.
+ * For detailed payload content, see the afe_port_param_data_v2
+ * structure
+ */
+
+
+	u32 module_id;
+/* ID of the module to be queried.
+ * Supported values: Valid module ID
+ */
+
+	u32 param_id;
+/* ID of the parameter to be queried.
+ * Supported values: Valid parameter ID
+ */
+} __packed;
+
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+
+/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which
+ * responds to an #AFE_PORT_CMD_GET_PARAM_V2 command.
+ *
+ * Immediately following this structure is the parameters structure
+ * (afe_port_param_data) containing the response(acknowledgment)
+ * parameter payload. This payload is included for an in-band
+ * scenario. For an address/shared memory-based set parameter, this
+ * payload is not needed.
+ */
+
+
+struct afe_port_cmdrsp_get_param_v2 {
+	u32                  status;
+} __packed;
+
+#define AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG	0x0001028C
+#define AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG	0x1
+/*
+ * Payload of the AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+*/
+struct afe_param_id_lpass_core_shared_clk_cfg {
+	u32	lpass_core_shared_clk_cfg_minor_version;
+/*
+ * Minor version used for lpass core shared clock configuration
+ * Supported value: AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG
+ */
+	u32	enable;
+/*
+ * Specifies whether the lpass core shared clock is
+ * enabled (1) or disabled (0).
+ */
+} __packed;
+
+struct afe_lpass_core_shared_clk_config_command {
+	struct apr_hdr		   hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	struct afe_param_id_lpass_core_shared_clk_cfg clk_cfg;
+} __packed;
+
+/* adsp_afe_service_commands.h */
+
+#define ADSP_MEMORY_MAP_EBI_POOL      0
+
+#define ADSP_MEMORY_MAP_SMI_POOL      1
+#define ADSP_MEMORY_MAP_IMEM_POOL      2
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL      3
+/*
+* Definition of virtual memory flag
+*/
+#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1
+
+/*
+* Definition of physical memory flag
+*/
+#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0
+
+#define NULL_POPP_TOPOLOGY				0x00010C68
+#define NULL_COPP_TOPOLOGY				0x00010312
+#define DEFAULT_COPP_TOPOLOGY				0x00010314
+#define DEFAULT_POPP_TOPOLOGY				0x00010BE4
+#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY         0x0001076B
+#define VPM_TX_SM_ECNS_COPP_TOPOLOGY			0x00010F71
+#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY			0x00010F72
+#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY		0x00010F75
+#define VPM_TX_DM_RFECNS_COPP_TOPOLOGY			0x00010F86
+#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_DTS_HPX		0x10015002
+#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_AUDIOSPHERE	0x10028000
+
+/* Memory map regions command payload used by the
+ * #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * commands.
+ *
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of avs_shared_map_region_payload.
+ */
+
+
+struct avs_cmd_shared_mem_map_regions {
+	struct apr_hdr hdr;
+	u16                  mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ *
+ * Supported values: - #ADSP_MEMORY_MAP_EBI_POOL -
+ * #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL
+ * (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values
+ * are reserved
+ *
+ * The memory ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned.
+ */
+
+
+	u16                  num_regions;
+	/* Number of regions to map.*/
+
+	u32                  property_flag;
+/* Configures one common property for all the regions in the
+ * payload. No two regions in the same memory map regions cmd can
+ * have differnt property. Supported values: - 0x00000000 to
+ * 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 shared memory
+ * address provided in avs_shared_map_regions_payload is physical
+ * address. The shared memory needs to be mapped( hardware TLB
+ * entry)
+ *
+ * and a software entry needs to be added for internal book keeping.
+ *
+ * 1 Shared memory address provided in MayPayload[usRegions] is
+ * virtual address. The shared memory must not be mapped (since
+ * hardware TLB entry is already available) but a software entry
+ * needs to be added for internal book keeping. This can be useful
+ * if two services with in ADSP is communicating via APR. They can
+ * now directly communicate via the Virtual address instead of
+ * Physical address. The virtual regions must be contiguous.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+} __packed;
+
+struct avs_shared_map_region_payload {
+	u32                  shm_addr_lsw;
+/* least significant word of shared memory address of the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ */
+
+	u32                  shm_addr_msw;
+/* most significant word of shared memory address of the memory
+ * region to map. For 32 bit shared memory address, this field must
+ * tbe set to zero. For 36 bit shared memory address, bit31 to bit 4
+ * must be set to zero
+ */
+
+	u32                  mem_size_bytes;
+/* Number of bytes in the region.
+ *
+ * The aDSP will always map the regions as virtual contiguous
+ * memory, but the memory size must be in multiples of 4 KB to avoid
+ * gaps in the virtually contiguous mapped memory.
+ */
+
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+	struct apr_hdr       hdr;
+	u32                  mem_map_handle;
+/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS
+ * , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands
+ */
+
+} __packed;
+
+/* Memory map command response payload used by the
+ * #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ * ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ */
+
+
+struct avs_cmdrsp_shared_mem_map_regions {
+	u32                  mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned
+ */
+
+} __packed;
+
+/*adsp_audio_memmap_api.h*/
+
+/* ASM related data structures */
+struct asm_wma_cfg {
+	u16 format_tag;
+	u16 ch_cfg;
+	u32 sample_rate;
+	u32 avg_bytes_per_sec;
+	u16 block_align;
+	u16 valid_bits_per_sample;
+	u32 ch_mask;
+	u16 encode_opt;
+	u16 adv_encode_opt;
+	u32 adv_encode_opt2;
+	u32 drc_peak_ref;
+	u32 drc_peak_target;
+	u32 drc_ave_ref;
+	u32 drc_ave_target;
+} __packed;
+
+struct asm_wmapro_cfg {
+	u16 format_tag;
+	u16 ch_cfg;
+	u32 sample_rate;
+	u32 avg_bytes_per_sec;
+	u16 block_align;
+	u16 valid_bits_per_sample;
+	u32 ch_mask;
+	u16 encode_opt;
+	u16 adv_encode_opt;
+	u32 adv_encode_opt2;
+	u32 drc_peak_ref;
+	u32 drc_peak_target;
+	u32 drc_ave_ref;
+	u32 drc_ave_target;
+} __packed;
+
+struct asm_aac_cfg {
+	u16 format;
+	u16 aot;
+	u16 ep_config;
+	u16 section_data_resilience;
+	u16 scalefactor_data_resilience;
+	u16 spectral_data_resilience;
+	u16 ch_cfg;
+	u16 reserved;
+	u32 sample_rate;
+} __packed;
+
+struct asm_amrwbplus_cfg {
+	u32  size_bytes;
+	u32  version;
+	u32  num_channels;
+	u32  amr_band_mode;
+	u32  amr_dtx_mode;
+	u32  amr_frame_fmt;
+	u32  amr_lsf_idx;
+} __packed;
+
+struct asm_flac_cfg {
+	u32 sample_rate;
+	u32 ext_sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+	u16 stream_info_present;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 ch_cfg;
+	u16 sample_size;
+	u16 md5_sum;
+};
+
+struct asm_alac_cfg {
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+};
+
+struct asm_vorbis_cfg {
+	u32 bit_stream_fmt;
+};
+
+struct asm_ape_cfg {
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+};
+
+struct asm_softpause_params {
+	u32 enable;
+	u32 period;
+	u32 step;
+	u32 rampingcurve;
+} __packed;
+
+struct asm_softvolume_params {
+	u32 period;
+	u32 step;
+	u32 rampingcurve;
+} __packed;
+
+#define ASM_END_POINT_DEVICE_MATRIX     0
+
+#define PCM_CHANNEL_NULL 0
+
+/* Front left channel. */
+#define PCM_CHANNEL_FL    1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR    2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC    3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS   4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS   5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE  6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS   7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB   8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB   9
+
+/* Top surround channel. */
+#define PCM_CHANNELS   10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH  11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS   12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC  13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC  14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC  15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC  16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL  8
+
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+
+#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
+
+#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
+
+#define ASM_MAX_EQ_BANDS 12
+
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+
+struct asm_data_cmd_media_fmt_update_v2 {
+u32                    fmt_blk_size;
+	/* Media format block size in bytes.*/
+}  __packed;
+
+struct asm_multi_channel_pcm_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+	u16  num_channels;
+	/* Number of channels. Supported values: 1 to 8 */
+	u16  bits_per_sample;
+/* Number of bits per sample per channel. * Supported values:
+ * 16, 24 * When used for playback, the client must send 24-bit
+ * samples packed in 32-bit words. The 24-bit samples must be placed
+ * in the most significant 24 bits of the 32-bit word. When used for
+ * recording, the aDSP sends 24-bit samples packed in 32-bit words.
+ * The 24-bit samples are placed in the most significant 24 bits of
+ * the 32-bit word.
+ */
+
+
+	u32  sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 2000 to 48000
+ */
+
+	u16  is_signed;
+	/* Flag that indicates the samples are signed (1). */
+
+	u16  reserved;
+	/* reserved field for 32 bit alignment. must be set to zero. */
+
+	u8   channel_mapping[8];
+/* Channel array of size 8.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ *
+ * Channel[i] mapping describes channel I. Each element i of the
+ * array describes channel I inside the buffer where 0 @le I <
+ * num_channels. An unused channel is set to zero.
+ */
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+	u32                  param_id;
+	/* ID of the parameter. */
+
+	u32                  param_size;
+/* Data size of this parameter, in bytes. The size is a multiple
+ * of 4 bytes.
+ */
+
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+	u32                  frames_per_buf;
+/* Number of encoded frames to pack into each buffer.
+ *
+ * @note1hang This is only guidance information for the aDSP. The
+ * number of encoded frames put into each buffer (specified by the
+ * client) is less than or equal to this number.
+ */
+
+	u32                  enc_cfg_blk_size;
+/* Size in bytes of the encoder configuration block that follows
+ * this member.
+ */
+
+} __packed;
+
+/* @brief Dolby Digital Plus end point configuration structure
+ */
+struct asm_dec_ddp_endp_param_v2 {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	int endp_param_value;
+} __packed;
+
+/* @brief Multichannel PCM encoder configuration structure used
+ * in the #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	uint16_t  num_channels;
+/*< Number of PCM channels.
+ *
+ * Supported values: - 0 -- Native mode - 1 -- 8 Native mode
+ * indicates that encoding must be performed with the number of
+ * channels at the input.
+ */
+
+	uint16_t  bits_per_sample;
+/*< Number of bits per sample per channel.
+ * Supported values: 16, 24
+ */
+
+	uint32_t  sample_rate;
+/*< Number of samples per second (in Hertz).
+ *
+ * Supported values: 0, 8000 to 48000 A value of 0 indicates the
+ * native sampling rate. Encoding is performed at the input sampling
+ * rate.
+ */
+
+	uint16_t  is_signed;
+/*< Specifies whether the samples are signed (1). Currently,
+ * only signed samples are supported.
+ */
+
+	uint16_t  reserved;
+/*< reserved field for 32 bit alignment. must be set to zero.*/
+
+
+	uint8_t   channel_mapping[8];
+} __packed;
+
+#define ASM_MEDIA_FMT_MP3 0x00010BE9
+#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6
+
+/* @xreflabel
+ * {hdr:AsmMediaFmtDolbyAac} Media format ID for the
+ * Dolby AAC decoder. This format ID is be used if the client wants
+ * to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC
+ * contents.
+ */
+
+#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86
+
+/* Enumeration for the audio data transport stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
+
+/* Enumeration for low overhead audio stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS                      1
+
+/* Enumeration for the audio data interchange format
+ * AAC format.
+ */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF   2
+
+/* Enumeration for the raw AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW    3
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC             2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR            5
+#define ASM_MEDIA_FMT_AAC_AOT_PS             29
+#define ASM_MEDIA_FMT_AAC_AOT_BSAC           22
+
+struct asm_aac_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+		u16          aac_fmt_flag;
+/* Bitstream format option.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+
+	u16          audio_objype;
+/* Audio Object Type (AOT) present in the AAC stream.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_BSAC
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ * - Otherwise -- Not supported
+ */
+
+	u16          channel_config;
+/* Number of channels present in the AAC stream.
+ * Supported values:
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - 6 -- 5.1 content
+ */
+
+	u16          total_size_of_PCE_bits;
+/* greater or equal to zero. * -In case of RAW formats and
+ * channel config = 0 (PCE), client can send * the bit stream
+ * containing PCE immediately following this structure * (in-band).
+ * -This number does not include bits included for 32 bit alignment.
+ * -If zero, then the PCE info is assumed to be available in the
+ * audio -bit stream & not in-band.
+ */
+
+	u32          sample_rate;
+/* Number of samples per second (in Hertz).
+ *
+ * Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000,
+ * 44100, 48000
+ *
+ * This field must be equal to the sample rate of the AAC-LC
+ * decoder's output. - For MP4 or 3GP containers, this is indicated
+ * by the samplingFrequencyIndex field in the AudioSpecificConfig
+ * element. - For ADTS format, this is indicated by the
+ * samplingFrequencyIndex in the ADTS fixed header. - For ADIF
+ * format, this is indicated by the samplingFrequencyIndex in the
+ * program_config_element present in the ADIF header.
+ */
+
+} __packed;
+
+struct asm_aac_enc_cfg_v2 {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+
+	u32          bit_rate;
+	/* Encoding rate in bits per second. */
+	u32          enc_mode;
+/* Encoding mode.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ */
+	u16          aac_fmt_flag;
+/* AAC format flag.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+	u16          channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ * @note1hang The eAAC+ encoder mode supports only stereo.
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+	u32          sample_rate;
+/* Number of samples per second.
+ * Supported values: - 0 -- Native mode - For other values,
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+} __packed;
+
+struct asm_vorbis_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+	u32          bit_stream_fmt;
+/* Bit stream format.
+ * Supported values:
+ * - 0 -- Raw bitstream
+ * - 1 -- Transcoded bitstream
+ *
+ * Transcoded bitstream containing the size of the frame as the first
+ * word in each frame.
+ */
+
+} __packed;
+
+struct asm_flac_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+	u16 is_stream_info_present;
+/* Specifies whether stream information is present in the FLAC format
+ * block.
+ *
+ * Supported values:
+ * - 0 -- Stream information is not present in this message
+ * - 1 -- Stream information is present in this message
+ *
+ * When set to 1, the FLAC bitstream was successfully parsed by the
+ * client, and other fields in the FLAC format block can be read by the
+ * decoder to get metadata stream information.
+ */
+
+	u16 num_channels;
+/* Number of channels for decoding.
+ * Supported values: 1 to 2
+ */
+
+	u16 min_blk_size;
+/* Minimum block size (in samples) used in the stream. It must be less
+ * than or equal to max_blk_size.
+ */
+
+	u16 max_blk_size;
+/* Maximum block size (in samples) used in the stream. If the
+ * minimum block size equals the maximum block size, a fixed block
+ * size stream is implied.
+ */
+
+	u16 md5_sum[8];
+/* MD5 signature array of the unencoded audio data. This allows the
+ * decoder to determine if an error exists in the audio data, even when
+ * the error does not result in an invalid bitstream.
+ */
+
+	u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: 8000 to 48000 Hz
+ */
+
+	u32 min_frame_size;
+/* Minimum frame size used in the stream.
+ * Supported values:
+ * - > 0 bytes
+ * - 0 -- The value is unknown
+ */
+
+	u32 max_frame_size;
+/* Maximum frame size used in the stream.
+ * Supported values:
+ * -- > 0 bytes
+ * -- 0 . The value is unknown
+ */
+
+	u16 sample_size;
+/* Bits per sample.Supported values: 8, 16 */
+
+	u16 reserved;
+/* Clients must set this field to zero
+ */
+
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRNB_FS                  0x00010BEB
+
+/* Enumeration for 4.75 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR475                0
+
+/* Enumeration for 5.15 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR515                1
+
+/* Enumeration for 5.90 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR59                2
+
+/* Enumeration for 6.70 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR67                3
+
+/* Enumeration for 7.40 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR74                4
+
+/* Enumeration for 7.95 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR795               5
+
+/* Enumeration for 10.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR102               6
+
+/* Enumeration for 12.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR122               7
+
+/* Enumeration for AMR-NB Discontinuous Transmission mode off. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF                     0
+
+/* Enumeration for AMR-NB DTX mode VAD1. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1                    1
+
+/* Enumeration for AMR-NB DTX mode VAD2. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD2                    2
+
+/* Enumeration for AMR-NB DTX mode auto.
+	*/
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_AUTO                    3
+
+struct asm_amrnb_enc_cfg {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+
+	u16          enc_mode;
+/* AMR-NB encoding rate.
+ * Supported values:
+ * Use the ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+	u16          dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRWB_FS                  0x00010BEC
+
+/* Enumeration for 6.6 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR66                 0
+
+/* Enumeration for 8.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR885                1
+
+/* Enumeration for 12.65 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1265               2
+
+/* Enumeration for 14.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1425               3
+
+/* Enumeration for 15.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1585               4
+
+/* Enumeration for 18.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1825               5
+
+/* Enumeration for 19.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1985               6
+
+/* Enumeration for 23.05 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2305               7
+
+/* Enumeration for 23.85 kbps AMR-WB Encoding mode.
+	*/
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2385               8
+
+struct asm_amrwb_enc_cfg {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+
+	u16          enc_mode;
+/* AMR-WB encoding rate.
+ * Suupported values:
+ * Use the ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+	u16          dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_V13K_FS                      0x00010BED
+
+/* Enumeration for 14.4 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440                0
+
+/* Enumeration for 12.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220                1
+
+/* Enumeration for 11.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120                2
+
+/* Enumeration for 9.0 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90                  3
+
+/* Enumeration for 7.2 kbps V13K eEncoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720                 4
+
+/* Enumeration for 1/8 vocoder rate.*/
+#define ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE          1
+
+/* Enumeration for 1/4 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE       2
+
+/* Enumeration for 1/2 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_HALF_RATE             3
+
+/* Enumeration for full vocoder rate.
+	*/
+#define ASM_MEDIA_FMT_VOC_FULL_RATE             4
+
+struct asm_v13k_enc_cfg {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+		u16          max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+	u16          min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+	u16          reduced_rate_cmd;
+/* Reduced rate command, used to change
+ * the average bitrate of the V13K
+ * vocoder.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 (Default)
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720
+ */
+
+	u16          rate_mod_cmd;
+/* Rate modulation command. Default = 0.
+ *- If bit 0=1, rate control is enabled.
+ *- If bit 1=1, the maximum number of consecutive full rate
+ *			frames is limited with numbers supplied in
+ *			bits 2 to 10.
+ *- If bit 1=0, the minimum number of non-full rate frames
+ *			in between two full rate frames is forced to
+ * the number supplied in bits 2 to 10. In both cases, if necessary,
+ * half rate is used to substitute full rate. - Bits 15 to 10 are
+ * reserved and must all be set to zero.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_EVRC_FS                   0x00010BEE
+
+/*  EVRC encoder configuration structure used in the
+ * #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
+ */
+struct asm_evrc_enc_cfg {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	u16          max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+	u16          min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+	u16          rate_mod_cmd;
+/* Rate modulation command. Default: 0.
+ * - If bit 0=1, rate control is enabled.
+ * - If bit 1=1, the maximum number of consecutive full rate frames
+ * is limited with numbers supplied in bits 2 to 10.
+ *
+ * - If bit 1=0, the minimum number of non-full rate frames in
+ * between two full rate frames is forced to the number supplied in
+ * bits 2 to 10. In both cases, if necessary, half rate is used to
+ * substitute full rate.
+ *
+ * - Bits 15 to 10 are reserved and must all be set to zero.
+ */
+
+	u16          reserved;
+	/* Reserved. Clients must set this field to zero. */
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V10PRO_V2                0x00010DA7
+
+struct asm_wmaprov10_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+	u16          fmtag;
+/* WMA format type.
+ * Supported values:
+ * - 0x162 -- WMA 9 Pro
+ * - 0x163 -- WMA 9 Pro Lossless
+ * - 0x166 -- WMA 10 Pro
+ * - 0x167 -- WMA 10 Pro Lossless
+ */
+
+	u16          num_channels;
+/* Number of channels encoded in the input stream.
+ * Supported values: 1 to 8
+ */
+
+	u32          sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 11025, 16000, 22050, 32000, 44100, 48000,
+ * 88200, 96000
+ */
+
+	u32          avg_bytes_per_sec;
+/* Bitrate expressed as the average bytes per second.
+ * Supported values: 2000 to 96000
+ */
+
+	u16          blk_align;
+/* Size of the bitstream packet size in bytes. WMA Pro files
+ * have a payload of one block per bitstream packet.
+ * Supported values: @le 13376
+ */
+
+	u16          bits_per_sample;
+/* Number of bits per sample in the encoded WMA stream.
+ * Supported values: 16, 24
+ */
+
+	u32          channel_mask;
+/* Bit-packed double word (32-bits) that indicates the
+ * recommended speaker positions for each source channel.
+ */
+
+	u16          enc_options;
+/* Bit-packed word with values that indicate whether certain
+ * features of the bitstream are used.
+ * Supported values: - 0x0001 -- ENCOPT3_PURE_LOSSLESS - 0x0006 --
+ * ENCOPT3_FRM_SIZE_MOD - 0x0038 -- ENCOPT3_SUBFRM_DIV - 0x0040 --
+ * ENCOPT3_WRITE_FRAMESIZE_IN_HDR - 0x0080 --
+ * ENCOPT3_GENERATE_DRC_PARAMS - 0x0100 -- ENCOPT3_RTMBITS
+ */
+
+
+	u16          usAdvancedEncodeOpt;
+	/* Advanced encoding option.  */
+
+	u32          advanced_enc_options2;
+	/* Advanced encoding option 2. */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V9_V2                    0x00010DA8
+struct asm_wmastdv9_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+	u16          fmtag;
+/* WMA format tag.
+ * Supported values: 0x161 (WMA 9 standard)
+ */
+
+	u16          num_channels;
+/* Number of channels in the stream.
+ * Supported values: 1, 2
+ */
+
+	u32          sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 48000
+ */
+
+	u32          avg_bytes_per_sec;
+	/* Bitrate expressed as the average bytes per second. */
+
+	u16          blk_align;
+/* Block align. All WMA files with a maximum packet size of
+ * 13376 are supported.
+ */
+
+
+	u16          bits_per_sample;
+/* Number of bits per sample in the output.
+ * Supported values: 16
+ */
+
+	u32          channel_mask;
+/* Channel mask.
+ * Supported values:
+ * - 3 -- Stereo (front left/front right)
+ * - 4 -- Mono (center)
+ */
+
+	u16          enc_options;
+	/* Options used during encoding. */
+
+	u16          reserved;
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V8                    0x00010D91
+
+struct asm_wmastdv8_enc_cfg {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	u32          bit_rate;
+	/* Encoding rate in bits per second. */
+
+	u32          sample_rate;
+/* Number of samples per second.
+ *
+ * Supported values:
+ * - 0 -- Native mode
+ * - Other Supported values are 22050, 32000, 44100, and 48000.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+	u16          channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+	u16          reserved;
+	/* Reserved. Clients must set this field to zero.*/
+	} __packed;
+
+#define ASM_MEDIA_FMT_AMR_WB_PLUS_V2               0x00010DA9
+
+struct asm_amrwbplus_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+	u32          amr_frame_fmt;
+/* AMR frame format.
+ * Supported values:
+ * - 6 -- Transport Interface Format (TIF)
+ * - Any other value -- File storage format (FSF)
+ *
+ * TIF stream contains 2-byte header for each frame within the
+ * superframe. FSF stream contains one 2-byte header per superframe.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AC3			0x00010DEE
+#define ASM_MEDIA_FMT_EAC3			0x00010DEF
+#define ASM_MEDIA_FMT_DTS                    0x00010D88
+#define ASM_MEDIA_FMT_MP2                    0x00010DE9
+#define ASM_MEDIA_FMT_FLAC                   0x00010C16
+#define ASM_MEDIA_FMT_ALAC                   0x00012F31
+#define ASM_MEDIA_FMT_VORBIS                 0x00010C15
+#define ASM_MEDIA_FMT_APE                    0x00012F32
+
+
+/* Media format ID for adaptive transform acoustic coding. This
+ * ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED command
+ * only.
+ */
+
+#define ASM_MEDIA_FMT_ATRAC                  0x00010D89
+
+/* Media format ID for metadata-enhanced audio transmission.
+ * This ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
+ * command only.
+ */
+
+#define ASM_MEDIA_FMT_MAT                    0x00010D8A
+
+/*  adsp_media_fmt.h */
+
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+
+struct asm_data_cmd_write_v2 {
+	struct apr_hdr hdr;
+	u32                  buf_addr_lsw;
+/* The 64 bit address msw-lsw should be a valid, mapped address.
+ * 64 bit address should be a multiple of 32 bytes
+ */
+
+	u32                  buf_addr_msw;
+/* The 64 bit address msw-lsw should be a valid, mapped address.
+ * 64 bit address should be a multiple of 32 bytes.
+ * -Address of the buffer containing the data to be decoded.
+ * The buffer should be aligned to a 32 byte boundary.
+ * -In the case of 32 bit Shared memory address, msw field must
+ * -be set to zero.
+ * -In the case of 36 bit shared memory address, bit 31 to bit 4
+ * -of msw must be set to zero.
+ */
+	u32                  mem_map_handle;
+/* memory map handle returned by DSP through
+ * ASM_CMD_SHARED_MEM_MAP_REGIONS command
+ */
+	u32                  buf_size;
+/* Number of valid bytes available in the buffer for decoding. The
+ * first byte starts at buf_addr.
+ */
+
+	u32                  seq_id;
+	/* Optional buffer sequence ID. */
+
+	u32                  timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first buffer sample.
+ */
+
+	u32                  timestamp_msw;
+/* Upper 32 bits of the 64-bit session time in microseconds of the
+ * first buffer sample.
+ */
+
+	u32                  flags;
+/* Bitfield of flags.
+ * Supported values for bit 31:
+ * - 1 -- Valid timestamp.
+ * - 0 -- Invalid timestamp.
+ * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG as the bitmask and
+ * #ASM_SHIFTIMESTAMP_VALID_FLAG as the shift value to set this bit.
+ * Supported values for bit 30:
+ * - 1 -- Last buffer.
+ * - 0 -- Not the last buffer.
+ *
+ * Supported values for bit 29:
+ * - 1 -- Continue the timestamp from the previous buffer.
+ * - 0 -- Timestamp of the current buffer is not related
+ * to the timestamp of the previous buffer.
+ * - Use #ASM_BIT_MASKS_CONTINUE_FLAG and #ASM_SHIFTS_CONTINUE_FLAG
+ * to set this bit.
+ *
+ * Supported values for bit 4:
+ * - 1 -- End of the frame.
+ * - 0 -- Not the end of frame, or this information is not known.
+ * - Use #ASM_BIT_MASK_EOF_FLAG as the bitmask and #ASM_SHIFT_EOF_FLAG
+ * as the shift value to set this bit.
+ *
+ * All other bits are reserved and must be set to 0.
+ *
+ * If bit 31=0 and bit 29=1: The timestamp of the first sample in
+ * this buffer continues from the timestamp of the last sample in
+ * the previous buffer. If there is no previous buffer (i.e., this
+ * is the first buffer sent after opening the stream or after a
+ * flush operation), or if the previous buffer does not have a valid
+ * timestamp, the samples in the current buffer also do not have a
+ * valid timestamp. They are played out as soon as possible.
+ *
+ *
+ * If bit 31=0 and bit 29=0: No timestamp is associated with the
+ * first sample in this buffer. The samples are played out as soon
+ * as possible.
+ *
+ *
+ * If bit 31=1 and bit 29 is ignored: The timestamp specified in
+ * this payload is honored.
+ *
+ *
+ * If bit 30=0: Not the last buffer in the stream. This is useful
+ * in removing trailing samples.
+ *
+ *
+ * For bit 4: The client can set this flag for every buffer sent in
+ * which the last byte is the end of a frame. If this flag is set,
+ * the buffer can contain data from multiple frames, but it should
+ * always end at a frame boundary. Restrictions allow the aDSP to
+ * detect an end of frame without requiring additional processing.
+ */
+
+} __packed;
+
+#define ASM_DATA_CMD_READ_V2 0x00010DAC
+
+struct asm_data_cmd_read_v2 {
+	struct apr_hdr       hdr;
+	u32                  buf_addr_lsw;
+/* the 64 bit address msw-lsw should be a valid mapped address
+ * and should be a multiple of 32 bytes
+ */
+
+
+	u32                  buf_addr_msw;
+/* the 64 bit address msw-lsw should be a valid mapped address
+ * and should be a multiple of 32 bytes.
+* - Address of the buffer where the DSP puts the encoded data,
+* potentially, at an offset specified by the uOffset field in
+* ASM_DATA_EVENT_READ_DONE structure. The buffer should be aligned
+* to a 32 byte boundary.
+*- In the case of 32 bit Shared memory address, msw field must
+*- be set to zero.
+*- In the case of 36 bit shared memory address, bit 31 to bit
+*- 4 of msw must be set to zero.
+*/
+	u32                  mem_map_handle;
+/* memory map handle returned by DSP through
+ * ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+ */
+
+	u32                  buf_size;
+/* Number of bytes available for the aDSP to write. The aDSP
+ * starts writing from buf_addr.
+ */
+
+	u32                  seq_id;
+	/* Optional buffer sequence ID.
+			*/
+} __packed;
+
+#define ASM_DATA_CMD_EOS               0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS    0x00010C1C
+#define ASM_DATA_EVENT_EOS             0x00010BDD
+
+#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
+struct asm_data_event_write_done_v2 {
+	u32                  buf_addr_lsw;
+	/* lsw of the 64 bit address */
+	u32                  buf_addr_msw;
+	/* msw of the 64 bit address. address given by the client in
+	* ASM_DATA_CMD_WRITE_V2 command.
+	*/
+	u32                  mem_map_handle;
+	/* memory map handle in the ASM_DATA_CMD_WRITE_V2  */
+
+	u32                  status;
+/* Status message (error code) that indicates whether the
+ * referenced buffer has been successfully consumed.
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+} __packed;
+
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+
+/* Definition of the frame metadata flag bitmask.*/
+#define ASM_BIT_MASK_FRAME_METADATA_FLAG (0x40000000UL)
+
+/* Definition of the frame metadata flag shift value. */
+#define ASM_SHIFT_FRAME_METADATA_FLAG 30
+
+struct asm_data_event_read_done_v2 {
+	u32                  status;
+/* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+
+u32                  buf_addr_lsw;
+/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
+ * address is a multiple of 32 bytes.
+ */
+
+u32                  buf_addr_msw;
+/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
+* address is a multiple of 32 bytes.
+*
+* -Same address provided by the client in ASM_DATA_CMD_READ_V2
+* -In the case of 32 bit Shared memory address, msw field is set to
+* zero.
+* -In the case of 36 bit shared memory address, bit 31 to bit 4
+* -of msw is set to zero.
+*/
+
+u32                  mem_map_handle;
+/* memory map handle in the ASM_DATA_CMD_READ_V2  */
+
+u32                  enc_framesotal_size;
+/* Total size of the encoded frames in bytes.
+ * Supported values: >0
+ */
+
+u32                  offset;
+/* Offset (from buf_addr) to the first byte of the first encoded
+ * frame. All encoded frames are consecutive, starting from this
+ * offset.
+ * Supported values: > 0
+ */
+
+u32                  timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of
+ * the first sample in the buffer. If Bit 5 of mode_flags flag of
+ * ASM_STREAM_CMD_OPEN_READ_V2 is 1 then the 64 bit timestamp is
+ * absolute capture time otherwise it is relative session time. The
+ * absolute timestamp doesnt reset unless the system is reset.
+ */
+
+
+u32                  timestamp_msw;
+/* Upper 32 bits of the 64-bit session time in microseconds of
+ * the first sample in the buffer.
+ */
+
+
+u32                  flags;
+/* Bitfield of flags. Bit 30 indicates whether frame metadata is
+ * present. If frame metadata is present, num_frames consecutive
+ * instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start
+ * at the buffer address.
+ * Supported values for bit 31:
+ * - 1 -- Timestamp is valid.
+ * - 0 -- Timestamp is invalid.
+ * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and
+ * #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit.
+ *
+ * Supported values for bit 30:
+ * - 1 -- Frame metadata is present.
+ * - 0 -- Frame metadata is absent.
+ * - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and
+ * #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit.
+ *
+ * All other bits are reserved; the aDSP sets them to 0.
+ */
+
+u32                  num_frames;
+/* Number of encoded frames in the buffer. */
+
+u32                  seq_id;
+/* Optional buffer sequence ID.	*/
+} __packed;
+
+struct asm_data_read_buf_metadata_v2 {
+	u32          offset;
+/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to
+ * the frame associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32          frm_size;
+/* Size of the encoded frame in bytes.
+ * Supported values: > 0
+ */
+
+u32          num_encoded_pcm_samples;
+/* Number of encoded PCM samples (per channel) in the frame
+ * associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32          timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ * If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1
+ * then the 64 bit timestamp is absolute capture time otherwise it
+ * is relative session time. The absolute timestamp doesnt reset
+ * unless the system is reset.
+ */
+
+
+u32          timestamp_msw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ */
+
+u32          flags;
+/* Frame flags.
+ * Supported values for bit 31:
+ * - 1 -- Time stamp is valid
+ * - 0 -- Time stamp is not valid
+ * - All other bits are reserved; the aDSP sets them to 0.
+*/
+} __packed;
+
+/* Notifies the client of a change in the data sampling rate or
+ * Channel mode. This event is raised by the decoder service. The
+ * event is enabled through the mode flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in the output sampling frequency or the number/positioning of
+ * output channels, or if it is the first frame decoded.The new
+ * sampling frequency or the new channel configuration is
+ * communicated back to the client asynchronously.
+ */
+
+#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
+
+/*  Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event.
+ * This event is raised when the following conditions are both true:
+ * - The event is enabled through the mode_flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in either the output sampling frequency or the number/positioning
+ * of output channels, or if it is the first frame decoded.
+ * This event is not raised (even if enabled) if the decoder is
+ * MIDI, because
+ */
+
+
+struct asm_data_event_sr_cm_change_notify {
+	u32                  sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * bitstream.
+ * Supported values: 2000 to 48000
+ */
+
+	u16                  num_channels;
+/* New number of channels after detecting a change in the
+ * bitstream.
+ * Supported values: 1 to 8
+ */
+
+
+	u16                  reserved;
+	/* Reserved for future use. This field must be set to 0.*/
+
+	u8                   channel_mapping[8];
+
+} __packed;
+
+/* Notifies the client of a data sampling rate or channel mode
+ * change. This event is raised by the encoder service.
+ * This event is raised when :
+ * - Native mode encoding was requested in the encoder
+ * configuration (i.e., the channel number was 0), the sample rate
+ * was 0, or both were 0.
+ *
+ * - The input data frame at the encoder is the first one, or the
+ * sampling rate/channel mode is different from the previous input
+ * data frame.
+ *
+ */
+#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE
+
+struct asm_data_event_enc_sr_cm_change_notify {
+	u32                  sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * input data.
+ * Supported values: 2000 to 48000
+ */
+
+
+	u16                  num_channels;
+/* New number of channels after detecting a change in the input
+ * data. Supported values: 1 to 8
+ */
+
+
+	u16                  bits_per_sample;
+/* New bits per sample after detecting a change in the input
+ * data.
+ * Supported values: 16, 24
+ */
+
+
+	u8                   channel_mapping[8];
+
+} __packed;
+#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87
+
+
+/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command,
+ * which is used to indicate the IEC 60958 frame rate of a given
+ * packetized audio stream.
+ */
+
+struct asm_data_cmd_iec_60958_frame_rate {
+	u32                  frame_rate;
+/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream.
+ * Supported values: Any valid frame rate
+ */
+} __packed;
+
+/* adsp_asm_data_commands.h*/
+/* Definition of the stream ID bitmask.*/
+#define ASM_BIT_MASK_STREAM_ID                 (0x000000FFUL)
+
+/* Definition of the stream ID shift value.*/
+#define ASM_SHIFT_STREAM_ID                    0
+
+/* Definition of the session ID bitmask.*/
+#define ASM_BIT_MASK_SESSION_ID                (0x0000FF00UL)
+
+/* Definition of the session ID shift value.*/
+#define ASM_SHIFT_SESSION_ID                   8
+
+/* Definition of the service ID bitmask.*/
+#define ASM_BIT_MASK_SERVICE_ID                (0x00FF0000UL)
+
+/* Definition of the service ID shift value.*/
+#define ASM_SHIFT_SERVICE_ID                   16
+
+/* Definition of the domain ID bitmask.*/
+#define ASM_BIT_MASK_DOMAIN_ID                (0xFF000000UL)
+
+/* Definition of the domain ID shift value.*/
+#define ASM_SHIFT_DOMAIN_ID                    24
+
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS               0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS     0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS              0x00010D94
+
+/* adsp_asm_service_commands.h */
+
+#define ASM_MAX_SESSION_ID  (15)
+
+/* Maximum number of sessions.*/
+#define ASM_MAX_NUM_SESSIONS                ASM_MAX_SESSION_ID
+
+/* Maximum number of streams per session.*/
+#define ASM_MAX_STREAMS_PER_SESSION (8)
+#define ASM_SESSION_CMD_RUN_V2                   0x00010DAA
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE  0
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY     3
+
+#define ASM_BIT_MASK_RUN_STARTIME                 (0x00000003UL)
+
+/* Bit shift value used to specify the start time for the
+ * ASM_SESSION_CMD_RUN_V2 command.
+ */
+#define ASM_SHIFT_RUN_STARTIME 0
+struct asm_session_cmd_run_v2 {
+	struct apr_hdr hdr;
+	u32                  flags;
+/* Specifies whether to run immediately or at a specific
+ * rendering time or with a specified delay. Run with delay is
+ * useful for delaying in case of ASM loopback opened through
+ * ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME
+ * and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag.
+ *
+ *
+ *Bits 0 and 1 can take one of four possible values:
+ *
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY
+ *
+ *All other bits are reserved; clients must set them to zero.
+ */
+
+	u32                  time_lsw;
+/* Lower 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+	u32                  time_msw;
+/* Upper 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D
+#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
+
+struct asm_session_cmd_rgstr_rx_underflow {
+	struct apr_hdr hdr;
+	u16                  enable_flag;
+/* Specifies whether a client is to receive events when an Rx
+ * session underflows.
+ * Supported values:
+ * - 0 -- Do not send underflow events
+ * - 1 -- Send underflow events
+ */
+	u16                  reserved;
+	/* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6
+
+struct asm_session_cmd_regx_overflow {
+	struct apr_hdr hdr;
+	u16                  enable_flag;
+/* Specifies whether a client is to receive events when a Tx
+* session overflows.
+ * Supported values:
+ * - 0 -- Do not send overflow events
+ * - 1 -- Send overflow events
+ */
+
+	u16                  reserved;
+	/* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_EVENT_RX_UNDERFLOW        0x00010C17
+#define ASM_SESSION_EVENTX_OVERFLOW           0x00010C18
+#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E
+
+struct asm_session_cmdrsp_get_sessiontime_v3 {
+	u32                  status;
+	/* Status message (error code).
+	* Supported values: Refer to @xhyperref{Q3,[Q3]}
+	*/
+
+	u32                  sessiontime_lsw;
+	/* Lower 32 bits of the current session time in microseconds.*/
+
+	u32                  sessiontime_msw;
+	/* Upper 32 bits of the current session time in microseconds.*/
+
+	u32                  absolutetime_lsw;
+/* Lower 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered * to hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+
+	u32                  absolutetime_msw;
+/* Upper 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered to * hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2     0x00010D9F
+
+struct asm_session_cmd_adjust_session_clock_v2 {
+	struct apr_hdr hdr;
+u32                  adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies the
+ * adjustment time in microseconds to the session clock.
+ *
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+
+	u32                  adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the adjustment time in microseconds to the session clock.
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2    0x00010DA0
+
+struct asm_session_cmdrsp_adjust_session_clock_v2 {
+	u32                  status;
+/* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ * An error means the session clock is not adjusted. In this case,
+ * the next two fields are irrelevant.
+ */
+
+
+	u32                  actual_adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+	u32                  actual_adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+	u32                  cmd_latency_lsw;
+/* Lower 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+
+	u32                  cmd_latency_msw;
+/* Upper 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_GET_PATH_DELAY_V2	 0x00010DAF
+#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0
+
+struct asm_session_cmdrsp_get_path_delay_v2 {
+	u32                  status;
+/* Status message (error code). Whether this get delay operation
+ * is successful or not. Delay value is valid only if status is
+ * success.
+ * Supported values: Refer to @xhyperref{Q5,[Q5]}
+ */
+
+	u32                  audio_delay_lsw;
+	/* Upper 32 bits of the aDSP delay in microseconds. */
+
+	u32                  audio_delay_msw;
+	/* Lower 32 bits of the aDSP delay  in microseconds. */
+
+} __packed;
+
+/* adsp_asm_session_command.h*/
+#define ASM_STREAM_CMD_OPEN_WRITE_V3       0x00010DB3
+
+#define ASM_LOW_LATENCY_STREAM_SESSION				0x10000000
+
+#define ASM_ULTRA_LOW_LATENCY_STREAM_SESSION			0x20000000
+
+#define ASM_ULL_POST_PROCESSING_STREAM_SESSION			0x40000000
+
+#define ASM_LEGACY_STREAM_SESSION                                      0
+
+
+struct asm_stream_cmd_open_write_v3 {
+	struct apr_hdr			hdr;
+	uint32_t                    mode_flags;
+/* Mode flags that configure the stream to notify the client
+ * whenever it detects an SR/CM change at the input to its POPP.
+ * Supported values for bits 0 to 1:
+ * - Reserved; clients must set them to zero.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled.
+ * - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit.
+ *
+ * Supported values for bit 31:
+ * - 0 -- Stream to be opened in on-Gapless mode.
+ * - 1 -- Stream to be opened in Gapless mode. In Gapless mode,
+ * successive streams must be opened with same session ID but
+ * different stream IDs.
+ *
+ * - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and
+ * #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit.
+ *
+ *
+ * @note1hang MIDI and DTMF streams cannot be opened in Gapless mode.
+ */
+
+	uint16_t                    sink_endpointype;
+/*< Sink point type.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - Other values are reserved.
+ *
+ * The device matrix is the gateway to the hardware ports.
+ */
+
+	uint16_t                    bits_per_sample;
+/*< Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+	uint32_t                    postprocopo_id;
+/*< Specifies the topology (order of processing) of
+ * postprocessing algorithms. <i>None</i> means no postprocessing.
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+	uint32_t                    dec_fmt_id;
+/*< Configuration ID of the decoder media format.
+ *
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_FR_FS
+ * - #ASM_MEDIA_FMT_VORBIS
+ * - #ASM_MEDIA_FMT_FLAC
+ * - #ASM_MEDIA_FMT_ALAC
+ * - #ASM_MEDIA_FMT_APE
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
+
+/* Definition of the timestamp type flag bitmask */
+#define ASM_BIT_MASKIMESTAMPYPE_FLAG        (0x00000020UL)
+
+/* Definition of the timestamp type flag shift value. */
+#define ASM_SHIFTIMESTAMPYPE_FLAG 5
+
+/* Relative timestamp is identified by this value.*/
+#define ASM_RELATIVEIMESTAMP      0
+
+/* Absolute timestamp is identified by this value.*/
+#define ASM_ABSOLUTEIMESTAMP      1
+
+/* Bit value for Low Latency Tx stream subfield */
+#define ASM_LOW_LATENCY_TX_STREAM_SESSION			1
+
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ              29
+
+struct asm_stream_cmd_open_read_v3 {
+	struct apr_hdr hdr;
+	u32                    mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ *
+ * - 0 -- Return data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ *
+ * - 1 -- Return data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG as the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG as the shift value for this bit.
+ *
+ *
+ * Supported values for bit 5:
+ *
+ * - ASM_RELATIVEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will have
+ * - relative time-stamp.
+ * - ASM_ABSOLUTEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will
+ * - have absolute time-stamp.
+ *
+ * - Use #ASM_BIT_MASKIMESTAMPYPE_FLAG as the bitmask and
+ * #ASM_SHIFTIMESTAMPYPE_FLAG as the shift value for this bit.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+	u32                    src_endpointype;
+/* Specifies the endpoint providing the input samples.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - All other values are reserved; clients must set them to zero.
+ * Otherwise, an error is returned.
+ * The device matrix is the gateway from the tunneled Tx ports.
+ */
+
+	u32                    preprocopo_id;
+/* Specifies the topology (order of processing) of preprocessing
+ * algorithms. <i>None</i> means no preprocessing.
+ * Supported values:
+ * - #ASM_STREAM_PREPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_PREPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+	u32                    enc_cfg_id;
+/* Media configuration ID for encoded output.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+	u16                    bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+	u16                    reserved;
+/* Reserved for future use. This field must be set to zero.*/
+} __packed;
+
+#define ASM_POPP_OUTPUT_SR_NATIVE_RATE                                  0
+
+/* Enumeration for the maximum sampling rate at the POPP output.*/
+#define ASM_POPP_OUTPUT_SR_MAX_RATE             48000
+
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+
+struct asm_stream_cmd_open_readwrite_v2 {
+	struct apr_hdr         hdr;
+	u32                    mode_flags;
+/* Mode flags.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled. Use
+ * #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or
+ * getting this flag.
+ *
+ * Supported values for bit 4:
+ * - 0 -- Return read data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ * - 1 -- Return read data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+	u32                    postprocopo_id;
+/* Specifies the topology (order of processing) of postprocessing
+ * algorithms. <i>None</i> means no postprocessing.
+ *
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ */
+
+	u32                    dec_fmt_id;
+/* Specifies the media type of the input data. PCM indicates that
+ * no decoding must be performed, e.g., this is an NT encoder
+ * session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+
+	u32                    enc_cfg_id;
+/* Specifies the media type for the output of the stream. PCM
+ * indicates that no encoding must be performed, e.g., this is an NT
+ * decoder session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+	u16                    bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+	u16                    reserved;
+/* Reserved for future use. This field must be set to zero.*/
+
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E
+struct asm_stream_cmd_open_loopback_v2 {
+	struct apr_hdr         hdr;
+	u32                    mode_flags;
+/* Mode flags.
+ * Bit 0-31: reserved; client should set these bits to 0
+ */
+	u16                    src_endpointype;
+	/* Endpoint type. 0 = Tx Matrix */
+	u16                    sink_endpointype;
+	/* Endpoint type. 0 = Rx Matrix */
+	u32                    postprocopo_id;
+/* Postprocessor topology ID. Specifies the topology of
+ * postprocessing algorithms.
+ */
+
+	u16                    bits_per_sample;
+/* The number of bits per sample processed by ASM modules
+ * Supported values: 16 and 24 bits per sample
+ */
+	u16                    reserved;
+/* Reserved for future use. This field must be set to zero. */
+} __packed;
+
+#define ASM_STREAM_CMD_CLOSE             0x00010BCD
+#define ASM_STREAM_CMD_FLUSH             0x00010BCE
+
+
+#define ASM_STREAM_CMD_FLUSH_READBUFS   0x00010C09
+#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1
+
+struct asm_stream_cmd_set_pp_params_v2 {
+	u32                  data_payload_addr_lsw;
+/* LSW of parameter data payload address. Supported values: any. */
+	u32                  data_payload_addr_msw;
+/* MSW of Parameter data payload address. Supported values: any.
+ * - Must be set to zero for in-band data.
+ * - In the case of 32 bit Shared memory address, msw  field must be
+ * - set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw
+ *
+ * - must be set to zero.
+ */
+	u32                  mem_map_handle;
+/* Supported Values: Any.
+* memory map handle returned by DSP through
+* ASM_CMD_SHARED_MEM_MAP_REGIONS
+* command.
+* if mmhandle is NULL, the ParamData payloads are within the
+* message payload (in-band).
+* If mmhandle is non-NULL, the ParamData payloads begin at the
+* address specified in the address msw and lsw (out-of-band).
+*/
+
+	u32                  data_payload_size;
+/* Size in bytes of the variable payload accompanying the
+message, or in shared memory. This field is used for parsing the
+parameter payload. */
+
+} __packed;
+
+
+struct asm_stream_param_data_v2 {
+	u32                  module_id;
+	/* Unique module ID. */
+
+	u32                  param_id;
+	/* Unique parameter ID. */
+
+	u16                  param_size;
+/* Data size of the param_id/module_id combination. This is
+ * a multiple of 4 bytes.
+ */
+
+	u16                  reserved;
+/* Reserved for future enhancements. This field must be set to
+ * zero.
+ */
+
+} __packed;
+
+#define ASM_STREAM_CMD_GET_PP_PARAMS_V2		0x00010DA2
+
+struct asm_stream_cmd_get_pp_params_v2 {
+	u32                  data_payload_addr_lsw;
+	/* LSW of the parameter data payload address. */
+	u32                  data_payload_addr_msw;
+/* MSW of the parameter data payload address.
+ * - Size of the shared memory, if specified, shall be large enough
+ * to contain the whole ParamData payload, including Module ID,
+ * Param ID, Param Size, and Param Values
+ * - Must be set to zero for in-band data
+ * - In the case of 32 bit Shared memory address, msw field must be
+ * set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw must be set to zero.
+ */
+
+	u32                  mem_map_handle;
+/* Supported Values: Any.
+* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS
+* command.
+* if mmhandle is NULL, the ParamData payloads in the ACK are within the
+* message payload (in-band).
+* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the
+* address specified in the address msw and lsw.
+* (out-of-band).
+*/
+
+	u32                  module_id;
+	/* Unique module ID. */
+
+	u32                  param_id;
+	/* Unique parameter ID. */
+
+	u16                  param_max_size;
+/* Maximum data size of the module_id/param_id combination. This
+ * is a multiple of 4 bytes.
+ */
+
+
+	u16                  reserved;
+/* Reserved for backward compatibility. Clients must set this
+* field to zero.
+*/
+
+} __packed;
+
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+
+#define ASM_PARAM_ID_ENCDEC_BITRATE     0x00010C13
+
+struct asm_bitrate_param {
+	u32                  bitrate;
+/* Maximum supported bitrate. Only the AAC encoder is supported.*/
+
+} __packed;
+
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
+#define ASM_PARAM_ID_AAC_SBR_PS_FLAG		 0x00010C63
+
+/* Flag to turn off both SBR and PS processing, if they are
+ * present in the bitstream.
+ */
+
+#define ASM_AAC_SBR_OFF_PS_OFF (2)
+
+/* Flag to turn on SBR but turn off PS processing,if they are
+ * present in the bitstream.
+ */
+
+#define ASM_AAC_SBR_ON_PS_OFF  (1)
+
+/* Flag to turn on both SBR and PS processing, if they are
+ * present in the bitstream (default behavior).
+ */
+
+
+#define ASM_AAC_SBR_ON_PS_ON   (0)
+
+/* Structure for an AAC SBR PS processing flag. */
+
+/*  Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_aac_sbr_ps_flag_param {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+
+	u32                  sbr_ps_flag;
+/* Control parameter to enable or disable SBR/PS processing in
+ * the AAC bitstream. Use the following macros to set this field:
+ * - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS
+ * processing, if they are present in the bitstream.
+ * - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS
+ * processing, if they are present in the bitstream.
+ * - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing,
+ * if they are present in the bitstream (default behavior).
+ * - All other values are invalid.
+ * Changes are applied to the next decoded frame.
+ */
+} __packed;
+
+#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING                      0x00010C64
+
+/*	First single channel element in a dual mono bitstream.*/
+#define ASM_AAC_DUAL_MONO_MAP_SCE_1                                 (1)
+
+/*	Second single channel element in a dual mono bitstream.*/
+#define ASM_AAC_DUAL_MONO_MAP_SCE_2                                 (2)
+
+/* Structure for AAC decoder dual mono channel mapping. */
+
+
+struct asm_aac_dual_mono_mapping_param {
+	struct apr_hdr							hdr;
+	struct asm_stream_cmd_set_encdec_param	encdec;
+	u16    left_channel_sce;
+	u16    right_channel_sce;
+
+} __packed;
+
+#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4
+
+struct asm_stream_cmdrsp_get_pp_params_v2 {
+	u32                  status;
+} __packed;
+
+#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73
+
+/* Enumeration for both vocals in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_NO_VOCAL     (0)
+
+/* Enumeration for only the left vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_LEFT_VOCAL   (1)
+
+/* Enumeration for only the right vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2)
+
+/* Enumeration for both vocal channels in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_BOTH_VOCAL             (3)
+#define ASM_PARAM_ID_AC3_DRC_MODE               0x00010D74
+/* Enumeration for the Custom Analog mode.*/
+#define AC3_DRC_MODE_CUSTOM_ANALOG              (0)
+
+/* Enumeration for the Custom Digital mode.*/
+#define AC3_DRC_MODE_CUSTOM_DIGITAL             (1)
+/* Enumeration for the Line Out mode (light compression).*/
+#define AC3_DRC_MODE_LINE_OUT  (2)
+
+/* Enumeration for the RF remodulation mode (heavy compression).*/
+#define AC3_DRC_MODE_RF_REMOD                         (3)
+#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE               0x00010D75
+
+/* Enumeration for playing dual mono in stereo mode.*/
+#define AC3_DUAL_MONO_MODE_STEREO                     (0)
+
+/* Enumeration for playing left mono.*/
+#define AC3_DUAL_MONO_MODE_LEFT_MONO                  (1)
+
+/* Enumeration for playing right mono.*/
+#define AC3_DUAL_MONO_MODE_RIGHT_MONO                 (2)
+
+/* Enumeration for mixing both dual mono channels and playing them.*/
+#define AC3_DUAL_MONO_MODE_MIXED_MONO        (3)
+#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76
+
+/* Enumeration for using the Downmix mode indicated in the bitstream. */
+
+#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT  (0)
+
+/* Enumeration for Surround Compatible mode (preserves the
+ * surround information).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LT_RT        (1)
+/* Enumeration for Mono Compatible mode (if the output is to be
+ * further downmixed to mono).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2)
+
+/* ID of the AC3 PCM scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78
+
+/* ID of the AC3 DRC boost scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79
+
+/* ID of the AC3 DRC cut scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A
+
+/* Structure for AC3 Generic Parameter. */
+
+/*  Payload of the AC3 parameters in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_ac3_generic_param {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	u32                  generic_parameter;
+/* AC3 generic parameter. Select from one of the following
+ * possible values.
+ *
+ * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are:
+ * - AC3_KARAOKE_MODE_NO_VOCAL
+ * - AC3_KARAOKE_MODE_LEFT_VOCAL
+ * - AC3_KARAOKE_MODE_RIGHT_VOCAL
+ * - AC3_KARAOKE_MODE_BOTH_VOCAL
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are:
+ * - AC3_DRC_MODE_CUSTOM_ANALOG
+ * - AC3_DRC_MODE_CUSTOM_DIGITAL
+ * - AC3_DRC_MODE_LINE_OUT
+ * - AC3_DRC_MODE_RF_REMOD
+ *
+ * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are:
+ * - AC3_DUAL_MONO_MODE_STEREO
+ * - AC3_DUAL_MONO_MODE_LEFT_MONO
+ * - AC3_DUAL_MONO_MODE_RIGHT_MONO
+ * - AC3_DUAL_MONO_MODE_MIXED_MONO
+ *
+ * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are:
+ * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT
+ * - AC3_STEREO_DOWNMIX_MODE_LT_RT
+ * - AC3_STEREO_DOWNMIX_MODE_LO_RO
+ *
+ * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ */
+} __packed;
+
+/* Enumeration for Raw mode (no downmixing), which specifies
+ * that all channels in the bitstream are to be played out as is
+ * without any downmixing. (Default)
+ */
+
+#define WMAPRO_CHANNEL_MASK_RAW (-1)
+
+/* Enumeration for setting the channel mask to 0. The 7.1 mode
+ * (Home Theater) is assigned.
+ */
+
+
+#define WMAPRO_CHANNEL_MASK_ZERO 0x0000
+
+/* Speaker layout mask for one channel (Home Theater, mono).
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_1_C 0x0004
+
+/* Speaker layout mask for two channels (Home Theater, stereo).
+ * - Speaker front left
+ * - Speaker front right
+ */
+#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003
+
+/* Speaker layout mask for three channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007
+
+/* Speaker layout mask for two channels (stereo).
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_2_Bl_Br  0x0030
+
+/* Speaker layout mask for four channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker back left
+ * - Speaker back right
+*/
+#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033
+
+/* Speaker layout mask for four channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back center
+*/
+#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107
+/* Speaker layout mask for five channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br  0x0037
+
+/* Speaker layout mask for five channels (5 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT   0x0607
+/* Speaker layout mask for six channels (5.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF  0x003F
+/* Speaker layout mask for six channels (5.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT  0x060F
+/* Speaker layout mask for six channels (5.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc  0x0137
+/* Speaker layout mask for six channels (5.1 mode, Home Theater,
+  * no LFE).
+  * - Speaker front left
+  * - Speaker front right
+  * - Speaker front center
+  * - Speaker back center
+  * - Speaker side left
+  * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT   0x0707
+
+/* Speaker layout mask for seven channels (6.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF   0x013F
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+  * Theater).
+  * - Speaker front left
+  * - Speaker front right
+  * - Speaker front center
+  * - Speaker low frequency
+  * - Speaker back center
+  * - Speaker side left
+  * - Speaker side right
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F
+
+/* Speaker layout mask for seven channels (6.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker front left of center
+ * - Speaker front right of center
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC   0x00F7
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+ * Theater, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker front left of center
+ * - Speaker front right of center
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637
+
+/* Speaker layout mask for eight channels (7.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ */
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \
+					0x00FF
+
+/* Speaker layout mask for eight channels (7.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ *
+*/
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \
+					0x063F
+
+#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP  0x00010D82
+
+/*	Maximum number of decoder output channels.*/
+#define MAX_CHAN_MAP_CHANNELS  16
+
+/* Structure for decoder output channel mapping. */
+
+/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_dec_out_chan_map_param {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	u32                 num_channels;
+/* Number of decoder output channels.
+ * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS
+ *
+ * A value of 0 indicates native channel mapping, which is valid
+ * only for NT mode. This means the output of the decoder is to be
+ * preserved as is.
+ */
+	u8                  channel_mapping[MAX_CHAN_MAP_CHANNELS];
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED  0x00010D84
+
+/* Bitmask for the IEC 61937 enable flag.*/
+#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG   (0x00000001UL)
+
+/* Shift value for the IEC 61937 enable flag.*/
+#define ASM_SHIFT_IEC_61937_STREAM_FLAG  0
+
+/* Bitmask for the IEC 60958 enable flag.*/
+#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG   (0x00000002UL)
+
+/* Shift value for the IEC 60958 enable flag.*/
+#define ASM_SHIFT_IEC_60958_STREAM_FLAG   1
+
+/* Payload format for open write compressed comand */
+
+/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
+ * comand, which opens a stream for a given session ID and stream ID
+ * to be rendered in the compressed format.
+ */
+
+struct asm_stream_cmd_open_write_compressed {
+	struct apr_hdr hdr;
+	u32                    flags;
+/* Mode flags that configure the stream for a specific format.
+ * Supported values:
+ * - Bit 0 -- IEC 61937 compatibility
+ *   - 0 -- Stream is not in IEC 61937 format
+ *   - 1 -- Stream is in IEC 61937 format
+ * - Bit 1 -- IEC 60958 compatibility
+ *   - 0 -- Stream is not in IEC 60958 format
+ *   - 1 -- Stream is in IEC 60958 format
+ * - Bits 2 to 31 -- 0 (Reserved)
+ *
+ * For the same stream, bit 0 cannot be set to 0 and bit 1 cannot
+ * be set to 1. A compressed stream connot have IEC 60958
+ * packetization applied without IEC 61937 packetization.
+ * @note1hang Currently, IEC 60958 packetized input streams are not
+ * supported.
+ */
+
+
+	u32                    fmt_id;
+/* Specifies the media type of the HDMI stream to be opened.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_DTS
+ * - #ASM_MEDIA_FMT_ATRAC
+ * - #ASM_MEDIA_FMT_MAT
+ *
+ * @note1hang This field must be set to a valid media type even if
+ * IEC 61937 packetization is not performed by the aDSP.
+ */
+
+} __packed;
+
+
+/*
+    Indicates the number of samples per channel to be removed from the
+    beginning of the stream.
+*/
+#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67
+/*
+    Indicates the number of samples per channel to be removed from
+    the end of the stream.
+*/
+#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
+struct asm_data_cmd_remove_silence {
+	struct apr_hdr hdr;
+	u32	num_samples_to_remove;
+	/**< Number of samples per channel to be removed.
+
+	   @values 0 to (2@sscr{32}-1) */
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED                        0x00010D95
+
+struct asm_stream_cmd_open_read_compressed {
+	struct apr_hdr hdr;
+	u32                    mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ * - 0 -- Return data buffer contains all encoded frames only; it does
+ *      not contain frame metadata.
+ * - 1 -- Return data buffer contains an array of metadata and encoded
+ *      frames.
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit.
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+	u32                    frames_per_buf;
+/* Indicates the number of frames that need to be returned per
+ * read buffer
+ * Supported values: should be greater than 0
+ */
+
+} __packed;
+
+/* adsp_asm_stream_commands.h*/
+
+
+/* adsp_asm_api.h (no changes)*/
+#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \
+								0x00010BE4
+#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \
+								0x00010D83
+#define ASM_STREAM_POSTPROCOPO_ID_NONE \
+								0x00010C68
+#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \
+								0x00010D8B
+#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \
+			ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+#define ASM_STREAM_PREPROCOPO_ID_NONE \
+			ASM_STREAM_POSTPROCOPO_ID_NONE
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \
+			0x00010312
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \
+								0x00010313
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \
+								0x00010314
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\
+								0x00010704
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\
+								0x0001070D
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\
+								0x0001070E
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\
+								0x0001070F
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRC_V3 \
+								0x11000000
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \
+								0x0001031B
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP  0x00010315
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316
+#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO           0x00010715
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP   0x00010BE3
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317
+#define AUDPROC_MODULE_ID_AIG   0x00010716
+#define AUDPROC_PARAM_ID_AIG_ENABLE		0x00010717
+#define AUDPROC_PARAM_ID_AIG_CONFIG		0x00010718
+
+struct Audio_AigParam {
+	uint16_t	mode;
+/*< Mode word for enabling AIG/SIG mode .
+ * Byte offset: 0
+ */
+	int16_t		staticGainL16Q12;
+/*< Static input gain when aigMode is set to 1.
+ * Byte offset: 2
+ */
+	int16_t		initialGainDBL16Q7;
+/*<Initial value that the adaptive gain update starts from dB
+ * Q7 Byte offset: 4
+ */
+	int16_t		idealRMSDBL16Q7;
+/*<Average RMS level that AIG attempts to achieve Q8.7
+ * Byte offset: 6
+ */
+	int32_t		noiseGateL32;
+/*Threshold below which signal is considered as noise and AIG
+ * Byte offset: 8
+ */
+	int32_t		minGainL32Q15;
+/*Minimum gain that can be provided by AIG Q16.15
+ * Byte offset: 12
+ */
+	int32_t		maxGainL32Q15;
+/*Maximum gain that can be provided by AIG Q16.15
+ * Byte offset: 16
+ */
+	uint32_t		gainAtRtUL32Q31;
+/*Attack/release time for AIG update Q1.31
+ * Byte offset: 20
+ */
+	uint32_t		longGainAtRtUL32Q31;
+/*Long attack/release time while updating gain for
+ * noise/silence Q1.31 Byte offset: 24
+ */
+
+	uint32_t		rmsTavUL32Q32;
+/* RMS smoothing time constant used for long-term RMS estimate
+ * Q0.32 Byte offset: 28
+ */
+
+	uint32_t		gainUpdateStartTimMsUL32Q0;
+/* The waiting time before which AIG starts to apply adaptive
+ * gain update Q32.0 Byte offset: 32
+ */
+
+} __packed;
+
+
+#define ADM_MODULE_ID_EANS                            0x00010C4A
+#define ADM_PARAM_ID_EANS_ENABLE                      0x00010C4B
+#define ADM_PARAM_ID_EANS_PARAMS                      0x00010C4C
+
+struct adm_eans_enable {
+
+	uint32_t                  enable_flag;
+/*< Specifies whether EANS is disabled (0) or enabled
+ * (nonzero).
+ * This is supported only for sampling rates of 8, 12, 16, 24, 32,
+ * and 48 kHz. It is not supported for sampling rates of 11.025,
+ * 22.05, or 44.1 kHz.
+ */
+
+} __packed;
+
+
+struct adm_eans_params {
+	int16_t                         eans_mode;
+/*< Mode word for enabling/disabling submodules.
+ * Byte offset: 0
+ */
+
+	int16_t                         eans_input_gain;
+/*< Q2.13 input gain to the EANS module.
+ * Byte offset: 2
+ */
+
+	int16_t                         eans_output_gain;
+/*< Q2.13 output gain to the EANS module.
+ * Byte offset: 4
+ */
+
+	int16_t                         eansarget_ns;
+/*< Target noise suppression level in dB.
+ * Byte offset: 6
+ */
+
+	int16_t                         eans_s_alpha;
+/*< Q3.12 over-subtraction factor for stationary noise
+ * suppression.
+ * Byte offset: 8
+ */
+
+	int16_t                         eans_n_alpha;
+/* < Q3.12 over-subtraction factor for nonstationary noise
+ * suppression.
+ * Byte offset: 10
+ */
+
+	int16_t                         eans_n_alphamax;
+/*< Q3.12 maximum over-subtraction factor for nonstationary
+ * noise suppression.
+ * Byte offset: 12
+ */
+	int16_t                         eans_e_alpha;
+/*< Q15 scaling factor for excess noise suppression.
+ * Byte offset: 14
+ */
+
+	int16_t                         eans_ns_snrmax;
+/*< Upper boundary in dB for SNR estimation.
+ * Byte offset: 16
+ */
+
+	int16_t                         eans_sns_block;
+/*< Quarter block size for stationary noise suppression.
+ * Byte offset: 18
+ */
+
+	int16_t                         eans_ns_i;
+/*< Initialization block size for noise suppression.
+ * Byte offset: 20
+ */
+	int16_t                         eans_np_scale;
+/*< Power scale factor for nonstationary noise update.
+ * Byte offset: 22
+ */
+
+	int16_t                         eans_n_lambda;
+/*< Smoothing factor for higher level nonstationary noise
+ * update.
+ * Byte offset: 24
+ */
+
+	int16_t                         eans_n_lambdaf;
+/*< Medium averaging factor for noise update.
+ * Byte offset: 26
+ */
+
+	int16_t                         eans_gs_bias;
+/*< Bias factor in dB for gain calculation.
+ * Byte offset: 28
+ */
+
+	int16_t                         eans_gs_max;
+/*< SNR lower boundary in dB for aggressive gain calculation.
+ * Byte offset: 30
+ */
+
+	int16_t                         eans_s_alpha_hb;
+/*< Q3.12 over-subtraction factor for high-band stationary
+ * noise suppression.
+ * Byte offset: 32
+ */
+
+	int16_t                         eans_n_alphamax_hb;
+/*< Q3.12 maximum over-subtraction factor for high-band
+ * nonstationary noise suppression.
+ * Byte offset: 34
+ */
+
+	int16_t                         eans_e_alpha_hb;
+/*< Q15 scaling factor for high-band excess noise suppression.
+ * Byte offset: 36
+ */
+
+	int16_t                         eans_n_lambda0;
+/*< Smoothing factor for nonstationary noise update during
+ * speech activity.
+ * Byte offset: 38
+ */
+
+	int16_t                         thresh;
+/*< Threshold for generating a binary VAD decision.
+ * Byte offset: 40
+ */
+
+	int16_t                         pwr_scale;
+/*< Indirect lower boundary of the noise level estimate.
+ * Byte offset: 42
+ */
+
+	int16_t                         hangover_max;
+/*< Avoids mid-speech clipping and reliably detects weak speech
+ * bursts at the end of speech activity.
+ * Byte offset: 44
+ */
+
+	int16_t                         alpha_snr;
+/*< Controls responsiveness of the VAD.
+ * Byte offset: 46
+ */
+
+	int16_t                         snr_diff_max;
+/*< Maximum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 48
+ */
+
+	int16_t                         snr_diff_min;
+/*< Minimum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 50
+ */
+
+	int16_t                         init_length;
+/*< Defines the number of frames for which a noise level
+ * estimate is set to a fixed value.
+ * Byte offset: 52
+ */
+
+	int16_t                         max_val;
+/*< Defines the upper limit of the noise level.
+ * Byte offset: 54
+ */
+
+	int16_t                         init_bound;
+/*< Defines the initial bounding value for the noise level
+ * estimate. This is used during the initial segment defined by the
+ * init_length parameter.
+ * Byte offset: 56
+ */
+
+	int16_t                         reset_bound;
+/*< Reset boundary for noise tracking.
+ * Byte offset: 58
+ */
+
+	int16_t                         avar_scale;
+/*< Defines the bias factor in noise estimation.
+ * Byte offset: 60
+ */
+
+	int16_t                         sub_nc;
+/*< Defines the window length for noise estimation.
+ * Byte offset: 62
+ */
+
+	int16_t                         spow_min;
+/*< Defines the minimum signal power required to update the
+ * boundaries for the noise floor estimate.
+ * Byte offset: 64
+ */
+
+	int16_t                         eans_gs_fast;
+/*< Fast smoothing factor for postprocessor gain.
+ * Byte offset: 66
+ */
+
+	int16_t                         eans_gs_med;
+/*< Medium smoothing factor for postprocessor gain.
+ * Byte offset: 68
+ */
+
+	int16_t                         eans_gs_slow;
+/*< Slow smoothing factor for postprocessor gain.
+ * Byte offset: 70
+ */
+
+	int16_t                         eans_swb_salpha;
+/*< Q3.12 super wideband aggressiveness factor for stationary
+ * noise suppression.
+ * Byte offset: 72
+ */
+
+	int16_t                         eans_swb_nalpha;
+/*< Q3.12 super wideband aggressiveness factor for
+ * nonstationary noise suppression.
+ * Byte offset: 74
+ */
+} __packed;
+#define ADM_MODULE_IDX_MIC_GAIN_CTRL   0x00010C35
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Tx mic gain control parameter used by the
+ * #ADM_MODULE_IDX_MIC_GAIN_CTRL module.
+ * @messagepayload
+ * @structure{admx_mic_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_MIC_GAIN       0x00010C36
+
+/* Structure for a Tx mic gain parameter for the mic gain
+ * control module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the
+ * Tx Mic Gain Control module.
+ */
+struct admx_mic_gain {
+	uint16_t                  tx_mic_gain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero. */
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Rx Codec Gain Control module.
+ *
+ * This module supports the following parameter ID:
+ * - #ADM_PARAM_ID_RX_CODEC_GAIN
+ */
+#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL       0x00010C37
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Rx codec gain control parameter used by the
+ * #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module.
+ *
+ * @messagepayload
+ * @structure{adm_rx_codec_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex}
+*/
+#define ADM_PARAM_ID_RX_CODEC_GAIN   0x00010C38
+
+/* Structure for the Rx common codec gain control module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter
+ * in the Rx Codec Gain Control module.
+ */
+
+
+struct adm_rx_codec_gain {
+	uint16_t                  rx_codec_gain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the HPF Tuning Filter module on the Tx path.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ */
+#define ADM_MODULE_ID_HPF_IIRX_FILTER    0x00010C3D
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the Tx HPF IIR filter enable parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG   0x00010C3E
+
+/* ID of the Tx HPF IIR filter pregain parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN   0x00010C3F
+
+/* ID of the Tx HPF IIR filter configuration parameters used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA
+ * RAMS.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS  0x00010C40
+
+/* Structure for enabling a configuration parameter for
+ * the HPF IIR tuning filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * parameter in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_enable_cfg {
+	uint32_t                  enable_flag;
+/*< Specifies whether the HPF tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the pregain parameter for the HPF
+	IIR tuning filter module on the Tx path. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter
+ * in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_pre_gain {
+	uint16_t                  pre_gain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+	HPF IIR tuning filter module on the Tx path. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ * parameters in the Tx path HPF Tuning Filter module. \n
+ * \n
+ * This structure is followed by tuning filter coefficients as follows: \n
+ * - Sequence of int32_t FilterCoeffs.
+ * Each band has five coefficients, each in int32_t format in the order of
+ * b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor.
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting.
+ * One uint16_t for each band to indicate application of the filter to
+ * left (0), right (1), or both (2) channels.
+ */
+struct adm_hpfx_iir_filter_cfg_params {
+	uint16_t                  num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_module_ids */
+/* ID of the Tx path IIR Tuning Filter module.
+ *	This module supports the following parameter IDs:
+ *	- #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ */
+#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41
+
+/* ID of the Rx path IIR Tuning Filter module for the left channel.
+ *	The parameter IDs of the IIR tuning filter module
+ *	(#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning
+ *	filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter; the pan
+ * parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER      0x00010705
+
+/* ID of the the Rx path IIR Tuning Filter module for the right
+ * channel.
+ * The parameter IDs of the IIR tuning filter module
+ * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx
+ * tuning filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter;
+ * the pan parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER    0x00010706
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_param_ids */
+
+/* ID of the Tx IIR filter enable parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG   0x00010C42
+
+/* ID of the Tx IIR filter pregain parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN    0x00010C43
+
+/* ID of the Tx IIR filter configuration parameters used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS     0x00010C44
+
+/* Structure for enabling the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_enable_cfg {
+	uint32_t                  enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+
+} __packed;
+
+
+/* Structure for the pregain parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_pre_gain {
+	uint16_t                  pre_gain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS
+ * parameter in the Tx Path IIR Tuning Filter module. \n
+ *	\n
+ * This structure is followed by the HPF IIR filter coefficients on
+ * the Tx path as follows: \n
+ * - Sequence of int32_t ulFilterCoeffs. Each band has five
+ * coefficients, each in int32_t format in the order of b0, b1, b2,
+ * a1, a2.
+ * - Sequence of int16_t sNumShiftFactor. One int16_t per band. The
+ * numerator shift factor is related to the Q factor of the filter
+ * coefficients.
+ * - Sequence of uint16_t usPanSetting. One uint16_t for each band
+ * to indicate if the filter is applied to left (0), right (1), or
+ * both (2) channels.
+ */
+struct admx_iir_filter_cfg_params {
+	uint16_t                  num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ *	ID of the QEnsemble module.
+ *	This module supports the following parameter IDs:
+ *	- #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ *	- #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ *	- #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ */
+#define ADM_MODULE_ID_QENSEMBLE    0x00010C59
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the QEnsemble enable parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_ENABLE   0x00010C60
+
+/* ID of the QEnsemble back gain parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_backgain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN   0x00010C61
+
+/* ID of the QEnsemble new angle parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_set_new_angle}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE    0x00010C62
+
+/* Structure for enabling the configuration parameter for the
+ * QEnsemble module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ * parameter used by the QEnsemble module.
+ */
+struct adm_qensemble_enable {
+	uint32_t                  enable_flag;
+/*< Specifies whether the QEnsemble module is disabled (0) or enabled
+ * (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the background gain for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ * parameter used by
+ * the QEnsemble module.
+ */
+struct adm_qensemble_param_backgain {
+	int16_t                  back_gain;
+/*< Linear gain in Q15 format.
+ * Supported values: 0 to 32767
+ */
+
+	uint16_t                 reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+/* Structure for setting a new angle for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ * parameter used
+ * by the QEnsemble module.
+ */
+struct adm_qensemble_param_set_new_angle {
+	int16_t                    new_angle;
+/*< New angle in degrees.
+ * Supported values: 0 to 359
+ */
+
+	int16_t                    time_ms;
+/*< Transition time in milliseconds to set the new angle.
+ * Supported values: 0 to 32767
+ */
+} __packed;
+
+
+#define ADM_CMD_GET_PP_TOPO_MODULE_LIST				0x00010349
+#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST			0x00010350
+#define AUDPROC_PARAM_ID_ENABLE					0x00010904
+ /*
+  * Payload of the ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
+  */
+struct adm_cmd_get_pp_topo_module_list_t {
+	struct apr_hdr hdr;
+	/* Lower 32 bits of the 64-bit parameter data payload address. */
+	uint32_t                  data_payload_addr_lsw;
+	/*
+	 * Upper 32 bits of the 64-bit parameter data payload address.
+	 *
+	 *
+	 * The size of the shared memory, if specified, must be large enough to
+	 * contain the entire parameter data payload, including the module ID,
+	 * parameter ID, parameter size, and parameter values.
+	 */
+	uint32_t                  data_payload_addr_msw;
+	/*
+	 *  Unique identifier for an address.
+	 *
+	 * This memory map handle is returned by the aDSP through the
+	 * #ADM_CMD_SHARED_MEM_MAP_REGIONS command.
+	 *
+	 * @values
+	 * - Non-NULL -- On acknowledgment, the parameter data payloads begin at
+	 * the address specified (out-of-band)
+	 * - NULL -- The acknowledgment's payload contains the parameter data
+	 * (in-band) @tablebulletend
+	 */
+	uint32_t                  mem_map_handle;
+	/*
+	 * Maximum data size of the list of modules. This
+	 * field is a multiple of 4 bytes.
+	 */
+	uint16_t                  param_max_size;
+	/* This field must be set to zero. */
+	uint16_t                  reserved;
+} __packed;
+
+/*
+ * Payload of the ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST message, which returns
+ * module ids in response to an ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
+ * Immediately following this structure is the acknowledgement <b>module id
+ * data variable payload</b> containing the pre/postprocessing module id
+ * values. For an in-band scenario, the variable payload depends on the size
+ * of the parameter.
+ */
+struct adm_cmd_rsp_get_pp_topo_module_list_t {
+	/* Status message (error code). */
+	uint32_t                  status;
+} __packed;
+
+struct audproc_topology_module_id_info_t {
+	uint32_t	num_modules;
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Volume Control module pre/postprocessing block.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * - #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * - #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * - #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ */
+#define ASM_MODULE_ID_VOL_CTRL   0x00010BFE
+#define ASM_MODULE_ID_VOL_CTRL2  0x00010910
+#define AUDPROC_MODULE_ID_VOL_CTRL ASM_MODULE_ID_VOL_CTRL
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_master_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN    0x00010BFF
+#define AUDPROC_PARAM_ID_VOL_CTRL_MASTER_GAIN ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+
+/* ID of the left/right channel gain parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_lr_chan_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MULTICHANNEL_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN     0x00010C00
+
+/* ID of the mute configuration parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_mute_config}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG   0x00010C01
+
+/* ID of the soft stepping volume parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_soft_step_volume_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET
+ * ERS.tex}
+ */
+#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS  0x00010C29
+#define AUDPROC_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS\
+			ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+
+/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ */
+#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS   0x00010D6A
+
+/* ID of the multiple-channel volume control parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+#define ASM_PARAM_ID_MULTICHANNEL_GAIN  0x00010713
+
+/* ID of the multiple-channel mute configuration parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+
+#define ASM_PARAM_ID_MULTICHANNEL_MUTE  0x00010714
+
+/* Structure for the master gain parameter for a volume control
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * parameter used by the Volume Control module.
+ */
+
+
+
+struct asm_volume_ctrl_master_gain {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint16_t                  master_gain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.
+		*/
+} __packed;
+
+
+struct asm_volume_ctrl_lr_chan_gain {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+
+	uint16_t                  l_chan_gain;
+	/*< Linear gain in Q13 format for the left channel. */
+
+	uint16_t                  r_chan_gain;
+	/*< Linear gain in Q13 format for the right channel.*/
+} __packed;
+
+
+/* Structure for the mute configuration parameter for a
+	volume control module. */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * parameter used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_mute_config {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint32_t                  mute_flag;
+/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/
+
+} __packed;
+
+/*
+ * Supported parameters for a soft stepping linear ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR  0
+
+/*
+ * Exponential ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_EXP    1
+
+/*
+ * Logarithmic ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LOG    2
+
+/* Structure for holding soft stepping volume parameters. */
+
+
+/*  Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+struct asm_soft_step_volume_params {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint32_t                  period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+	uint32_t                  step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+	uint32_t                  ramping_curve;
+/*< Ramping curve type.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Structure for holding soft pause parameters. */
+
+
+/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_soft_pause_params {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint32_t                  enable_flag;
+/*< Specifies whether soft pause is disabled (0) or enabled
+ * (nonzero).
+ */
+
+
+
+	uint32_t                  period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+	uint32_t                  step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+	uint32_t                  ramping_curve;
+/*< Ramping curve.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Maximum number of channels.*/
+#define VOLUME_CONTROL_MAX_CHANNELS                       8
+
+/* Structure for holding one channel type - gain pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel
+ * type/gain pairs used by the Volume Control module. \n \n This
+ * structure immediately follows the
+ * asm_volume_ctrl_multichannel_gain structure.
+ */
+
+
+struct asm_volume_ctrl_channeltype_gain_pair {
+	uint8_t                   channeltype;
+	/*
+	 * Channel type for which the gain setting is to be applied.
+	 * Supported values:
+	 * - #PCM_CHANNEL_L
+	 * - #PCM_CHANNEL_R
+	 * - #PCM_CHANNEL_C
+	 * - #PCM_CHANNEL_LS
+	 * - #PCM_CHANNEL_RS
+	 * - #PCM_CHANNEL_LFE
+	 * - #PCM_CHANNEL_CS
+	 * - #PCM_CHANNEL_LB
+	 * - #PCM_CHANNEL_RB
+	 * - #PCM_CHANNELS
+	 * - #PCM_CHANNEL_CVH
+	 * - #PCM_CHANNEL_MS
+	 * - #PCM_CHANNEL_FLC
+	 * - #PCM_CHANNEL_FRC
+	 * - #PCM_CHANNEL_RLC
+	 * - #PCM_CHANNEL_RRC
+	 */
+
+	uint8_t                   reserved1;
+	/* Clients must set this field to zero. */
+
+	uint8_t                   reserved2;
+	/* Clients must set this field to zero. */
+
+	uint8_t                   reserved3;
+	/* Clients must set this field to zero. */
+
+	uint32_t                  gain;
+	/*
+	 * Gain value for this channel in Q28 format.
+	 * Supported values: Any
+	 */
+} __packed;
+
+
+/* Structure for the multichannel gain command */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_gain {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint32_t                  num_channels;
+	/*
+	 * Number of channels for which gain values are provided. Any
+	 * channels present in the data for which gain is not provided are
+	 * set to unity gain.
+	 * Supported values: 1 to 8
+	 */
+
+	struct asm_volume_ctrl_channeltype_gain_pair
+		gain_data[VOLUME_CONTROL_MAX_CHANNELS];
+	/* Array of channel type/gain pairs.*/
+} __packed;
+
+
+/* Structure for holding one channel type - mute pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel
+ * type/mute setting pairs used by the Volume Control module. \n \n
+ * This structure immediately follows the
+ * asm_volume_ctrl_multichannel_mute structure.
+ */
+
+
+struct asm_volume_ctrl_channelype_mute_pair {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint8_t                   channelype;
+/*< Channel type for which the mute setting is to be applied.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ */
+
+	uint8_t                   reserved1;
+	/*< Clients must set this field to zero. */
+
+	uint8_t                   reserved2;
+	/*< Clients must set this field to zero. */
+
+	uint8_t                   reserved3;
+	/*< Clients must set this field to zero. */
+
+	uint32_t                  mute;
+/*< Mute setting for this channel.
+ * Supported values:
+ * - 0 = Unmute
+ * - Nonzero = Mute
+ */
+} __packed;
+
+
+/* Structure for the multichannel mute command */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_mute {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	uint32_t                  num_channels;
+/*< Number of channels for which mute configuration is
+ * provided. Any channels present in the data for which mute
+ * configuration is not provided are set to unmute.
+ * Supported values: 1 to 8
+ */
+
+struct asm_volume_ctrl_channelype_mute_pair
+				mute_data[VOLUME_CONTROL_MAX_CHANNELS];
+	/*< Array of channel type/mute setting pairs.*/
+} __packed;
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the IIR Tuning Filter module.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_IIRUNING_FILTER   0x00010C02
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the IIR tuning filter enable parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ * @messagepayload
+ * @structure{asm_iiruning_filter_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO
+ * NFIG.tex}
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG   0x00010C03
+
+/* ID of the IIR tuning filter pregain parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN  0x00010C04
+
+/* ID of the IIR tuning filter configuration parameters used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS  0x00010C05
+
+/* Structure for an enable configuration parameter for an
+ * IIR tuning filter module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * parameter used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_enable {
+	uint32_t                  enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (1).
+ */
+} __packed;
+
+/* Structure for the pregain parameter for an IIR tuning filter module. */
+
+
+/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * parameters used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_pregain {
+	uint16_t                  pregain;
+	/*< Linear gain in Q13 format. */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+/* Structure for the configuration parameter for an IIR tuning filter
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ * parameters used by the IIR Tuning Filter module. \n
+ * \n
+ * This structure is followed by the IIR filter coefficients: \n
+ * - Sequence of int32_t FilterCoeffs \n
+ * Five coefficients for each band. Each coefficient is in int32_t format, in
+ * the order of b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor \n
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting \n
+ * One uint16_t per band, indicating if the filter is applied to left (0),
+ * right (1), or both (2) channels.
+ */
+struct asm_iir_filter_config_params {
+	uint16_t                  num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero.*/
+} __packed;
+
+/* audio_pp_module_ids
+ * ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx
+ * paths.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_MBDRC_ENABLE
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_MBDRC   0x00010C06
+
+/* audio_pp_param_ids */
+/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex}
+ */
+#define ASM_PARAM_ID_MBDRC_ENABLE   0x00010C07
+
+/* ID of the MBDRC configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex}
+ *
+ * @parspace Sub-band DRC configuration parameters
+ * @structure{asm_subband_drc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex}
+ *
+ * @keep{6}
+ * To obtain legacy ADRC from MBDRC, use the calibration tool to:
+ *
+ * - Enable MBDRC (EnableFlag = TRUE)
+ * - Set number of bands to 1 (uiNumBands = 1)
+ * - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1)
+ * - Clear the first band mute flag (MuteFlag[0] = 0)
+ * - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000)
+ * - Use the legacy ADRC parameters to calibrate the rest of the MBDRC
+ * parameters.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS  0x00010C08
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the MMBDRC module version 2 pre/postprocessing block.
+ * This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in
+ * the length of the filters used in each sub-band.
+ * This module supports the following parameter ID:
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2
+ */
+#define ASM_MODULE_ID_MBDRCV2                                0x0001070B
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure
+ * of the MBDRC v2 pre/postprocessing block.
+ * The update to this configuration structure from the original
+ * MBDRC is the number of filter coefficients in the filter
+ * structure. The sequence for is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t
+ * padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags +
+ * uint16_t padding
+ *	This block uses the same parameter structure as
+ *	#ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \
+								0x0001070C
+
+#define ASM_MODULE_ID_MBDRCV3					0x0001090B
+/*
+ * ID of the MMBDRC module version 3 pre/postprocessing block.
+ * This module differs from MBDRCv2 (#ASM_MODULE_ID_MBDRCV2) in
+ * that it supports both 16- and 24-bit data.
+ * This module supports the following parameter ID:
+ * - #ASM_PARAM_ID_MBDRC_ENABLE
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_V3
+ * - #ASM_PARAM_ID_MBDRC_FILTER_XOVER_FREQS
+ */
+
+/* Structure for the enable parameter for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the
+ * MBDRC module.
+ */
+struct asm_mbdrc_enable {
+	uint32_t                  enable_flag;
+/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/
+} __packed;
+
+/* Structure for the configuration parameters for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ * parameters used by the MBDRC module. \n \n Following this
+ * structure is the payload for sub-band DRC configuration
+ * parameters (asm_subband_drc_config_params). This sub-band
+ * structure must be repeated for each band.
+ */
+
+
+struct asm_mbdrc_config_params {
+	uint16_t                  num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 5
+ */
+
+	int16_t                   limiterhreshold;
+/*< Threshold in decibels for the limiter output.
+ * Supported values: -72 to 18 \n
+ * Recommended value: 3994 (-0.22 db in Q3.12 format)
+ */
+
+	int16_t                   limiter_makeup_gain;
+/*< Makeup gain in decibels for the limiter output.
+ * Supported values: -42 to 42 \n
+ * Recommended value: 256 (0 dB in Q7.8 format)
+ */
+
+	int16_t                   limiter_gc;
+/*< Limiter gain recovery coefficient.
+ * Supported values: 0.5 to 0.99 \n
+ * Recommended value: 32440 (0.99 in Q15 format)
+ */
+
+	int16_t                   limiter_delay;
+/*< Limiter delay in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+
+	int16_t                   limiter_max_wait;
+/*< Maximum limiter waiting time in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+} __packed;
+
+/* DRC configuration structure for each sub-band of an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC
+ * configuration parameters for each sub-band in the MBDRC module.
+ * After this DRC structure is configured for valid bands, the next
+ * MBDRC setparams expects the sequence of sub-band MBDRC filter
+ * coefficients (the length depends on the number of bands) plus the
+ * mute flag for that band plus uint16_t padding.
+ *
+ * @keep{10}
+ * The filter coefficient and mute flag are of type int16_t:
+ * - FIR coefficient = int16_t firFilter
+ * - Mute flag = int16_t fMuteFlag
+ *
+ * The sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding
+ *
+ * For improved filterbank, the sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding
+ */
+struct asm_subband_drc_config_params {
+	int16_t                   drc_stereo_linked_flag;
+/*< Specifies whether all stereo channels have the same applied
+ * dynamics (1) or if they process their dynamics independently (0).
+ * Supported values:
+ * - 0 -- Not linked
+ * - 1 -- Linked
+ */
+
+	int16_t                   drc_mode;
+/*< Specifies whether DRC mode is bypassed for sub-bands.
+ * Supported values:
+ * - 0 -- Disabled
+ * - 1 -- Enabled
+ */
+
+	int16_t                   drc_down_sample_level;
+/*< DRC down sample level.
+ * Supported values: @ge 1
+ */
+
+	int16_t                   drc_delay;
+/*< DRC delay in samples.
+ * Supported values: 0 to 1200
+ */
+
+	uint16_t                  drc_rmsime_avg_const;
+/*< RMS signal energy time-averaging constant.
+ * Supported values: 0 to 2^16-1
+ */
+
+	uint16_t                  drc_makeup_gain;
+/*< DRC makeup gain in decibels.
+ * Supported values: 258 to 64917
+ */
+	/* Down expander settings */
+	int16_t                   down_expdrhreshold;
+/*< Down expander threshold.
+ * Supported Q7 format values: 1320 to up_cmpsrhreshold
+ */
+
+	int16_t                   down_expdr_slope;
+/*< Down expander slope.
+ * Supported Q8 format values: -32768 to 0.
+ */
+
+	uint32_t                  down_expdr_attack;
+/*< Down expander attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+	uint32_t                  down_expdr_release;
+/*< Down expander release constant.
+ * Supported Q31 format values: 19685 to 2^31
+ */
+
+	uint16_t                  down_expdr_hysteresis;
+/*< Down expander hysteresis constant.
+ * Supported Q14 format values: 1 to 32690
+ */
+
+	uint16_t                  reserved;
+	/*< Clients must set this field to zero. */
+
+	int32_t                   down_expdr_min_gain_db;
+/*< Down expander minimum gain.
+ * Supported Q23 format values: -805306368 to 0.
+ */
+
+	/* Up compressor settings */
+
+	int16_t                   up_cmpsrhreshold;
+/*< Up compressor threshold.
+ * Supported Q7 format values: down_expdrhreshold to
+ * down_cmpsrhreshold.
+ */
+
+	uint16_t                  up_cmpsr_slope;
+/*< Up compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+	uint32_t                  up_cmpsr_attack;
+/*< Up compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+	uint32_t                  up_cmpsr_release;
+/*< Up compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+	uint16_t                  up_cmpsr_hysteresis;
+/*< Up compressor hysteresis constant.
+  * Supported Q14 format values: 1 to 32690.
+  */
+
+	/* Down compressor settings */
+
+	int16_t                   down_cmpsrhreshold;
+/*< Down compressor threshold.
+ * Supported Q7 format values: up_cmpsrhreshold to 11560.
+ */
+
+	uint16_t                  down_cmpsr_slope;
+/*< Down compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+	uint16_t                  reserved1;
+/*< Clients must set this field to zero. */
+
+	uint32_t                  down_cmpsr_attack;
+/*< Down compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+	uint32_t                  down_cmpsr_release;
+/*< Down compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+	uint16_t                  down_cmpsr_hysteresis;
+/*< Down compressor hysteresis constant.
+ * Supported Q14 values: 1 to 32690.
+ */
+
+	uint16_t                  reserved2;
+/*< Clients must set this field to zero.*/
+} __packed;
+
+#define ASM_MODULE_ID_EQUALIZER            0x00010C27
+#define ASM_PARAM_ID_EQUALIZER_PARAMETERS  0x00010C28
+
+#define ASM_MAX_EQ_BANDS 12
+
+struct asm_eq_per_band_params {
+	uint32_t                  band_idx;
+/*< Band index.
+ * Supported values: 0 to 11
+ */
+
+	uint32_t                  filterype;
+/*< Type of filter.
+ * Supported values:
+ * - #ASM_PARAM_EQYPE_NONE
+ * - #ASM_PARAM_EQ_BASS_BOOST
+ * - #ASM_PARAM_EQ_BASS_CUT
+ * - #ASM_PARAM_EQREBLE_BOOST
+ * - #ASM_PARAM_EQREBLE_CUT
+ * - #ASM_PARAM_EQ_BAND_BOOST
+ * - #ASM_PARAM_EQ_BAND_CUT
+ */
+
+	uint32_t                  center_freq_hz;
+	/*< Filter band center frequency in Hertz. */
+
+	int32_t                   filter_gain;
+/*< Filter band initial gain.
+ * Supported values: +12 to -12 dB in 1 dB increments
+ */
+
+	int32_t                   q_factor;
+/*< Filter band quality factor expressed as a Q8 number, i.e., a
+ * fixed-point number with q factor of 8. For example, 3000/(2^8).
+ */
+} __packed;
+
+struct asm_eq_params {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+		uint32_t                  enable_flag;
+/*< Specifies whether the equalizer module is disabled (0) or enabled
+ * (nonzero).
+ */
+
+		uint32_t                  num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 12
+ */
+	struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS];
+
+} __packed;
+
+/*	No equalizer effect.*/
+#define ASM_PARAM_EQYPE_NONE      0
+
+/*	Bass boost equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_BOOST     1
+
+/*Bass cut equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_CUT       2
+
+/*	Treble boost equalizer effect */
+#define ASM_PARAM_EQREBLE_BOOST   3
+
+/*	Treble cut equalizer effect.*/
+#define ASM_PARAM_EQREBLE_CUT     4
+
+/*	Band boost equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_BOOST     5
+
+/*	Band cut equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_CUT       6
+
+/* Voice get & set params */
+#define VOICE_CMD_SET_PARAM				0x0001133D
+#define VOICE_CMD_GET_PARAM				0x0001133E
+#define VOICE_EVT_GET_PARAM_ACK				0x00011008
+
+
+/** ID of the Bass Boost module.
+    This module supports the following parameter IDs:
+    - #AUDPROC_PARAM_ID_BASS_BOOST_ENABLE
+    - #AUDPROC_PARAM_ID_BASS_BOOST_MODE
+    - #AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH
+*/
+#define AUDPROC_MODULE_ID_BASS_BOOST                             0x000108A1
+/** ID of the Bass Boost enable parameter used by
+    AUDPROC_MODULE_ID_BASS_BOOST.
+*/
+#define AUDPROC_PARAM_ID_BASS_BOOST_ENABLE                       0x000108A2
+/** ID of the Bass Boost mode parameter used by
+    AUDPROC_MODULE_ID_BASS_BOOST.
+*/
+#define AUDPROC_PARAM_ID_BASS_BOOST_MODE                         0x000108A3
+/** ID of the Bass Boost strength parameter used by
+    AUDPROC_MODULE_ID_BASS_BOOST.
+*/
+#define AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH                     0x000108A4
+
+/** ID of the PBE module.
+    This module supports the following parameter IDs:
+    - #AUDPROC_PARAM_ID_PBE_ENABLE
+    - #AUDPROC_PARAM_ID_PBE_PARAM_CONFIG
+*/
+#define AUDPROC_MODULE_ID_PBE                                    0x00010C2A
+/** ID of the Bass Boost enable parameter used by
+    AUDPROC_MODULE_ID_BASS_BOOST.
+*/
+#define AUDPROC_PARAM_ID_PBE_ENABLE                              0x00010C2B
+/** ID of the Bass Boost mode parameter used by
+    AUDPROC_MODULE_ID_BASS_BOOST.
+*/
+#define AUDPROC_PARAM_ID_PBE_PARAM_CONFIG                        0x00010C49
+
+/** ID of the Virtualizer module. This module supports the
+    following parameter IDs:
+    - #AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE
+    - #AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH
+    - #AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE
+    - #AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST
+*/
+#define AUDPROC_MODULE_ID_VIRTUALIZER                            0x000108A5
+/** ID of the Virtualizer enable parameter used by
+    AUDPROC_MODULE_ID_VIRTUALIZER.
+*/
+#define AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE                      0x000108A6
+/** ID of the Virtualizer strength parameter used by
+    AUDPROC_MODULE_ID_VIRTUALIZER.
+*/
+#define AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH                    0x000108A7
+/** ID of the Virtualizer out type parameter used by
+    AUDPROC_MODULE_ID_VIRTUALIZER.
+*/
+#define AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE                    0x000108A8
+/** ID of the Virtualizer out type parameter used by
+    AUDPROC_MODULE_ID_VIRTUALIZER.
+*/
+#define AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST                 0x000108A9
+
+/** ID of the Reverb module. This module supports the following
+    parameter IDs:
+    - #AUDPROC_PARAM_ID_REVERB_ENABLE
+    - #AUDPROC_PARAM_ID_REVERB_MODE
+    - #AUDPROC_PARAM_ID_REVERB_PRESET
+    - #AUDPROC_PARAM_ID_REVERB_WET_MIX
+    - #AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST
+    - #AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL
+    - #AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL
+    - #AUDPROC_PARAM_ID_REVERB_DECAY_TIME
+    - #AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO
+    - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL
+    - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY
+    - #AUDPROC_PARAM_ID_REVERB_LEVEL
+    - #AUDPROC_PARAM_ID_REVERB_DELAY
+    - #AUDPROC_PARAM_ID_REVERB_DIFFUSION
+    - #AUDPROC_PARAM_ID_REVERB_DENSITY
+*/
+#define AUDPROC_MODULE_ID_REVERB                          0x000108AA
+/** ID of the Reverb enable parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_ENABLE                    0x000108AB
+/** ID of the Reverb mode parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_MODE                      0x000108AC
+/** ID of the Reverb preset parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_PRESET                    0x000108AD
+/** ID of the Reverb wet mix parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_WET_MIX                   0x000108AE
+/** ID of the Reverb gain adjust parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST               0x000108AF
+/** ID of the Reverb room level parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL                0x000108B0
+/** ID of the Reverb room hf level parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL             0x000108B1
+/** ID of the Reverb decay time parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_DECAY_TIME                0x000108B2
+/** ID of the Reverb decay hf ratio parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO            0x000108B3
+/** ID of the Reverb reflections level parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL         0x000108B4
+/** ID of the Reverb reflections delay parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY         0x000108B5
+/** ID of the Reverb level parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_LEVEL                      0x000108B6
+/** ID of the Reverb delay parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_DELAY                      0x000108B7
+/** ID of the Reverb diffusion parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_DIFFUSION                  0x000108B8
+/** ID of the Reverb density parameter used by
+    AUDPROC_MODULE_ID_REVERB.
+*/
+#define AUDPROC_PARAM_ID_REVERB_DENSITY                    0x000108B9
+
+/** ID of the Popless Equalizer module. This module supports the
+    following parameter IDs:
+    - #AUDPROC_PARAM_ID_EQ_ENABLE
+    - #AUDPROC_PARAM_ID_EQ_CONFIG
+    - #AUDPROC_PARAM_ID_EQ_NUM_BANDS
+    - #AUDPROC_PARAM_ID_EQ_BAND_LEVELS
+    - #AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE
+    - #AUDPROC_PARAM_ID_EQ_BAND_FREQS
+    - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE
+    - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ
+    - #AUDPROC_PARAM_ID_EQ_BAND_INDEX
+    - #AUDPROC_PARAM_ID_EQ_PRESET_ID
+    - #AUDPROC_PARAM_ID_EQ_NUM_PRESETS
+    - #AUDPROC_PARAM_ID_EQ_GET_PRESET_NAME
+*/
+#define AUDPROC_MODULE_ID_POPLESS_EQUALIZER                    0x000108BA
+/** ID of the Popless Equalizer enable parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+*/
+#define AUDPROC_PARAM_ID_EQ_ENABLE                             0x000108BB
+/** ID of the Popless Equalizer config parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+*/
+#define AUDPROC_PARAM_ID_EQ_CONFIG                             0x000108BC
+/** ID of the Popless Equalizer number of bands parameter used
+    by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+    used for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_NUM_BANDS                          0x000108BD
+/** ID of the Popless Equalizer band levels parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+    used for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_BAND_LEVELS                        0x000108BE
+/** ID of the Popless Equalizer band level range parameter used
+    by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+    used for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE                   0x000108BF
+/** ID of the Popless Equalizer band frequencies parameter used
+    by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+    used for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_BAND_FREQS                         0x000108C0
+/** ID of the Popless Equalizer single band frequency range
+    parameter used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+    This param ID is used for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE             0x000108C1
+/** ID of the Popless Equalizer single band frequency parameter
+    used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID
+    is used for set param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ                   0x000108C2
+/** ID of the Popless Equalizer band index parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+*/
+#define AUDPROC_PARAM_ID_EQ_BAND_INDEX                         0x000108C3
+/** ID of the Popless Equalizer preset id parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+    for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_PRESET_ID                          0x000108C4
+/** ID of the Popless Equalizer number of presets parameter used
+    by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+    for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_NUM_PRESETS                        0x000108C5
+/** ID of the Popless Equalizer preset name parameter used by
+    AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+    for get param only.
+*/
+#define AUDPROC_PARAM_ID_EQ_PRESET_NAME                        0x000108C6
+
+/* Set Q6 topologies */
+#define ASM_CMD_ADD_TOPOLOGIES				0x00010DBE
+#define ADM_CMD_ADD_TOPOLOGIES				0x00010335
+#define AFE_CMD_ADD_TOPOLOGIES				0x000100f8
+/* structure used for both ioctls */
+struct cmd_set_topologies {
+	struct apr_hdr hdr;
+	u32		payload_addr_lsw;
+	/* LSW of parameter data payload address.*/
+	u32		payload_addr_msw;
+	/* MSW of parameter data payload address.*/
+	u32		mem_map_handle;
+	/* Memory map handle returned by mem map command */
+	u32		payload_size;
+	/* Size in bytes of the variable payload in shared memory */
+} __packed;
+
+/* This module represents the Rx processing of Feedback speaker protection.
+ * It contains the excursion control, thermal protection,
+ * analog clip manager features in it.
+ * This module id will support following param ids.
+ * - AFE_PARAM_ID_FBSP_MODE_RX_CFG
+ */
+
+#define AFE_MODULE_FB_SPKR_PROT_RX 0x0001021C
+#define AFE_MODULE_FB_SPKR_PROT_V2_RX 0x0001025F
+
+#define AFE_PARAM_ID_FBSP_MODE_RX_CFG 0x0001021D
+#define AFE_PARAM_ID_FBSP_PTONE_RAMP_CFG 0x00010260
+
+struct asm_fbsp_mode_rx_cfg {
+	uint32_t minor_version;
+	uint32_t mode;
+} __packed;
+
+/* This module represents the VI processing of feedback speaker protection.
+ * It will receive Vsens and Isens from codec and generates necessary
+ * parameters needed by Rx processing.
+ * This module id will support following param ids.
+ * - AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG
+ * - AFE_PARAM_ID_CALIB_RES_CFG
+ * - AFE_PARAM_ID_FEEDBACK_PATH_CFG
+ */
+
+#define AFE_MODULE_FB_SPKR_PROT_VI_PROC 0x00010226
+#define AFE_MODULE_FB_SPKR_PROT_VI_PROC_V2 0x0001026A
+
+#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG 0x0001022A
+#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG_V2  0x0001026B
+
+struct asm_spkr_calib_vi_proc_cfg {
+	uint32_t minor_version;
+	uint32_t operation_mode;
+	uint32_t r0_t0_selection_flag[SP_V2_NUM_MAX_SPKR];
+	int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
+	int16_t	t0_cali_q6[SP_V2_NUM_MAX_SPKR];
+	uint32_t quick_calib_flag;
+} __packed;
+
+#define AFE_PARAM_ID_CALIB_RES_CFG 0x0001022B
+#define AFE_PARAM_ID_CALIB_RES_CFG_V2 0x0001026E
+
+struct asm_calib_res_cfg {
+	uint32_t minor_version;
+	int32_t	r0_cali_q24[SP_V2_NUM_MAX_SPKR];
+	uint32_t th_vi_ca_state;
+} __packed;
+
+#define AFE_PARAM_ID_FEEDBACK_PATH_CFG 0x0001022C
+#define AFE_MODULE_FEEDBACK 0x00010257
+
+struct asm_feedback_path_cfg {
+	uint32_t minor_version;
+	int32_t	dst_portid;
+	int32_t	num_channels;
+	int32_t	chan_info[4];
+} __packed;
+
+#define AFE_PARAM_ID_MODE_VI_PROC_CFG 0x00010227
+
+struct asm_mode_vi_proc_cfg {
+	uint32_t minor_version;
+	uint32_t cal_mode;
+} __packed;
+
+union afe_spkr_prot_config {
+	struct asm_fbsp_mode_rx_cfg mode_rx_cfg;
+	struct asm_spkr_calib_vi_proc_cfg vi_proc_cfg;
+	struct asm_feedback_path_cfg feedback_path_cfg;
+	struct asm_mode_vi_proc_cfg mode_vi_proc_cfg;
+} __packed;
+
+struct afe_spkr_prot_config_command {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2 pdata;
+	union afe_spkr_prot_config prot_config;
+} __packed;
+
+struct afe_spkr_prot_get_vi_calib {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_get_param_v2 get_param;
+	struct afe_port_param_data_v2 pdata;
+	struct asm_calib_res_cfg res_cfg;
+} __packed;
+
+struct afe_spkr_prot_calib_get_resp {
+	uint32_t status;
+	struct afe_port_param_data_v2 pdata;
+	struct asm_calib_res_cfg res_cfg;
+} __packed;
+
+
+/* SRS TRUMEDIA start */
+/* topology */
+#define SRS_TRUMEDIA_TOPOLOGY_ID			0x00010D90
+/* module */
+#define SRS_TRUMEDIA_MODULE_ID				0x10005010
+/* parameters */
+#define SRS_TRUMEDIA_PARAMS				0x10005011
+#define SRS_TRUMEDIA_PARAMS_WOWHD			0x10005012
+#define SRS_TRUMEDIA_PARAMS_CSHP			0x10005013
+#define SRS_TRUMEDIA_PARAMS_HPF				0x10005014
+#define SRS_TRUMEDIA_PARAMS_AEQ				0x10005015
+#define SRS_TRUMEDIA_PARAMS_HL				0x10005016
+#define SRS_TRUMEDIA_PARAMS_GEQ				0x10005017
+
+#define SRS_ID_GLOBAL	0x00000001
+#define SRS_ID_WOWHD	0x00000002
+#define SRS_ID_CSHP	0x00000003
+#define SRS_ID_HPF	0x00000004
+#define SRS_ID_AEQ	0x00000005
+#define SRS_ID_HL		0x00000006
+#define SRS_ID_GEQ	0x00000007
+
+#define SRS_CMD_UPLOAD		0x7FFF0000
+#define SRS_PARAM_OFFSET_MASK	0x3FFF0000
+#define SRS_PARAM_VALUE_MASK	0x0000FFFF
+
+struct srs_trumedia_params_GLOBAL {
+	uint8_t                  v1;
+	uint8_t                  v2;
+	uint8_t                  v3;
+	uint8_t                  v4;
+	uint8_t                  v5;
+	uint8_t                  v6;
+	uint8_t                  v7;
+	uint8_t                  v8;
+	uint16_t                 v9;
+} __packed;
+
+struct srs_trumedia_params_WOWHD {
+	uint32_t				v1;
+	uint16_t				v2;
+	uint16_t				v3;
+	uint16_t				v4;
+	uint16_t				v5;
+	uint16_t				v6;
+	uint16_t				v7;
+	uint16_t				v8;
+	uint16_t				v____A1;
+	uint32_t				v9;
+	uint16_t				v10;
+	uint16_t				v11;
+	uint32_t				v12[16];
+	uint32_t	v13[16];
+	uint32_t	v14[16];
+	uint32_t	v15[16];
+	uint32_t	v16;
+	uint16_t	v17;
+	uint16_t	v18;
+} __packed;
+
+struct srs_trumedia_params_CSHP {
+	uint32_t		v1;
+	uint16_t		v2;
+	uint16_t		v3;
+	uint16_t		v4;
+	uint16_t		v5;
+	uint16_t		v6;
+	uint16_t		v____A1;
+	uint32_t		v7;
+	uint16_t		v8;
+	uint16_t		v9;
+	uint32_t		v10[16];
+} __packed;
+
+struct srs_trumedia_params_HPF {
+	uint32_t		v1;
+	uint32_t		v2[26];
+} __packed;
+
+struct srs_trumedia_params_AEQ {
+	uint32_t		v1;
+	uint16_t		v2;
+	uint16_t		v3;
+	uint16_t		v4;
+	uint16_t		v____A1;
+	uint32_t	v5[74];
+	uint32_t	v6[74];
+	uint16_t	v7[2048];
+} __packed;
+
+struct srs_trumedia_params_HL {
+	uint16_t		v1;
+	uint16_t		v2;
+	uint16_t		v3;
+	uint16_t		v____A1;
+	int32_t			v4;
+	uint32_t		v5;
+	uint16_t		v6;
+	uint16_t		v____A2;
+	uint32_t		v7;
+} __packed;
+
+struct srs_trumedia_params_GEQ {
+	int16_t		v1[10];
+} __packed;
+struct srs_trumedia_params {
+	struct srs_trumedia_params_GLOBAL	global;
+	struct srs_trumedia_params_WOWHD	wowhd;
+	struct srs_trumedia_params_CSHP		cshp;
+	struct srs_trumedia_params_HPF		hpf;
+	struct srs_trumedia_params_AEQ		aeq;
+	struct srs_trumedia_params_HL		hl;
+	struct srs_trumedia_params_GEQ		geq;
+} __packed;
+/* SRS TruMedia end */
+
+#define AUDPROC_PARAM_ID_ENABLE		0x00010904
+#define ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF
+/* DTS Eagle */
+#define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C
+#define AUDPROC_MODULE_ID_DTS_HPX_POSTMIX 0x0001077B
+#define ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX 0x00010DED
+#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS  0x10015000
+#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER  0x10015001
+struct asm_dts_eagle_param {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_set_pp_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+} __packed;
+
+struct asm_dts_eagle_param_get {
+	struct apr_hdr	hdr;
+	struct asm_stream_cmd_get_pp_params_v2 param;
+} __packed;
+
+/* LSM Specific */
+#define VW_FEAT_DIM					(39)
+
+#define APRV2_IDS_SERVICE_ID_ADSP_LSM_V			(0xD)
+#define APRV2_IDS_DOMAIN_ID_ADSP_V			(0x4)
+#define APRV2_IDS_DOMAIN_ID_APPS_V			(0x5)
+
+#define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS		(0x00012A7F)
+#define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS	(0x00012A80)
+#define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS	(0x00012A81)
+#define LSM_SESSION_CMD_OPEN_TX				(0x00012A82)
+#define LSM_SESSION_CMD_CLOSE_TX			(0x00012A88)
+#define LSM_SESSION_CMD_SET_PARAMS			(0x00012A83)
+#define LSM_SESSION_CMD_SET_PARAMS_V2			(0x00012A8F)
+#define LSM_SESSION_CMD_REGISTER_SOUND_MODEL		(0x00012A84)
+#define LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL		(0x00012A85)
+#define LSM_SESSION_CMD_START				(0x00012A86)
+#define LSM_SESSION_CMD_STOP				(0x00012A87)
+#define LSM_SESSION_CMD_EOB				(0x00012A89)
+#define LSM_SESSION_CMD_READ				(0x00012A8A)
+#define LSM_SESSION_CMD_OPEN_TX_V2			(0x00012A8B)
+#define LSM_CMD_ADD_TOPOLOGIES				(0x00012A8C)
+
+#define LSM_SESSION_EVENT_DETECTION_STATUS		(0x00012B00)
+#define LSM_SESSION_EVENT_DETECTION_STATUS_V2		(0x00012B01)
+#define LSM_DATA_EVENT_READ_DONE			(0x00012B02)
+#define LSM_DATA_EVENT_STATUS				(0x00012B03)
+
+#define LSM_MODULE_ID_VOICE_WAKEUP			(0x00012C00)
+#define LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD		(0x00012C01)
+#define LSM_PARAM_ID_OPERATION_MODE			(0x00012C02)
+#define LSM_PARAM_ID_GAIN				(0x00012C03)
+#define LSM_PARAM_ID_CONNECT_TO_PORT			(0x00012C04)
+#define LSM_PARAM_ID_FEATURE_COMPENSATION_DATA		(0x00012C07)
+#define LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS		(0x00012C07)
+#define LSM_MODULE_ID_LAB				(0x00012C08)
+#define LSM_PARAM_ID_LAB_ENABLE				(0x00012C09)
+#define LSM_PARAM_ID_LAB_CONFIG				(0x00012C0A)
+#define LSM_MODULE_ID_FRAMEWORK				(0x00012C0E)
+
+/* HW MAD specific */
+#define AFE_MODULE_HW_MAD				(0x00010230)
+#define AFE_PARAM_ID_HW_MAD_CFG				(0x00010231)
+#define AFE_PARAM_ID_HW_MAD_CTRL			(0x00010232)
+#define AFE_PARAM_ID_SLIMBUS_SLAVE_PORT_CFG		(0x00010233)
+
+/* SW MAD specific */
+#define AFE_MODULE_SW_MAD				(0x0001022D)
+#define AFE_PARAM_ID_SW_MAD_CFG				(0x0001022E)
+#define AFE_PARAM_ID_SVM_MODEL				(0x0001022F)
+
+/* Commands/Params to pass the codec/slimbus data to DSP */
+#define AFE_SVC_CMD_SET_PARAM				(0x000100f3)
+#define AFE_MODULE_CDC_DEV_CFG				(0x00010234)
+#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG		(0x00010235)
+#define AFE_PARAM_ID_CDC_REG_CFG			(0x00010236)
+#define AFE_PARAM_ID_CDC_REG_CFG_INIT			(0x00010237)
+#define AFE_PARAM_ID_CDC_REG_PAGE_CFG                   (0x00010296)
+
+#define AFE_MAX_CDC_REGISTERS_TO_CONFIG			(20)
+
+/* AANC Port Config Specific */
+#define AFE_PARAM_ID_AANC_PORT_CONFIG			(0x00010215)
+#define AFE_API_VERSION_AANC_PORT_CONFIG		(0x1)
+#define AANC_TX_MIC_UNUSED				(0)
+#define AANC_TX_VOICE_MIC				(1)
+#define AANC_TX_ERROR_MIC				(2)
+#define AANC_TX_NOISE_MIC				(3)
+#define AFE_PORT_MAX_CHANNEL_CNT			(8)
+#define AFE_MODULE_AANC					(0x00010214)
+#define AFE_PARAM_ID_CDC_AANC_VERSION			(0x0001023A)
+#define AFE_API_VERSION_CDC_AANC_VERSION		(0x1)
+#define AANC_HW_BLOCK_VERSION_1				(1)
+#define AANC_HW_BLOCK_VERSION_2				(2)
+
+/*Clip bank selection*/
+#define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1
+#define AFE_CLIP_MAX_BANKS		4
+#define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242
+
+struct afe_param_aanc_port_cfg {
+	/* Minor version used for tracking the version of the module's
+	* source port configuration.
+	*/
+	uint32_t aanc_port_cfg_minor_version;
+
+	/* Sampling rate of the source Tx port. 8k - 192k*/
+	uint32_t tx_port_sample_rate;
+
+	/* Channel mapping for the Tx port signal carrying Noise (X),
+	* Error (E), and Voice (V) signals.
+	*/
+	uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT];
+
+	/* Number of channels on the source Tx port. */
+	uint16_t tx_port_num_channels;
+
+	/* Port ID of the Rx path reference signal. */
+	uint16_t rx_path_ref_port_id;
+
+	/* Sampling rate of the reference port. 8k - 192k*/
+	uint32_t ref_port_sample_rate;
+} __packed;
+
+struct afe_param_id_cdc_aanc_version {
+	/* Minor version used for tracking the version of the module's
+	* hw version
+	*/
+	uint32_t cdc_aanc_minor_version;
+
+	/* HW version. */
+	uint32_t aanc_hw_version;
+} __packed;
+
+struct afe_param_id_clip_bank_sel {
+	/* Minor version used for tracking the version of the module's
+	* hw version
+	*/
+	uint32_t minor_version;
+
+	/* Number of banks to be read */
+	uint32_t num_banks;
+
+	uint32_t bank_map[AFE_CLIP_MAX_BANKS];
+} __packed;
+
+/* ERROR CODES */
+/* Success. The operation completed with no errors. */
+#define ADSP_EOK          0x00000000
+/* General failure. */
+#define ADSP_EFAILED      0x00000001
+/* Bad operation parameter. */
+#define ADSP_EBADPARAM    0x00000002
+/* Unsupported routine or operation. */
+#define ADSP_EUNSUPPORTED 0x00000003
+/* Unsupported version. */
+#define ADSP_EVERSION     0x00000004
+/* Unexpected problem encountered. */
+#define ADSP_EUNEXPECTED  0x00000005
+/* Unhandled problem occurred. */
+#define ADSP_EPANIC       0x00000006
+/* Unable to allocate resource. */
+#define ADSP_ENORESOURCE  0x00000007
+/* Invalid handle. */
+#define ADSP_EHANDLE      0x00000008
+/* Operation is already processed. */
+#define ADSP_EALREADY     0x00000009
+/* Operation is not ready to be processed. */
+#define ADSP_ENOTREADY    0x0000000A
+/* Operation is pending completion. */
+#define ADSP_EPENDING     0x0000000B
+/* Operation could not be accepted or processed. */
+#define ADSP_EBUSY        0x0000000C
+/* Operation aborted due to an error. */
+#define ADSP_EABORTED     0x0000000D
+/* Operation preempted by a higher priority. */
+#define ADSP_EPREEMPTED   0x0000000E
+/* Operation requests intervention to complete. */
+#define ADSP_ECONTINUE    0x0000000F
+/* Operation requests immediate intervention to complete. */
+#define ADSP_EIMMEDIATE   0x00000010
+/* Operation is not implemented. */
+#define ADSP_ENOTIMPL     0x00000011
+/* Operation needs more data or resources. */
+#define ADSP_ENEEDMORE    0x00000012
+/* Operation does not have memory. */
+#define ADSP_ENOMEMORY    0x00000014
+/* Item does not exist. */
+#define ADSP_ENOTEXIST    0x00000015
+/* Max count for adsp error code sent to HLOS*/
+#define ADSP_ERR_MAX      (ADSP_ENOTEXIST + 1)
+/* Operation is finished. */
+#define ADSP_ETERMINATED    0x00011174
+
+/*bharath, adsp_error_codes.h */
+
+/* LPASS clock for I2S Interface */
+
+/* Supported OSR clock values */
+#define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ		0xBB8000
+#define Q6AFE_LPASS_OSR_CLK_9_P600_MHZ		0x927C00
+#define Q6AFE_LPASS_OSR_CLK_8_P192_MHZ		0x7D0000
+#define Q6AFE_LPASS_OSR_CLK_6_P144_MHZ		0x5DC000
+#define Q6AFE_LPASS_OSR_CLK_4_P096_MHZ		0x3E8000
+#define Q6AFE_LPASS_OSR_CLK_3_P072_MHZ		0x2EE000
+#define Q6AFE_LPASS_OSR_CLK_2_P048_MHZ		0x1F4000
+#define Q6AFE_LPASS_OSR_CLK_1_P536_MHZ		0x177000
+#define Q6AFE_LPASS_OSR_CLK_1_P024_MHZ		 0xFA000
+#define Q6AFE_LPASS_OSR_CLK_768_kHZ		 0xBB800
+#define Q6AFE_LPASS_OSR_CLK_512_kHZ		 0x7D000
+#define Q6AFE_LPASS_OSR_CLK_DISABLE		     0x0
+
+/* Supported Bit clock values */
+#define Q6AFE_LPASS_IBIT_CLK_12_P288_MHZ	0xBB8000
+#define Q6AFE_LPASS_IBIT_CLK_11_P2896_MHZ	0xAC4400
+#define Q6AFE_LPASS_IBIT_CLK_8_P192_MHZ		0x7D0000
+#define Q6AFE_LPASS_IBIT_CLK_6_P144_MHZ		0x5DC000
+#define Q6AFE_LPASS_IBIT_CLK_4_P096_MHZ		0x3E8000
+#define Q6AFE_LPASS_IBIT_CLK_3_P072_MHZ		0x2EE000
+#define Q6AFE_LPASS_IBIT_CLK_2_P048_MHZ		0x1F4000
+#define Q6AFE_LPASS_IBIT_CLK_1_P536_MHZ		0x177000
+#define Q6AFE_LPASS_IBIT_CLK_1_P024_MHZ		 0xFA000
+#define Q6AFE_LPASS_IBIT_CLK_768_KHZ		 0xBB800
+#define Q6AFE_LPASS_IBIT_CLK_512_KHZ		 0x7D000
+#define Q6AFE_LPASS_IBIT_CLK_256_KHZ		 0x3E800
+#define Q6AFE_LPASS_IBIT_CLK_DISABLE		     0x0
+
+/* Supported LPASS CLK sources */
+#define Q6AFE_LPASS_CLK_SRC_EXTERNAL 0
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+
+/* Supported LPASS CLK root*/
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
+
+enum afe_lpass_clk_mode {
+	Q6AFE_LPASS_MODE_BOTH_INVALID,
+	Q6AFE_LPASS_MODE_CLK1_VALID,
+	Q6AFE_LPASS_MODE_CLK2_VALID,
+	Q6AFE_LPASS_MODE_BOTH_VALID,
+} __packed;
+
+/* Clock ID Enumeration Define. */
+/* Clock ID for Primary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT                          0x100
+/* Clock ID for Primary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT                          0x101
+/* Clock ID for Secondary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT                          0x102
+/* Clock ID for Secondary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT                          0x103
+/* Clock ID for Tertiary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT                          0x104
+/* Clock ID for Tertiary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT                          0x105
+/* Clock ID for Quartnery I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT                         0x106
+/* Clock ID for Quartnery I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT                         0x107
+/* Clock ID for Speaker I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT                       0x108
+/* Clock ID for Speaker I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT                       0x109
+/* Clock ID for Speaker I2S OSR */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR                        0x10A
+
+/* Clock ID for QUINARY  I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT			0x10B
+/* Clock ID for QUINARY  I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT			0x10C
+/* Clock ID for SENARY  I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT			0x10D
+/* Clock ID for SENARY  I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT			0x10E
+
+/* Clock ID for Primary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT                           0x200
+/* Clock ID for Primary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT                           0x201
+/* Clock ID for Secondary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT                           0x202
+/* Clock ID for Secondary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT                           0x203
+/* Clock ID for Tertiary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT                           0x204
+/* Clock ID for Tertiary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT                           0x205
+/* Clock ID for Quartery PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT                          0x206
+/* Clock ID for Quartery PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT                          0x207
+
+/** Clock ID for Primary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT                           0x200
+/** Clock ID for Primary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT                           0x201
+/** Clock ID for Secondary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT                           0x202
+/** Clock ID for Secondary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT                           0x203
+/** Clock ID for Tertiary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT                           0x204
+/** Clock ID for Tertiary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT                           0x205
+/** Clock ID for Quartery TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT                          0x206
+/** Clock ID for Quartery TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT                          0x207
+
+/* Clock ID for MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_1                                 0x300
+/* Clock ID for MCLK2 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_2                                 0x301
+/* Clock ID for MCLK3 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_3                                 0x302
+/* Clock ID for Internal Digital Codec Core */
+#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE            0x303
+
+/* Clock ID for AHB HDMI input */
+#define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT                         0x400
+
+/* Clock ID for SPDIF core */
+#define Q6AFE_LPASS_CLK_ID_SPDIF_CORE                             0x500
+
+
+/* Clock attribute for invalid use (reserved for internal usage) */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID		0x0
+/* Clock attribute for no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO		0x1
+/* Clock attribute for dividend couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND	0x2
+/* Clock attribute for divisor couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR	0x3
+/* Clock attribute for invert and no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO	0x4
+/* Clock set API version */
+#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION		0x1
+
+struct afe_clk_set {
+	/*
+	 * Minor version used for tracking clock set.
+	 *	@values #AFE_API_VERSION_CLOCK_SET
+	 */
+	uint32_t clk_set_minor_version;
+
+	/*
+	 * Clock ID
+	 *	@values
+	 *	- 0x100 to 0x10A - MSM8996
+	 *	- 0x200 to 0x207 - MSM8996
+	 *	- 0x300 to 0x302 - MSM8996 @tablebulletend
+	 */
+	uint32_t clk_id;
+
+	/*
+	 * Clock frequency  (in Hertz) to be set.
+	 *	@values
+	 *	- >= 0 for clock frequency to set @tablebulletend
+	 */
+	uint32_t clk_freq_in_hz;
+
+	/* Use to specific divider for two clocks if needed.
+	 *	Set to Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO for no divider
+	 *	relation clocks
+	 *	@values
+	 *	- #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO
+	 *	- #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND
+	 *	- #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR @tablebulletend
+	 */
+	uint16_t clk_attri;
+
+	/*
+	 * Specifies the root clock source.
+	 *	Currently, only Q6AFE_LPASS_CLK_ROOT_DEFAULT is valid
+	 *	@values
+	 *	- 0 @tablebulletend
+	 */
+	uint16_t clk_root;
+
+	/*
+	 * for enable and disable clock.
+	 *	"clk_freq_in_hz", "clk_attri", and "clk_root"
+	 *	are ignored in disable clock case.
+	 *	@values 
+	 *	- 0 -- Disabled
+	 *	- 1 -- Enabled  @tablebulletend
+	 */
+	uint32_t enable;
+};
+
+struct afe_clk_cfg {
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+	u32                  i2s_cfg_minor_version;
+
+/* clk value 1 in MHz. */
+	u32                  clk_val1;
+
+/* clk value 2 in MHz. */
+	u32                  clk_val2;
+
+/* clk_src
+ * #Q6AFE_LPASS_CLK_SRC_EXTERNAL
+ * #Q6AFE_LPASS_CLK_SRC_INTERNAL
+ */
+
+	u16                  clk_src;
+
+/* clk_root -0 for default */
+	u16                  clk_root;
+
+/* clk_set_mode
+ * #Q6AFE_LPASS_MODE_BOTH_INVALID
+ * #Q6AFE_LPASS_MODE_CLK1_VALID
+ * #Q6AFE_LPASS_MODE_CLK2_VALID
+ * #Q6AFE_LPASS_MODE_BOTH_VALID
+ */
+	u16                  clk_set_mode;
+
+/* This param id is used to configure I2S clk */
+	u16                  reserved;
+} __packed;
+
+/* This param id is used to configure I2S clk */
+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG	0x00010238
+#define AFE_MODULE_CLOCK_SET		0x0001028F
+#define AFE_PARAM_ID_CLOCK_SET		0x00010290
+
+struct afe_lpass_clk_config_command {
+	struct apr_hdr			 hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	struct afe_clk_cfg clk_cfg;
+} __packed;
+
+enum afe_lpass_digital_clk_src {
+	Q6AFE_LPASS_DIGITAL_ROOT_INVALID,
+	Q6AFE_LPASS_DIGITAL_ROOT_PRI_MI2S_OSR,
+	Q6AFE_LPASS_DIGITAL_ROOT_SEC_MI2S_OSR,
+	Q6AFE_LPASS_DIGITAL_ROOT_TER_MI2S_OSR,
+	Q6AFE_LPASS_DIGITAL_ROOT_QUAD_MI2S_OSR,
+	Q6AFE_LPASS_DIGITAL_ROOT_CDC_ROOT_CLK,
+} __packed;
+
+/* This param id is used to configure internal clk */
+#define AFE_PARAM_ID_INTERNAL_DIGIATL_CDC_CLK_CONFIG	0x00010239
+
+struct afe_digital_clk_cfg {
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+	u32                  i2s_cfg_minor_version;
+
+/* clk value in MHz. */
+	u32                  clk_val;
+
+/*	INVALID
+ *	PRI_MI2S_OSR
+ *	SEC_MI2S_OSR
+ *	TER_MI2S_OSR
+ *	QUAD_MI2S_OSR
+ *	DIGT_CDC_ROOT
+ */
+	u16                  clk_root;
+
+/* This field must be set to zero. */
+	u16                  reserved;
+} __packed;
+
+
+struct afe_lpass_digital_clk_config_command {
+	struct apr_hdr			 hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2    pdata;
+	struct afe_digital_clk_cfg clk_cfg;
+} __packed;
+
+/*
+ * Opcode for AFE to start DTMF.
+ */
+#define AFE_PORTS_CMD_DTMF_CTL	0x00010102
+
+/** DTMF payload.*/
+struct afe_dtmf_generation_command {
+	struct apr_hdr hdr;
+
+	/*
+	 * Duration of the DTMF tone in ms.
+	 * -1      -> continuous,
+	 *  0      -> disable
+	 */
+	int64_t                   duration_in_ms;
+
+	/*
+	 * The DTMF high tone frequency.
+	 */
+	uint16_t                  high_freq;
+
+	/*
+	 * The DTMF low tone frequency.
+	 */
+	uint16_t                  low_freq;
+
+	/*
+	 * The DTMF volume setting
+	 */
+	uint16_t                  gain;
+
+	/*
+	 * The number of ports to enable/disable on.
+	 */
+	uint16_t                  num_ports;
+
+	/*
+	 * The Destination ports - array  .
+	 * For DTMF on multiple ports, portIds needs to
+	 * be populated numPorts times.
+	 */
+	uint16_t                  port_ids;
+
+	/*
+	 * variable for 32 bit alignment of APR packet.
+	 */
+	uint16_t                  reserved;
+} __packed;
+
+enum afe_config_type {
+	AFE_SLIMBUS_SLAVE_PORT_CONFIG,
+	AFE_SLIMBUS_SLAVE_CONFIG,
+	AFE_CDC_REGISTERS_CONFIG,
+	AFE_AANC_VERSION,
+	AFE_CDC_CLIP_REGISTERS_CONFIG,
+	AFE_CLIP_BANK_SEL,
+	AFE_CDC_REGISTER_PAGE_CONFIG,
+	AFE_MAX_CONFIG_TYPES,
+};
+
+struct afe_param_slimbus_slave_port_cfg {
+	uint32_t minor_version;
+	uint16_t slimbus_dev_id;
+	uint16_t slave_dev_pgd_la;
+	uint16_t slave_dev_intfdev_la;
+	uint16_t bit_width;
+	uint16_t data_format;
+	uint16_t num_channels;
+	uint16_t slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+} __packed;
+
+struct afe_param_cdc_slimbus_slave_cfg {
+	uint32_t minor_version;
+	uint32_t device_enum_addr_lsw;
+	uint32_t device_enum_addr_msw;
+	uint16_t tx_slave_port_offset;
+	uint16_t rx_slave_port_offset;
+} __packed;
+
+struct afe_param_cdc_reg_cfg {
+	uint32_t minor_version;
+	uint32_t reg_logical_addr;
+	uint32_t reg_field_type;
+	uint32_t reg_field_bit_mask;
+	uint16_t reg_bit_width;
+	uint16_t reg_offset_scale;
+} __packed;
+
+#define AFE_API_VERSION_CDC_REG_PAGE_CFG   1
+
+enum {
+	AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_0 = 0,
+	AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_1,
+	AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_2,
+	AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_3,
+};
+
+struct afe_param_cdc_reg_page_cfg {
+	uint32_t minor_version;
+	uint32_t enable;
+	uint32_t proc_id;
+} __packed;
+
+struct afe_param_cdc_reg_cfg_data {
+	uint32_t num_registers;
+	struct afe_param_cdc_reg_cfg *reg_data;
+} __packed;
+
+struct afe_svc_cmd_set_param {
+	uint32_t payload_size;
+	uint32_t payload_address_lsw;
+	uint32_t payload_address_msw;
+	uint32_t mem_map_handle;
+} __packed;
+
+struct afe_svc_param_data {
+	uint32_t module_id;
+	uint32_t param_id;
+	uint16_t param_size;
+	uint16_t reserved;
+} __packed;
+
+struct afe_param_hw_mad_ctrl {
+	uint32_t minor_version;
+	uint16_t mad_type;
+	uint16_t mad_enable;
+} __packed;
+
+struct afe_cmd_hw_mad_ctrl {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_hw_mad_ctrl payload;
+} __packed;
+
+struct afe_cmd_hw_mad_slimbus_slave_port_cfg {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_slimbus_slave_port_cfg sb_port_cfg;
+} __packed;
+
+struct afe_cmd_sw_mad_enable {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2 pdata;
+} __packed;
+
+struct afe_param_cdc_reg_cfg_payload {
+	struct afe_svc_param_data     common;
+	struct afe_param_cdc_reg_cfg  reg_cfg;
+} __packed;
+
+struct afe_lpass_clk_config_command_v2 {
+	struct apr_hdr			hdr;
+	struct afe_svc_cmd_set_param	param;
+	struct afe_svc_param_data	pdata;
+	struct afe_clk_set		clk_cfg;
+} __packed;
+
+/*
+ * reg_data's size can be up to AFE_MAX_CDC_REGISTERS_TO_CONFIG
+ */
+struct afe_svc_cmd_cdc_reg_cfg {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_param_cdc_reg_cfg_payload reg_data[0];
+} __packed;
+
+struct afe_svc_cmd_init_cdc_reg_cfg {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 init;
+} __packed;
+
+struct afe_svc_cmd_sb_slave_cfg {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_cdc_slimbus_slave_cfg sb_slave_cfg;
+} __packed;
+
+struct afe_svc_cmd_cdc_reg_page_cfg {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_cdc_reg_page_cfg cdc_reg_page_cfg;
+} __packed;
+
+struct afe_svc_cmd_cdc_aanc_version {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_id_cdc_aanc_version version;
+} __packed;
+
+struct afe_port_cmd_set_aanc_param {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+	struct afe_port_param_data_v2 pdata;
+	union {
+		struct afe_param_aanc_port_cfg aanc_port_cfg;
+		struct afe_mod_enable_param    mod_enable;
+	} __packed data;
+} __packed;
+
+struct afe_port_cmd_set_aanc_acdb_table {
+	struct apr_hdr hdr;
+	struct afe_port_cmd_set_param_v2 param;
+} __packed;
+
+/* Dolby DAP topology */
+#define DOLBY_ADM_COPP_TOPOLOGY_ID	0x0001033B
+#define DS2_ADM_COPP_TOPOLOGY_ID	0x1301033B
+
+/* RMS value from DSP */
+#define RMS_MODULEID_APPI_PASSTHRU  0x10009011
+#define RMS_PARAM_FIRST_SAMPLE 0x10009012
+#define RMS_PAYLOAD_LEN 4
+
+/* Customized mixing in matix mixer */
+#define MTMX_MODULE_ID_DEFAULT_CHMIXER  0x00010341
+#define DEFAULT_CHMIXER_PARAM_ID_COEFF  0x00010342
+#define CUSTOM_STEREO_PAYLOAD_SIZE	9
+#define CUSTOM_STEREO_CMD_PARAM_SIZE	24
+#define CUSTOM_STEREO_NUM_OUT_CH	0x0002
+#define CUSTOM_STEREO_NUM_IN_CH		0x0002
+#define CUSTOM_STEREO_INDEX_PARAM	0x0002
+#define Q14_GAIN_ZERO_POINT_FIVE	0x2000
+#define Q14_GAIN_UNITY			0x4000
+
+struct afe_svc_cmd_set_clip_bank_selection {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 pdata;
+	struct afe_param_id_clip_bank_sel bank_sel;
+} __packed;
+
+/* Ultrasound supported formats */
+#define US_POINT_EPOS_FORMAT_V2 0x0001272D
+#define US_RAW_FORMAT_V2        0x0001272C
+#define US_PROX_FORMAT_V4       0x0001273B
+#define US_RAW_SYNC_FORMAT      0x0001272F
+#define US_GES_SYNC_FORMAT      0x00012730
+
+#define AFE_MODULE_GROUP_DEVICE	0x00010254
+#define AFE_PARAM_ID_GROUP_DEVICE_CFG	0x00010255
+#define AFE_PARAM_ID_GROUP_DEVICE_ENABLE 0x00010256
+#define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX	0x1102
+
+/*  Payload of the #AFE_PARAM_ID_GROUP_DEVICE_CFG
+ * parameter, which configures max of 8 AFE ports
+ * into a group.
+ * The fixed size of this structure is sixteen bytes.
+ */
+struct afe_group_device_group_cfg {
+	u32 minor_version;
+	u16 group_id;
+	u16 num_channels;
+	u16 port_id[8];
+} __packed;
+
+#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX \
+	(AFE_PORT_ID_PRIMARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX \
+	(AFE_PORT_ID_PRIMARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX \
+	(AFE_PORT_ID_SECONDARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX \
+	(AFE_PORT_ID_SECONDARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX \
+	(AFE_PORT_ID_TERTIARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX \
+	(AFE_PORT_ID_TERTIARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX \
+	(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX \
+	(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x100)
+
+/** ID of the parameter used by #AFE_MODULE_GROUP_DEVICE to configure the
+	group device. #AFE_SVC_CMD_SET_PARAM can use this parameter ID.
+
+	Requirements:
+	- Configure the group before the member ports in the group are
+	configured and started.
+	- Enable the group only after it is configured.
+	- Stop all member ports in the group before disabling the group.
+*/
+#define AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG	0x0001029E
+
+/** Version information used to handle future additions to
+	AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG processing (for backward compatibility).
+ */
+#define AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG	0x1
+
+/** Number of AFE ports in group device  */
+#define AFE_GROUP_DEVICE_NUM_PORTS					8
+
+/* Payload of the AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG parameter ID
+	used by AFE_MODULE_GROUP_DEVICE.
+*/
+struct afe_param_id_group_device_tdm_cfg {
+	u32	group_device_cfg_minor_version;
+	/**< Minor version used to track group device configuration.
+	@values #AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG */
+
+	u16	group_id;
+	/**< ID for the group device.
+	@values
+	- #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX
+	- #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX
+	- #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX
+	- #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX
+	- #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX
+	- #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX
+	- #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX
+	- #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX */
+
+	u16	reserved;
+	/** 0 */
+
+	u16	port_id[AFE_GROUP_DEVICE_NUM_PORTS];
+	/**< Array of member port IDs of this group.
+	@values
+	- #AFE_PORT_ID_PRIMARY_TDM_RX
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_1
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_2
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_3
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_4
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_5
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_6
+	- #AFE_PORT_ID_PRIMARY_TDM_RX_7
+
+	- #AFE_PORT_ID_PRIMARY_TDM_TX
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_1
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_2
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_3
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_4
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_5
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_6
+	- #AFE_PORT_ID_PRIMARY_TDM_TX_7
+
+	- #AFE_PORT_ID_SECONDARY_TDM_RX
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_1
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_2
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_3
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_4
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_5
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_6
+	- #AFE_PORT_ID_SECONDARY_TDM_RX_7
+
+	- #AFE_PORT_ID_SECONDARY_TDM_TX
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_1
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_2
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_3
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_4
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_5
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_6
+	- #AFE_PORT_ID_SECONDARY_TDM_TX_7
+
+	- #AFE_PORT_ID_TERTIARY_TDM_RX
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_1
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_2
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_3
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_4
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_5
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_6
+	- #AFE_PORT_ID_TERTIARY_TDM_RX_7
+
+	- #AFE_PORT_ID_TERTIARY_TDM_TX
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_1
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_2
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_3
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_4
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_5
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_6
+	- #AFE_PORT_ID_TERTIARY_TDM_TX_7
+
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_1
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_2
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_3
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_4
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_5
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_6
+	- #AFE_PORT_ID_QUATERNARY_TDM_RX_7
+
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_1
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_2
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_3
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_4
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_5
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_6
+	- #AFE_PORT_ID_QUATERNARY_TDM_TX_7
+	@tablebulletend */
+
+	u32	num_channels;
+	/**< Number of enabled slots for TDM frame.
+	@values 1 to 8 */
+
+	u32	sample_rate;
+	/**< Sampling rate of the port.
+	@values
+	- #AFE_PORT_SAMPLE_RATE_8K
+	- #AFE_PORT_SAMPLE_RATE_16K
+	- #AFE_PORT_SAMPLE_RATE_24K
+	- #AFE_PORT_SAMPLE_RATE_32K
+	- #AFE_PORT_SAMPLE_RATE_48K @tablebulletend */
+
+	u32	bit_width;
+	/**< Bit width of the sample.
+	@values 16, 24, (32) */
+
+	u16	nslots_per_frame;
+	/**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
+	@values 1 - 32 */
+
+	u16	slot_width;
+	/**< Slot width of the slot in a TDM frame.  (slot_width >= bit_width)
+	have to be satisfied.
+	@values 16, 24, 32 */
+
+	u32	slot_mask;
+	/**< Position of active slots.  When that bit is set, that paricular
+	slot is active.
+	Number of active slots can be inferred by number of bits set in
+	the mask.  Only 8 individual bits can be enabled.
+	Bits 0..31 corresponding to slot 0..31
+	@values 1 to 2^32 -1 */
+} __packed;
+
+/*  Payload of the #AFE_PARAM_ID_GROUP_DEVICE_ENABLE
+ * parameter, which enables or
+ * disables any module.
+ * The fixed size of this structure is four bytes.
+ */
+
+struct afe_group_device_enable {
+	u16 group_id;
+	/* valid value is AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX */
+	u16 enable;
+	/* Enables (1) or disables (0) the module. */
+} __packed;
+
+union afe_port_group_config {
+	struct afe_group_device_group_cfg group_cfg;
+	struct afe_group_device_enable group_enable;
+	struct afe_param_id_group_device_tdm_cfg tdm_cfg;
+} __packed;
+
+struct afe_port_group_create {
+	struct apr_hdr hdr;
+	struct afe_svc_cmd_set_param param;
+	struct afe_port_param_data_v2 pdata;
+	union afe_port_group_config data;
+} __packed;
+
+/* Command for Matrix or Stream Router */
+#define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2    0x00010DCE
+/* Module for AVSYNC */
+#define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC    0x00010DC6
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
+ * render window start value. This parameter is supported only for a Set
+ * command (not a Get command) in the Rx direction
+ * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2).
+ * Render window start is a value (session time minus timestamp, or ST-TS)
+ * below which frames are held, and after which frames are immediately
+ * rendered.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
+ * render window end value. This parameter is supported only for a Set
+ * command (not a Get command) in the Rx direction
+ * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value
+ * (session time minus timestamp) above which frames are dropped, and below
+ * which frames are immediately rendered.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2   0x00010DD2
+
+/* Generic payload of the window parameters in the
+ * #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module.
+ * This payload is supported only for a Set command
+ * (not a Get command) on the Rx path.
+ */
+struct asm_session_mtmx_strtr_param_window_v2_t {
+	u32    window_lsw;
+	/* Lower 32 bits of the render window start value. */
+
+	u32    window_msw;
+	/* Upper 32 bits of the render window start value.
+
+	 * The 64-bit number formed by window_lsw and window_msw specifies a
+	 * signed 64-bit window value in microseconds. The sign extension is
+	 * necessary. This value is used by the following parameter IDs:
+	 * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2
+	 * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2
+	 * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2
+	 * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2
+	 * The value depends on which parameter ID is used.
+	 * The aDSP honors the windows at a granularity of 1 ms.
+	 */
+};
+
+struct asm_session_cmd_set_mtmx_strstr_params_v2 {
+	uint32_t                  data_payload_addr_lsw;
+	/* Lower 32 bits of the 64-bit data payload address. */
+
+	uint32_t                  data_payload_addr_msw;
+	/* Upper 32 bits of the 64-bit data payload address.
+	 * If the address is not sent (NULL), the message is in the payload.
+	 * If the address is sent (non-NULL), the parameter data payloads
+	 * begin at the specified address.
+	 */
+
+	uint32_t                  mem_map_handle;
+	/* Unique identifier for an address. This memory map handle is returned
+	 * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+	 * values
+	 * - NULL -- Parameter data payloads are within the message payload
+	 * (in-band).
+	 * - Non-NULL -- Parameter data payloads begin at the address specified
+	 * in the data_payload_addr_lsw and data_payload_addr_msw fields
+	 * (out-of-band).
+	 */
+
+	uint32_t                  data_payload_size;
+	/* Actual size of the variable payload accompanying the message, or in
+	 * shared memory. This field is used for parsing the parameter payload.
+	 * values > 0 bytes
+	 */
+
+	uint32_t                  direction;
+	/* Direction of the entity (matrix mixer or stream router) on which
+	 * the parameter is to be set.
+	 * values
+	 * - 0 -- Rx (for Rx stream router or Rx matrix mixer)
+	 * - 1 -- Tx (for Tx stream router or Tx matrix mixer)
+	 */
+};
+
+struct asm_mtmx_strtr_params {
+	struct apr_hdr  hdr;
+	struct asm_session_cmd_set_mtmx_strstr_params_v2 param;
+	struct asm_stream_param_data_v2 data;
+	u32 window_lsw;
+	u32 window_msw;
+} __packed;
+
+#define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF
+#define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0
+
+#define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B
+#define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL)
+
+struct asm_session_cmd_get_mtmx_strstr_params_v2 {
+	uint32_t                  data_payload_addr_lsw;
+	/* Lower 32 bits of the 64-bit data payload address. */
+
+	uint32_t                  data_payload_addr_msw;
+	/*
+	 * Upper 32 bits of the 64-bit data payload address.
+	 * If the address is not sent (NULL), the message is in the payload.
+	 * If the address is sent (non-NULL), the parameter data payloads
+	 * begin at the specified address.
+	 */
+
+	uint32_t                  mem_map_handle;
+	/*
+	 * Unique identifier for an address. This memory map handle is returned
+	 * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+	 * values
+	 * - NULL -- Parameter data payloads are within the message payload
+	 * (in-band).
+	 * - Non-NULL -- Parameter data payloads begin at the address specified
+	 * in the data_payload_addr_lsw and data_payload_addr_msw fields
+	 * (out-of-band).
+	 */
+	uint32_t                  direction;
+	/*
+	 * Direction of the entity (matrix mixer or stream router) on which
+	 * the parameter is to be set.
+	 * values
+	 * - 0 -- Rx (for Rx stream router or Rx matrix mixer)
+	 * - 1 -- Tx (for Tx stream router or Tx matrix mixer)
+	 */
+	uint32_t                  module_id;
+	/* Unique module ID. */
+
+	uint32_t                  param_id;
+	/* Unique parameter ID. */
+
+	uint32_t                  param_max_size;
+};
+
+struct asm_session_mtmx_strtr_param_session_time_v3_t {
+	uint32_t                  session_time_lsw;
+	/* Lower 32 bits of the current session time in microseconds */
+
+	uint32_t                  session_time_msw;
+	/*
+	 * Upper 32 bits of the current session time in microseconds.
+	 * The 64-bit number formed by session_time_lsw and session_time_msw
+	 * is treated as signed.
+	 */
+
+	uint32_t                  absolute_time_lsw;
+	/*
+	 * Lower 32 bits of the 64-bit absolute time in microseconds.
+	 * This is the time when the sample corresponding to the
+	 * session_time_lsw is rendered to the hardware. This absolute
+	 * time can be slightly in the future or past.
+	 */
+
+	uint32_t                  absolute_time_msw;
+	/*
+	 * Upper 32 bits of the 64-bit absolute time in microseconds.
+	 * This is the time when the sample corresponding to the
+	 * session_time_msw is rendered to hardware. This absolute
+	 * time can be slightly in the future or past. The 64-bit number
+	 * formed by absolute_time_lsw and absolute_time_msw is treated as
+	 * unsigned.
+	 */
+
+	uint32_t                  time_stamp_lsw;
+	/* Lower 32 bits of the last processed timestamp in microseconds */
+
+	uint32_t                  time_stamp_msw;
+	/*
+	 * Upper 32 bits of the last processed timestamp in microseconds.
+	 * The 64-bit number formed by time_stamp_lsw and time_stamp_lsw
+	 * is treated as unsigned.
+	 */
+
+	uint32_t                  flags;
+	/*
+	 * Keeps track of any additional flags needed.
+	 * @values{for bit 31}
+	 * - 0 -- Uninitialized/invalid
+	 * - 1 -- Valid
+	 * All other bits are reserved; clients must set them to zero.
+	 */
+};
+
+union asm_session_mtmx_strtr_data_type {
+	struct asm_session_mtmx_strtr_param_session_time_v3_t session_time;
+};
+
+struct asm_mtmx_strtr_get_params {
+	struct apr_hdr hdr;
+	struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info;
+} __packed;
+
+struct asm_mtmx_strtr_get_params_cmdrsp {
+	uint32_t err_code;
+	struct asm_stream_param_data_v2 param_info;
+	union asm_session_mtmx_strtr_data_type param_data;
+} __packed;
+
+#define AUDPROC_MODULE_ID_RESAMPLER 0x00010719
+
+enum {
+	LEGACY_PCM = 0,
+	COMPRESSED_PASSTHROUGH,
+	COMPRESSED_PASSTHROUGH_CONVERT,
+};
+
+#define AUDPROC_MODULE_ID_COMPRESSED_MUTE                0x00010770
+#define AUDPROC_PARAM_ID_COMPRESSED_MUTE                 0x00010771
+
+struct adm_set_compressed_device_mute {
+	struct adm_cmd_set_pp_params_v5 command;
+	struct adm_param_data_v5 params;
+	u32    mute_on;
+} __packed;
+
+#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY             0x0001076E
+#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY              0x0001076F
+
+struct adm_set_compressed_device_latency {
+	struct adm_cmd_set_pp_params_v5 command;
+	struct adm_param_data_v5 params;
+	u32    latency;
+} __packed;
+
+#define VOICEPROC_MODULE_ID_GENERIC_TX                      0x00010EF6
+#define VOICEPROC_PARAM_ID_FLUENCE_SOUNDFOCUS               0x00010E37
+#define VOICEPROC_PARAM_ID_FLUENCE_SOURCETRACKING           0x00010E38
+#define MAX_SECTORS                                         8
+#define MAX_NOISE_SOURCE_INDICATORS                         3
+#define MAX_POLAR_ACTIVITY_INDICATORS                       360
+
+struct sound_focus_param {
+	uint16_t start_angle[MAX_SECTORS];
+	uint8_t enable[MAX_SECTORS];
+	uint16_t gain_step;
+} __packed;
+
+struct source_tracking_param {
+	uint8_t vad[MAX_SECTORS];
+	uint16_t doa_speech;
+	uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
+	uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
+} __packed;
+
+struct adm_param_fluence_soundfocus_t {
+	uint16_t start_angles[MAX_SECTORS];
+	uint8_t enables[MAX_SECTORS];
+	uint16_t gain_step;
+	uint16_t reserved;
+} __packed;
+
+struct adm_set_fluence_soundfocus_param {
+	struct adm_cmd_set_pp_params_v5 params;
+	struct adm_param_data_v5 data;
+	struct adm_param_fluence_soundfocus_t soundfocus_data;
+} __packed;
+
+struct adm_param_fluence_sourcetracking_t {
+	uint8_t vad[MAX_SECTORS];
+	uint16_t doa_speech;
+	uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
+	uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
+} __packed;
+
+#define AUDPROC_MODULE_ID_AUDIOSPHERE               0x00010916
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE         0x00010917
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH       0x00010918
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE    0x00010919
+
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT         0x0001091A
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT   0x0001091B
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT         0x0001091C
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT   0x0001091D
+
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO  0x0001091E
+#endif /*_APR_AUDIO_V2_H_ */
diff --git a/include/sound/apr_audio.h b/include/sound/apr_audio.h
new file mode 100644
index 000000000000..4e6e2b8405ce
--- /dev/null
+++ b/include/sound/apr_audio.h
@@ -0,0 +1,1929 @@
+/*
+ *
+ * Copyright (c) 2010-2013, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+
+#ifndef _APR_AUDIO_H_
+#define _APR_AUDIO_H_
+
+/* ASM opcodes without APR payloads*/
+#include <linux/qdsp6v2/apr.h>
+
+/*
+ * Audio Front End (AFE)
+ */
+
+/* Port ID. Update afe_get_port_index when a new port is added here. */
+#define PRIMARY_I2S_RX 0		/* index = 0 */
+#define PRIMARY_I2S_TX 1		/* index = 1 */
+#define PCM_RX 2			/* index = 2 */
+#define PCM_TX 3			/* index = 3 */
+#define SECONDARY_I2S_RX 4		/* index = 4 */
+#define SECONDARY_I2S_TX 5		/* index = 5 */
+#define MI2S_RX 6			/* index = 6 */
+#define MI2S_TX 7			/* index = 7 */
+#define HDMI_RX 8			/* index = 8 */
+#define RSVD_2 9			/* index = 9 */
+#define RSVD_3 10			/* index = 10 */
+#define DIGI_MIC_TX 11			/* index = 11 */
+#define VOICE_RECORD_RX 0x8003		/* index = 12 */
+#define VOICE_RECORD_TX 0x8004		/* index = 13 */
+#define VOICE_PLAYBACK_TX 0x8005	/* index = 14 */
+
+/* Slimbus Multi channel port id pool  */
+#define SLIMBUS_0_RX		0x4000		/* index = 15 */
+#define SLIMBUS_0_TX		0x4001		/* index = 16 */
+#define SLIMBUS_1_RX		0x4002		/* index = 17 */
+#define SLIMBUS_1_TX		0x4003		/* index = 18 */
+#define SLIMBUS_2_RX		0x4004
+#define SLIMBUS_2_TX		0x4005
+#define SLIMBUS_3_RX		0x4006
+#define SLIMBUS_3_TX		0x4007
+#define SLIMBUS_4_RX		0x4008
+#define SLIMBUS_4_TX		0x4009		/* index = 24 */
+
+#define INT_BT_SCO_RX 0x3000		/* index = 25 */
+#define INT_BT_SCO_TX 0x3001		/* index = 26 */
+#define INT_BT_A2DP_RX 0x3002		/* index = 27 */
+#define INT_FM_RX 0x3004		/* index = 28 */
+#define INT_FM_TX 0x3005		/* index = 29 */
+#define RT_PROXY_PORT_001_RX	0x2000    /* index = 30 */
+#define RT_PROXY_PORT_001_TX	0x2001    /* index = 31 */
+#define SECONDARY_PCM_RX 12			/* index = 32 */
+#define SECONDARY_PCM_TX 13			/* index = 33 */
+#define PSEUDOPORT_01           0x8001    /* index =34 */
+
+#define AFE_PORT_INVALID 0xFFFF
+#define SLIMBUS_EXTPROC_RX AFE_PORT_INVALID
+
+#define AFE_PORT_CMD_START 0x000100ca
+
+#define AFE_EVENT_RTPORT_START 0
+#define AFE_EVENT_RTPORT_STOP 1
+#define AFE_EVENT_RTPORT_LOW_WM 2
+#define AFE_EVENT_RTPORT_HI_WM 3
+
+struct afe_port_start_command {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 gain;		/* Q13 */
+	u32 sample_rate;	/* 8 , 16, 48khz */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_STOP 0x000100cb
+struct afe_port_stop_command {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_APPLY_GAIN 0x000100cc
+struct afe_port_gain_command {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16	gain;/* Q13 */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_SIDETONE_CTL 0x000100cd
+struct afe_port_sidetone_command {
+	struct apr_hdr hdr;
+	u16 rx_port_id;		/* Primary i2s tx = 1 */
+				/* PCM tx = 3 */
+				/* Secondary i2s tx = 5 */
+				/* Mi2s tx = 7 */
+				/* Digital mic tx = 11 */
+	u16 tx_port_id;		/* Primary i2s rx = 0 */
+				/* PCM rx = 2 */
+				/* Secondary i2s rx = 4 */
+				/* Mi2S rx = 6 */
+				/* HDMI rx = 8 */
+	u16 gain;		/* Q13 */
+	u16 enable;		/* 1 = enable, 0 = disable */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_LOOPBACK 0x000100ce
+struct afe_loopback_command {
+	struct apr_hdr hdr;
+	u16 tx_port_id;		/* Primary i2s rx = 0 */
+				/* PCM rx = 2 */
+				/* Secondary i2s rx = 4 */
+				/* Mi2S rx = 6 */
+				/* HDMI rx = 8 */
+	u16 rx_port_id;		/* Primary i2s tx = 1 */
+				/* PCM tx = 3 */
+				/* Secondary i2s tx = 5 */
+				/* Mi2s tx = 7 */
+				/* Digital mic tx = 11 */
+	u16 mode;		/* Default -1, DSP will conver
+					the tx to rx format */
+	u16 enable;		/* 1 = enable, 0 = disable */
+} __attribute__ ((packed));
+
+#define AFE_PSEUDOPORT_CMD_START 0x000100cf
+struct afe_pseudoport_start_command {
+	struct apr_hdr hdr;
+	u16 port_id;		/* Pseudo Port 1 = 0x8000 */
+				/* Pseudo Port 2 = 0x8001 */
+				/* Pseudo Port 3 = 0x8002 */
+	u16 timing;		/* FTRT = 0 , AVTimer = 1, */
+} __attribute__ ((packed));
+
+#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
+struct afe_pseudoport_stop_command {
+	struct apr_hdr hdr;
+	u16 port_id;		/* Pseudo Port 1 = 0x8000 */
+				/* Pseudo Port 2 = 0x8001 */
+				/* Pseudo Port 3 = 0x8002 */
+	u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_CMD_GET_ACTIVE_PORTS 0x000100d1
+
+
+#define AFE_CMD_GET_ACTIVE_HANDLES_FOR_PORT 0x000100d2
+struct afe_get_active_handles_command {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 reserved;
+} __attribute__ ((packed));
+
+/*
+ * Opcode for AFE to start DTMF.
+ */
+#define AFE_PORTS_CMD_DTMF_CTL	0x00010102
+
+/** DTMF payload.*/
+struct afe_dtmf_generation_command {
+	struct apr_hdr hdr;
+
+	/*
+	 * Duration of the DTMF tone in ms.
+	 * -1      -> continuous,
+	 *  0      -> disable
+	 */
+	int64_t                   duration_in_ms;
+
+	/*
+	 * The DTMF high tone frequency.
+	 */
+	uint16_t                  high_freq;
+
+	/*
+	 * The DTMF low tone frequency.
+	 */
+	uint16_t                  low_freq;
+
+	/*
+	 * The DTMF volume setting
+	 */
+	uint16_t                  gain;
+
+	/*
+	 * The number of ports to enable/disable on.
+	 */
+	uint16_t                  num_ports;
+
+	/*
+	 * The Destination ports - array  .
+	 * For DTMF on multiple ports, portIds needs to
+	 * be populated numPorts times.
+	 */
+	uint16_t                  port_ids;
+
+	/*
+	 * variable for 32 bit alignment of APR packet.
+	 */
+	uint16_t                  reserved;
+} __packed;
+
+#define AFE_PCM_CFG_MODE_PCM			0x0
+#define AFE_PCM_CFG_MODE_AUX			0x1
+#define AFE_PCM_CFG_SYNC_EXT			0x0
+#define AFE_PCM_CFG_SYNC_INT			0x1
+#define AFE_PCM_CFG_FRM_8BPF			0x0
+#define AFE_PCM_CFG_FRM_16BPF			0x1
+#define AFE_PCM_CFG_FRM_32BPF			0x2
+#define AFE_PCM_CFG_FRM_64BPF			0x3
+#define AFE_PCM_CFG_FRM_128BPF			0x4
+#define AFE_PCM_CFG_FRM_256BPF			0x5
+#define AFE_PCM_CFG_QUANT_ALAW_NOPAD		0x0
+#define AFE_PCM_CFG_QUANT_MULAW_NOPAD		0x1
+#define AFE_PCM_CFG_QUANT_LINEAR_NOPAD		0x2
+#define AFE_PCM_CFG_QUANT_ALAW_PAD		0x3
+#define AFE_PCM_CFG_QUANT_MULAW_PAD		0x4
+#define AFE_PCM_CFG_QUANT_LINEAR_PAD		0x5
+#define AFE_PCM_CFG_CDATAOE_MASTER		0x0
+#define AFE_PCM_CFG_CDATAOE_SHARE		0x1
+
+struct afe_port_pcm_cfg {
+	u16	mode;	/* PCM (short sync) = 0, AUXPCM (long sync) = 1 */
+	u16	sync;	/* external = 0 , internal = 1 */
+	u16	frame;	/* 8 bpf = 0 */
+			/* 16 bpf = 1 */
+			/* 32 bpf = 2 */
+			/* 64 bpf = 3 */
+			/* 128 bpf = 4 */
+			/* 256 bpf = 5 */
+	u16     quant;
+	u16	slot;	/* Slot for PCM stream , 0 - 31 */
+	u16	data;	/* 0, PCM block is the only master */
+			/* 1, PCM block is shares to driver data out signal */
+			/*    other master                                  */
+	u16	reserved;
+} __attribute__ ((packed));
+
+enum {
+	AFE_I2S_SD0 = 1,
+	AFE_I2S_SD1,
+	AFE_I2S_SD2,
+	AFE_I2S_SD3,
+	AFE_I2S_QUAD01,
+	AFE_I2S_QUAD23,
+	AFE_I2S_6CHS,
+	AFE_I2S_8CHS,
+};
+
+#define AFE_MI2S_MONO 0
+#define AFE_MI2S_STEREO 3
+#define AFE_MI2S_4CHANNELS 4
+#define AFE_MI2S_6CHANNELS 6
+#define AFE_MI2S_8CHANNELS 8
+
+struct afe_port_mi2s_cfg {
+	u16	bitwidth;	/* 16,24,32 */
+	u16	line;		/* Called ChannelMode in documentation */
+				/* i2s_sd0 = 1 */
+				/* i2s_sd1 = 2 */
+				/* i2s_sd2 = 3 */
+				/* i2s_sd3 = 4 */
+				/* i2s_quad01 = 5 */
+				/* i2s_quad23 = 6 */
+				/* i2s_6chs = 7 */
+				/* i2s_8chs = 8 */
+	u16	channel;	/* Called MonoStereo in documentation */
+				/* i2s mono = 0 */
+				/* i2s mono right = 1 */
+				/* i2s mono left = 2 */
+				/* i2s stereo = 3 */
+	u16	ws;		/* 0, word select signal from external source */
+				/* 1, word select signal from internal source */
+	u16	format;	/* don't touch this field if it is not for */
+				/* AFE_PORT_CMD_I2S_CONFIG opcode */
+} __attribute__ ((packed));
+
+struct afe_port_hdmi_cfg {
+	u16	bitwidth;	/* 16,24,32 */
+	u16	channel_mode;	/* HDMI Stereo = 0 */
+				/* HDMI_3Point1 (4-ch) = 1 */
+				/* HDMI_5Point1 (6-ch) = 2 */
+				/* HDMI_6Point1 (8-ch) = 3 */
+	u16	data_type;	/* HDMI_Linear = 0 */
+				/* HDMI_non_Linear = 1 */
+} __attribute__ ((packed));
+
+
+struct afe_port_hdmi_multi_ch_cfg {
+	u16	data_type;		/* HDMI_Linear = 0 */
+					/* HDMI_non_Linear = 1 */
+	u16	channel_allocation;	/* The default is 0 (Stereo) */
+	u16	reserved;		/* must be set to 0 */
+} __packed;
+
+
+/* Slimbus Device Ids */
+#define AFE_SLIMBUS_DEVICE_1		0x0
+#define AFE_SLIMBUS_DEVICE_2		0x1
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT	16
+
+struct afe_port_slimbus_cfg {
+	u16	slimbus_dev_id;		/* SLIMBUS Device id.*/
+
+	u16	slave_dev_pgd_la;	/* Slave ported generic device
+					* logical address.
+					*/
+	u16	slave_dev_intfdev_la;	/* Slave interface device logical
+					* address.
+					*/
+	u16	bit_width;		/**  bit width of the samples, 16, 24.*/
+
+	u16	data_format;		/** data format.*/
+
+	u16	num_channels;		/** Number of channels.*/
+
+	/** Slave port mapping for respective channels.*/
+	u16	slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+
+	u16	reserved;
+} __packed;
+
+struct afe_port_slimbus_sch_cfg {
+	u16	slimbus_dev_id;		/* SLIMBUS Device id.*/
+	u16	bit_width;		/**  bit width of the samples, 16, 24.*/
+	u16	data_format;		/** data format.*/
+	u16	num_channels;		/** Number of channels.*/
+	u16	reserved;
+	/** Slave channel  mapping for respective channels.*/
+	u8	slave_ch_mapping[8];
+} __packed;
+
+struct afe_port_rtproxy_cfg {
+	u16	bitwidth;	/* 16,24,32 */
+	u16	interleaved;    /* interleaved = 1 */
+				/* Noninterleaved = 0 */
+	u16	frame_sz;	/* 5ms buffers = 160bytes */
+	u16	jitter;		/* 10ms of jitter = 320 */
+	u16	lw_mark;	/* Low watermark in bytes for triggering event*/
+	u16	hw_mark;	/* High watermark bytes for triggering event*/
+	u16	rsvd;
+	int	num_ch;		/* 1 to 8 */
+} __packed;
+
+struct afe_port_pseudo_cfg {
+	u16 bit_width;
+	u16 num_channels;
+	u16 data_format;
+	u16 timing_mode;
+	u16 reserved;
+} __packed;
+
+#define AFE_PORT_AUDIO_IF_CONFIG 0x000100d3
+#define AFE_PORT_AUDIO_SLIM_SCH_CONFIG 0x000100e4
+#define AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG	0x000100D9
+#define AFE_PORT_CMD_I2S_CONFIG	0x000100E7
+
+union afe_port_config {
+	struct afe_port_pcm_cfg           pcm;
+	struct afe_port_mi2s_cfg          mi2s;
+	struct afe_port_hdmi_cfg          hdmi;
+	struct afe_port_hdmi_multi_ch_cfg hdmi_multi_ch;
+	struct afe_port_slimbus_cfg	  slimbus;
+	struct afe_port_slimbus_sch_cfg	  slim_sch;
+	struct afe_port_rtproxy_cfg       rtproxy;
+	struct afe_port_pseudo_cfg        pseudo;
+} __attribute__((packed));
+
+struct afe_audioif_config_command {
+	struct apr_hdr hdr;
+	u16 port_id;
+	union afe_port_config port;
+} __attribute__ ((packed));
+
+#define AFE_TEST_CODEC_LOOPBACK_CTL 0x000100d5
+struct afe_codec_loopback_command {
+	u16	port_inf;	/* Primary i2s = 0 */
+				/* PCM = 2 */
+				/* Secondary i2s = 4 */
+				/* Mi2s = 6 */
+	u16	enable;		/* 0, disable. 1, enable */
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_SIDETONE_GAIN	0x00010300
+struct afe_param_sidetone_gain {
+	u16 gain;
+	u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_PARAM_ID_SAMPLING_RATE	0x00010301
+struct afe_param_sampling_rate {
+	u32 sampling_rate;
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_CHANNELS		0x00010302
+struct afe_param_channels {
+	u16 channels;
+	u16 reserved;
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_LOOPBACK_GAIN	0x00010303
+struct afe_param_loopback_gain {
+	u16 gain;
+	u16 reserved;
+} __attribute__ ((packed));
+
+/* Parameter ID used to configure and enable/disable the loopback path. The
+ * difference with respect to the existing API, AFE_PORT_CMD_LOOPBACK, is that
+ * it allows Rx port to be configured as source port in loopback path. Port-id
+ * in AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be Tx or Rx port.
+ * In addition, we can configure the type of routing mode to handle different
+ * use cases.
+*/
+enum {
+	/* Regular loopback from source to destination port */
+	LB_MODE_DEFAULT = 1,
+	/* Sidetone feed from Tx source to Rx destination port */
+	LB_MODE_SIDETONE,
+	/* Echo canceller reference, voice + audio + DTMF */
+	LB_MODE_EC_REF_VOICE_AUDIO,
+	/* Echo canceller reference, voice alone */
+	LB_MODE_EC_REF_VOICE
+};
+
+#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
+#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
+struct afe_param_loopback_cfg {
+	/* Minor version used for tracking the version of the configuration
+	 * interface.
+	 */
+	uint32_t loopback_cfg_minor_version;
+
+	/* Destination Port Id. */
+	uint16_t dst_port_id;
+
+	/* Specifies data path type from src to dest port. Supported values:
+	 * LB_MODE_DEFAULT
+	 * LB_MODE_SIDETONE
+	 * LB_MODE_EC_REF_VOICE_AUDIO
+	 * LB_MODE_EC_REF_VOICE
+	 */
+	uint16_t routing_mode;
+
+	/* Specifies whether to enable (1) or disable (0) an AFE loopback. */
+	uint16_t enable;
+
+	/* Reserved for 32-bit alignment. This field must be set to 0. */
+	uint16_t reserved;
+} __packed;
+
+#define AFE_MODULE_ID_PORT_INFO		0x00010200
+/* Module ID for the loopback-related parameters. */
+#define AFE_MODULE_LOOPBACK           0x00010205
+struct afe_param_payload_base {
+	u32 module_id;
+	u32 param_id;
+	u16 param_size;
+	u16 reserved;
+} __packed;
+
+struct afe_param_payload {
+	struct afe_param_payload_base base;
+	union {
+		struct afe_param_sidetone_gain sidetone_gain;
+		struct afe_param_sampling_rate sampling_rate;
+		struct afe_param_channels      channels;
+		struct afe_param_loopback_gain loopback_gain;
+		struct afe_param_loopback_cfg loopback_cfg;
+	} __attribute__((packed)) param;
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_SET_PARAM		0x000100dc
+
+struct afe_port_cmd_set_param {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 payload_size;
+	u32 payload_address;
+	struct afe_param_payload payload;
+} __attribute__ ((packed));
+
+struct afe_port_cmd_set_param_no_payload {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 payload_size;
+	u32 payload_address;
+} __packed;
+
+#define AFE_EVENT_GET_ACTIVE_PORTS 0x00010100
+struct afe_get_active_ports_rsp {
+	u16	num_ports;
+	u16	port_id;
+} __attribute__ ((packed));
+
+
+#define AFE_EVENT_GET_ACTIVE_HANDLES 0x00010102
+struct afe_get_active_handles_rsp {
+	u16	port_id;
+	u16	num_handles;
+	u16	mode;		/* 0, voice rx */
+				/* 1, voice tx */
+				/* 2, audio rx */
+				/* 3, audio tx */
+	u16	handle;
+} __attribute__ ((packed));
+
+#define AFE_SERVICE_CMD_MEMORY_MAP 0x000100DE
+struct afe_cmd_memory_map {
+	struct apr_hdr hdr;
+	u32 phy_addr;
+	u32 mem_sz;
+	u16 mem_id;
+	u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_MEMORY_UNMAP 0x000100DF
+struct afe_cmd_memory_unmap {
+	struct apr_hdr hdr;
+	u32 phy_addr;
+} __packed;
+
+#define AFE_SERVICE_CMD_REG_RTPORT 0x000100E0
+struct afe_cmd_reg_rtport {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_UNREG_RTPORT 0x000100E1
+struct afe_cmd_unreg_rtport {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_RTPORT_WR 0x000100E2
+struct afe_cmd_rtport_wr {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 rsvd;
+	u32 buf_addr;
+	u32 bytes_avail;
+} __packed;
+
+#define AFE_SERVICE_CMD_RTPORT_RD 0x000100E3
+struct afe_cmd_rtport_rd {
+	struct apr_hdr hdr;
+	u16 port_id;
+	u16 rsvd;
+	u32 buf_addr;
+	u32 bytes_avail;
+} __packed;
+
+#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105
+
+#define ADM_MAX_COPPS 5
+
+#define ADM_SERVICE_CMD_GET_COPP_HANDLES                 0x00010300
+struct adm_get_copp_handles_command {
+	struct apr_hdr hdr;
+} __attribute__ ((packed));
+
+#define ADM_CMD_MATRIX_MAP_ROUTINGS                      0x00010301
+struct adm_routings_session {
+	u16 id;
+	u16 num_copps;
+	u16 copp_id[ADM_MAX_COPPS+1]; /*Padding if numCopps is odd */
+} __packed;
+
+struct adm_routings_command {
+	struct apr_hdr hdr;
+	u32 path; /* 0 = Rx, 1 Tx */
+	u32 num_sessions;
+	struct adm_routings_session session[8];
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_MATRIX_RAMP_GAINS                        0x00010302
+struct adm_ramp_gain {
+	struct apr_hdr hdr;
+	u16 session_id;
+	u16 copp_id;
+	u16 initial_gain;
+	u16 gain_increment;
+	u16 ramp_duration;
+	u16 reserved;
+} __attribute__ ((packed));
+
+struct adm_ramp_gains_command {
+	struct apr_hdr hdr;
+	u32 id;
+	u32 num_gains;
+	struct adm_ramp_gain gains[ADM_MAX_COPPS];
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_COPP_OPEN                                0x00010304
+struct adm_copp_open_command {
+	struct apr_hdr hdr;
+	u16 flags;
+	u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */
+	u16 endpoint_id1;
+	u16 endpoint_id2;
+	u32 topology_id;
+	u16 channel_config;
+	u16 reserved;
+	u32 rate;
+} __attribute__ ((packed));
+
+#define ADM_CMD_COPP_CLOSE                               0x00010305
+
+#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN                  0x00010310
+#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN_V3               0x00010333
+struct adm_multi_ch_copp_open_command {
+	struct apr_hdr hdr;
+	u16 flags;
+	u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */
+	u16 endpoint_id1;
+	u16 endpoint_id2;
+	u32 topology_id;
+	u16 channel_config;
+	u16 reserved;
+	u32 rate;
+	u8 dev_channel_mapping[8];
+} __packed;
+
+struct adm_multi_channel_copp_open_v3 {
+	struct apr_hdr hdr;
+	u16 flags;
+	u16 mode;
+	u16 endpoint_id1;
+	u16 endpoint_id2;
+	u32 topology_id;
+	u16 channel_config;
+	u16 bit_width;
+	u32 rate;
+	u8  dev_channel_mapping[8];
+};
+#define ADM_CMD_MEMORY_MAP				0x00010C30
+struct adm_cmd_memory_map{
+	struct apr_hdr	hdr;
+	u32		buf_add;
+	u32		buf_size;
+	u16		mempool_id;
+	u16		reserved;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_UNMAP				0x00010C31
+struct adm_cmd_memory_unmap{
+	struct apr_hdr	hdr;
+	u32		buf_add;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_MAP_REGIONS			0x00010C47
+struct adm_memory_map_regions{
+	u32		phys;
+	u32		buf_size;
+} __attribute__((packed));
+
+struct adm_cmd_memory_map_regions{
+	struct apr_hdr	hdr;
+	u16		mempool_id;
+	u16		nregions;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_UNMAP_REGIONS			0x00010C48
+struct adm_memory_unmap_regions{
+	u32		phys;
+} __attribute__((packed));
+
+struct adm_cmd_memory_unmap_regions{
+	struct apr_hdr	hdr;
+	u16		nregions;
+	u16		reserved;
+} __attribute__((packed));
+
+#define DEFAULT_COPP_TOPOLOGY				0x00010be3
+#define DEFAULT_POPP_TOPOLOGY				0x00010be4
+#define VPM_TX_SM_ECNS_COPP_TOPOLOGY			0x00010F71
+#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY			0x00010F72
+#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY		0x00010F75
+
+#define LOWLATENCY_POPP_TOPOLOGY			0x00010C68
+#define LOWLATENCY_COPP_TOPOLOGY			0x00010312
+#define PCM_BITS_PER_SAMPLE				16
+
+#define ASM_OPEN_WRITE_PERF_MODE_BIT			(1<<28)
+#define ASM_OPEN_READ_PERF_MODE_BIT			(1<<29)
+#define ADM_MULTI_CH_COPP_OPEN_PERF_MODE_BIT		(1<<13)
+
+
+#define ASM_MAX_EQ_BANDS 12
+
+struct asm_eq_band {
+	u32 band_idx; /* The band index, 0 .. 11 */
+	u32 filter_type; /* Filter band type */
+	u32 center_freq_hz; /* Filter band center frequency */
+	u32 filter_gain; /* Filter band initial gain (dB) */
+			/* Range is +12 dB to -12 dB with 1dB increments. */
+	u32 q_factor;
+} __attribute__ ((packed));
+
+struct asm_equalizer_params {
+	u32 enable;
+	u32 num_bands;
+	struct asm_eq_band eq_bands[ASM_MAX_EQ_BANDS];
+} __attribute__ ((packed));
+
+struct asm_master_gain_params {
+	u16 master_gain;
+	u16 padding;
+} __attribute__ ((packed));
+
+struct asm_lrchannel_gain_params {
+	u16 left_gain;
+	u16 right_gain;
+} __attribute__ ((packed));
+
+struct asm_mute_params {
+	u32 muteflag;
+} __attribute__ ((packed));
+
+struct asm_softvolume_params {
+	u32 period;
+	u32 step;
+	u32 rampingcurve;
+} __attribute__ ((packed));
+
+struct asm_softpause_params {
+	u32 enable;
+	u32 period;
+	u32 step;
+	u32 rampingcurve;
+} __packed;
+
+struct asm_pp_param_data_hdr {
+	u32 module_id;
+	u32 param_id;
+	u16 param_size;
+	u16 reserved;
+} __attribute__ ((packed));
+
+struct asm_pp_params_command {
+	struct apr_hdr	hdr;
+	u32    *payload;
+	u32	payload_size;
+	struct  asm_pp_param_data_hdr params;
+} __attribute__ ((packed));
+
+#define EQUALIZER_MODULE_ID		0x00010c27
+#define EQUALIZER_PARAM_ID		0x00010c28
+
+#define VOLUME_CONTROL_MODULE_ID	0x00010bfe
+#define MASTER_GAIN_PARAM_ID		0x00010bff
+#define L_R_CHANNEL_GAIN_PARAM_ID	0x00010c00
+#define MUTE_CONFIG_PARAM_ID 0x00010c01
+#define SOFT_PAUSE_PARAM_ID 0x00010D6A
+#define SOFT_VOLUME_PARAM_ID 0x00010C29
+
+#define IIR_FILTER_ENABLE_PARAM_ID 0x00010c03
+#define IIR_FILTER_PREGAIN_PARAM_ID 0x00010c04
+#define IIR_FILTER_CONFIG_PARAM_ID 0x00010c05
+
+#define MBADRC_MODULE_ID 0x00010c06
+#define MBADRC_ENABLE_PARAM_ID 0x00010c07
+#define MBADRC_CONFIG_PARAM_ID 0x00010c08
+
+
+#define ADM_CMD_SET_PARAMS                               0x00010306
+#define ADM_CMD_GET_PARAMS                               0x0001030B
+#define ADM_CMDRSP_GET_PARAMS                            0x0001030C
+struct adm_set_params_command {
+	struct apr_hdr		hdr;
+	u32			payload;
+	u32			payload_size;
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_TAP_COPP_PCM                             0x00010307
+struct adm_tap_copp_pcm_command {
+	struct apr_hdr hdr;
+} __attribute__ ((packed));
+
+
+/* QDSP6 to Client messages
+*/
+#define ADM_SERVICE_CMDRSP_GET_COPP_HANDLES              0x00010308
+struct adm_get_copp_handles_respond {
+	struct apr_hdr hdr;
+	u32 handles;
+	u32 copp_id;
+} __attribute__ ((packed));
+
+#define ADM_CMDRSP_COPP_OPEN                             0x0001030A
+struct adm_copp_open_respond {
+	u32 status;
+	u16 copp_id;
+	u16 reserved;
+} __attribute__ ((packed));
+
+#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN               0x00010311
+#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN_V3            0x00010334
+
+
+#define ASM_STREAM_PRIORITY_NORMAL	0
+#define ASM_STREAM_PRIORITY_LOW		1
+#define ASM_STREAM_PRIORITY_HIGH	2
+#define ASM_STREAM_PRIORITY_RESERVED	3
+
+#define ASM_END_POINT_DEVICE_MATRIX	0
+#define ASM_END_POINT_STREAM		1
+
+#define AAC_ENC_MODE_AAC_LC            0x02
+#define AAC_ENC_MODE_AAC_P             0x05
+#define AAC_ENC_MODE_EAAC_P            0x1D
+
+#define ASM_STREAM_CMD_CLOSE                             0x00010BCD
+#define ASM_STREAM_CMD_FLUSH                             0x00010BCE
+#define ASM_STREAM_CMD_SET_PP_PARAMS                     0x00010BCF
+#define ASM_STREAM_CMD_GET_PP_PARAMS                     0x00010BD0
+#define ASM_STREAM_CMDRSP_GET_PP_PARAMS                  0x00010BD1
+#define ASM_SESSION_CMD_PAUSE                            0x00010BD3
+#define ASM_SESSION_CMD_GET_SESSION_TIME                 0x00010BD4
+#define ASM_DATA_CMD_EOS                                 0x00010BDB
+#define ASM_DATA_EVENT_EOS                               0x00010BDD
+
+#define ASM_SERVICE_CMD_GET_STREAM_HANDLES               0x00010C0B
+#define ASM_STREAM_CMD_FLUSH_READBUFS                    0x00010C09
+
+#define ASM_SESSION_EVENT_RX_UNDERFLOW			 0x00010C17
+#define ASM_SESSION_EVENT_TX_OVERFLOW			 0x00010C18
+#define ASM_SERVICE_CMD_GET_WALLCLOCK_TIME               0x00010C19
+#define ASM_DATA_CMDRSP_EOS                              0x00010C1C
+
+/* ASM Data structures */
+
+/* common declarations */
+struct asm_pcm_cfg {
+	u16 ch_cfg;
+	u16 bits_per_sample;
+	u32 sample_rate;
+	u16 is_signed;
+	u16 interleaved;
+};
+
+#define PCM_CHANNEL_NULL 0
+
+/* Front left channel. */
+#define PCM_CHANNEL_FL    1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR    2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC    3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS   4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS   5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE  6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS   7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB   8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB   9
+
+/* Top surround channel. */
+#define PCM_CHANNEL_TS   10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH  11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS   12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC  13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC  14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC  15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC  16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL  8
+
+/* Maximum number of channels supported
+ * in ASM_ENCDEC_DEC_CHAN_MAP command
+ */
+#define MAX_CHAN_MAP_CHANNELS 16
+/*
+ *  Multiple-channel PCM decoder format block structure used in the
+ *  #ASM_STREAM_CMD_OPEN_WRITE command.
+ *  The data must be in little-endian format.
+ */
+struct asm_multi_channel_pcm_fmt_blk {
+
+	u16 num_channels;	/*
+				 * Number of channels.
+				 * Supported values:1 to 8
+				 */
+
+	u16 bits_per_sample;	/*
+				 * Number of bits per sample per channel.
+				 * Supported values: 16, 24 When used for
+				 * playback, the client must send 24-bit
+				 * samples packed in 32-bit words. The
+				 * 24-bit samples must be placed in the most
+				 * significant 24 bits of the 32-bit word. When
+				 * used for recording, the aDSP sends 24-bit
+				 * samples packed in 32-bit words. The 24-bit
+				 * samples are placed in the most significant
+				 * 24 bits of the 32-bit word.
+				 */
+
+	u32 sample_rate;	/*
+				 * Number of samples per second
+				 * (in Hertz). Supported values:
+				 * 2000 to 48000
+				 */
+
+	u16 is_signed;		/*
+				 * Flag that indicates the samples
+				 * are signed (1).
+				 */
+
+	u16 is_interleaved;	/*
+				 * Flag that indicates whether the channels are
+				 * de-interleaved (0) or interleaved (1).
+				 * Interleaved format means corresponding
+				 * samples from the left and right channels are
+				 * interleaved within the buffer.
+				 * De-interleaved format means samples from
+				 * each channel are contiguous in the buffer.
+				 * The samples from one channel immediately
+				 * follow those of the previous channel.
+				 */
+
+	u8 channel_mapping[8];	/*
+				 * Supported values:
+				 * PCM_CHANNEL_NULL, PCM_CHANNEL_FL,
+				 * PCM_CHANNEL_FR, PCM_CHANNEL_FC,
+				 * PCM_CHANNEL_LS, PCM_CHANNEL_RS,
+				 * PCM_CHANNEL_LFE, PCM_CHANNEL_CS,
+				 * PCM_CHANNEL_LB, PCM_CHANNEL_RB,
+				 * PCM_CHANNEL_TS, PCM_CHANNEL_CVH,
+				 * PCM_CHANNEL_MS, PCM_CHANNEL_FLC,
+				 * PCM_CHANNEL_FRC, PCM_CHANNEL_RLC,
+				 * PCM_CHANNEL_RRC.
+				 * Channel[i] mapping describes channel I. Each
+				 * element i of the array describes channel I
+				 * inside the buffer where  I < num_channels.
+				 * An unused channel is set to zero.
+				 */
+};
+struct asm_dts_enc_cfg {
+	uint32_t	sample_rate;
+	/*
+	* Samples at which input is to be encoded.
+	* Supported values:
+	* 44100 -- encode at 44.1 Khz
+	* 48000 -- encode at 48 Khz
+	*/
+
+	uint32_t	num_channels;
+	/*
+	* Number of channels for multi-channel encoding.
+	* Supported values: 1 to 6
+	*/
+
+	uint8_t		channel_mapping[6];
+	/*
+	* Channel array of size 16. Channel[i] mapping describes channel I.
+	* Each element i of the array describes channel I inside the buffer
+	* where num_channels. An unused channel is set to zero. Only first
+	* num_channels elements are valid
+
+	* Supported values:
+	* - # PCM_CHANNEL_L
+	* - # PCM_CHANNEL_R
+	* - # PCM_CHANNEL_C
+	* - # PCM_CHANNEL_LS
+	* - # PCM_CHANNEL_RS
+	* - # PCM_CHANNEL_LFE
+	*/
+
+};
+struct asm_adpcm_cfg {
+	u16 ch_cfg;
+	u16 bits_per_sample;
+	u32 sample_rate;
+	u32 block_size;
+};
+
+struct asm_yadpcm_cfg {
+	u16 ch_cfg;
+	u16 bits_per_sample;
+	u32 sample_rate;
+};
+
+struct asm_midi_cfg {
+	u32 nMode;
+};
+
+struct asm_wma_cfg {
+	u16 format_tag;
+	u16 ch_cfg;
+	u32 sample_rate;
+	u32 avg_bytes_per_sec;
+	u16 block_align;
+	u16 valid_bits_per_sample;
+	u32 ch_mask;
+	u16 encode_opt;
+	u16 adv_encode_opt;
+	u32 adv_encode_opt2;
+	u32 drc_peak_ref;
+	u32 drc_peak_target;
+	u32 drc_ave_ref;
+	u32 drc_ave_target;
+};
+
+struct asm_wmapro_cfg {
+	u16 format_tag;
+	u16 ch_cfg;
+	u32 sample_rate;
+	u32 avg_bytes_per_sec;
+	u16 block_align;
+	u16 valid_bits_per_sample;
+	u32 ch_mask;
+	u16 encode_opt;
+	u16 adv_encode_opt;
+	u32 adv_encode_opt2;
+	u32 drc_peak_ref;
+	u32 drc_peak_target;
+	u32 drc_ave_ref;
+	u32 drc_ave_target;
+};
+
+struct asm_aac_cfg {
+	u16 format;
+	u16 aot;
+	u16 ep_config;
+	u16 section_data_resilience;
+	u16 scalefactor_data_resilience;
+	u16 spectral_data_resilience;
+	u16 ch_cfg;
+	u16 reserved;
+	u32 sample_rate;
+};
+
+struct asm_amrwbplus_cfg {
+	u32  size_bytes;
+	u32  version;
+	u32  num_channels;
+	u32  amr_band_mode;
+	u32  amr_dtx_mode;
+	u32  amr_frame_fmt;
+	u32  amr_lsf_idx;
+};
+
+struct asm_flac_cfg {
+	u16 stream_info_present;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 ch_cfg;
+	u16 sample_size;
+	u16 sample_rate;
+	u16 md5_sum;
+	u32 ext_sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+};
+
+struct asm_vorbis_cfg {
+	u32 ch_cfg;
+	u32 bit_rate;
+	u32 min_bit_rate;
+	u32 max_bit_rate;
+	u16 bit_depth_pcm_sample;
+	u16 bit_stream_format;
+};
+
+struct asm_aac_read_cfg {
+	u32 bitrate;
+	u32 enc_mode;
+	u16 format;
+	u16 ch_cfg;
+	u32 sample_rate;
+};
+
+struct asm_amrnb_read_cfg {
+	u16 mode;
+	u16 dtx_mode;
+};
+
+struct asm_amrwb_read_cfg {
+	u16 mode;
+	u16 dtx_mode;
+};
+
+struct asm_evrc_read_cfg {
+	u16 max_rate;
+	u16 min_rate;
+	u16 rate_modulation_cmd;
+	u16 reserved;
+};
+
+struct asm_qcelp13_read_cfg {
+	u16 max_rate;
+	u16 min_rate;
+	u16 reduced_rate_level;
+	u16 rate_modulation_cmd;
+};
+
+struct asm_sbc_read_cfg {
+	u32 subband;
+	u32 block_len;
+	u32 ch_mode;
+	u32 alloc_method;
+	u32 bit_rate;
+	u32 sample_rate;
+};
+
+struct asm_sbc_bitrate {
+	u32 bitrate;
+};
+
+struct asm_immed_decode {
+	u32 mode;
+};
+
+struct asm_sbr_ps {
+	u32 enable;
+};
+
+struct asm_dual_mono {
+	u16 sce_left;
+	u16 sce_right;
+};
+
+struct asm_dec_chan_map {
+	u32 num_channels;			  /* Number of decoder output
+						   * channels. A value of 0
+						   * indicates native channel
+						   * mapping, which is valid
+						   * only for NT mode. This
+						   * means the output of the
+						   * decoder is to be preserved
+						   * as is.
+						   */
+
+	u8 channel_mapping[MAX_CHAN_MAP_CHANNELS];/* Channel array of size
+						   * num_channels. It can grow
+						   * till MAX_CHAN_MAP_CHANNELS.
+						   * Channel[i] mapping
+						   * describes channel I inside
+						   * the decoder output buffer.
+						   * Valid channel mapping
+						   * values are to be present at
+						   * the beginning of the array.
+						   * All remaining elements of
+						   * the array are to be filled
+						   * with PCM_CHANNEL_NULL.
+						   */
+};
+
+struct asm_encode_cfg_blk {
+	u32 frames_per_buf;
+	u32 format_id;
+	u32 cfg_size;
+	union {
+		struct asm_pcm_cfg          pcm;
+		struct asm_aac_read_cfg     aac;
+		struct asm_amrnb_read_cfg   amrnb;
+		struct asm_evrc_read_cfg    evrc;
+		struct asm_qcelp13_read_cfg qcelp13;
+		struct asm_sbc_read_cfg     sbc;
+		struct asm_amrwb_read_cfg   amrwb;
+		struct asm_multi_channel_pcm_fmt_blk      mpcm;
+		struct asm_dts_enc_cfg      dts;
+	} __attribute__((packed)) cfg;
+};
+
+struct asm_frame_meta_info {
+	u32 offset_to_frame;
+	u32 frame_size;
+	u32 encoded_pcm_samples;
+	u32 msw_ts;
+	u32 lsw_ts;
+	u32 nflags;
+};
+
+/* Stream level commands */
+#define ASM_STREAM_CMD_OPEN_READ                         0x00010BCB
+#define ASM_STREAM_CMD_OPEN_READ_V2_1                    0x00010DB2
+struct asm_stream_cmd_open_read {
+	struct apr_hdr hdr;
+	u32            uMode;
+	u32            src_endpoint;
+	u32            pre_proc_top;
+	u32            format;
+} __attribute__((packed));
+
+struct asm_stream_cmd_open_read_v2_1 {
+	struct apr_hdr hdr;
+	u32            uMode;
+	u32            src_endpoint;
+	u32            pre_proc_top;
+	u32            format;
+	u16            bits_per_sample;
+	u16            reserved;
+} __packed;
+
+/* Supported formats */
+#define LINEAR_PCM   0x00010BE5
+#define DTMF         0x00010BE6
+#define ADPCM        0x00010BE7
+#define YADPCM       0x00010BE8
+#define MP3          0x00010BE9
+#define MPEG4_AAC    0x00010BEA
+#define AMRNB_FS     0x00010BEB
+#define AMRWB_FS     0x00010BEC
+#define V13K_FS      0x00010BED
+#define EVRC_FS      0x00010BEE
+#define EVRCB_FS     0x00010BEF
+#define EVRCWB_FS    0x00010BF0
+#define MIDI         0x00010BF1
+#define SBC          0x00010BF2
+#define WMA_V10PRO   0x00010BF3
+#define WMA_V9       0x00010BF4
+#define AMR_WB_PLUS  0x00010BF5
+#define AC3_DECODER  0x00010BF6
+#define EAC3_DECODER 0x00010C3C
+#define DTS	0x00010D88
+#define DTS_LBR	0x00010DBB
+#define MP2          0x00010DBE
+#define ATRAC	0x00010D89
+#define MAT	0x00010D8A
+#define G711_ALAW_FS 0x00010BF7
+#define G711_MLAW_FS 0x00010BF8
+#define G711_PCM_FS  0x00010BF9
+#define MPEG4_MULTI_AAC 0x00010D86
+#define US_POINT_EPOS_FORMAT 0x00012310
+#define US_RAW_FORMAT        0x0001127C
+#define US_PROX_FORMAT       0x0001272B
+#define MULTI_CHANNEL_PCM    0x00010C66
+
+#define ASM_ENCDEC_SBCRATE         0x00010C13
+#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14
+#define ASM_ENCDEC_CFG_BLK         0x00010C2C
+
+#define ASM_ENCDEC_SBCRATE         0x00010C13
+#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14
+#define ASM_ENCDEC_CFG_BLK         0x00010C2C
+
+#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED               0x00010D95
+struct asm_stream_cmd_open_read_compressed {
+	struct apr_hdr hdr;
+	u32            uMode;
+	u32            frame_per_buf;
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_WRITE                        0x00010BCA
+#define ASM_STREAM_CMD_OPEN_WRITE_V2_1                   0x00010DB1
+struct asm_stream_cmd_open_write {
+	struct apr_hdr hdr;
+	u32            uMode;
+	u16            sink_endpoint;
+	u16            stream_handle;
+	u32            post_proc_top;
+	u32            format;
+} __attribute__((packed));
+
+#define IEC_61937_MASK	0x00000001
+#define IEC_60958_MASK	0x00000002
+
+#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED	0x00010D84
+struct asm_stream_cmd_open_write_compressed {
+	struct apr_hdr hdr;
+	u32	flags;
+	u32	format;
+} __packed;
+#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK     0x00010DBA
+struct asm_stream_cmd_open_transcode_loopback {
+	struct apr_hdr hdr;
+	uint32_t	mode_flags;
+	/*
+	* All bits are reserved. Clients must set them to zero.
+	*/
+
+	uint32_t	src_format_id;
+	/*
+	* Specifies the media format of the input audio stream.
+
+	* Supported values:
+	* - #ASM_MEDIA_FMT_LINEAR_PCM
+	* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM
+	*/
+
+	uint32_t	sink_format_id;
+	/*
+	* Specifies the media format of the output stream.
+
+	* Supported values:
+	* - #ASM_MEDIA_FMT_LINEAR_PCM
+	* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM
+	* - #ASM_MEDIA_FMT_DTS
+	*/
+
+	uint32_t	audproc_topo_id;
+	/*
+	* Postprocessing topology ID, which specifies the topology (order of
+	* processing) of postprocessing algorithms.
+
+	* Supported values:
+	* - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT
+	* - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER
+	* - #ASM_STREAM_POSTPROC_TOPO_ID_NONE
+	* - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL
+	*/
+
+	uint16_t	src_endpoint_type;
+	/*
+	* Specifies the source endpoint that provides the input samples.
+
+	* Supported values:
+	* - 0 -- Tx device matrix or stream router
+	* (gateway to the hardware ports)
+	* - All other values are reserved
+
+	* Clients must set this field to zero. Otherwise, an error is returned.
+	*/
+
+	uint16_t	sink_endpoint_type;
+	/*
+	* Specifies the sink endpoint type.
+
+	* Supported values:
+	* - 0 -- Rx device matrix or stream router
+	* (gateway to the hardware ports)
+	* - All other values are reserved
+
+	* Clients must set this field to zero. Otherwise, an error is returned.
+	*/
+
+	uint16_t	bits_per_sample;
+	/*
+	* Number of bits per sample processed by the ASM modules.
+	* Supported values: 16, 24
+	*/
+
+	uint16_t	reserved;
+	/*
+	* This field must be set to zero.
+	*/
+} __packed;
+
+/*
+* ID of the DTS mix LFE channel to front channels parameter in the
+* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+* asm_dts_generic_param_t
+* ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT
+*/
+#define ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT                          0x00010DB6
+
+/*
+* ID of the DTS DRC ratio parameter in the
+* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+* asm_dts_generic_param_t
+* ASM_PARAM_ID_DTS_DRC_RATIO
+*/
+#define ASM_PARAM_ID_DTS_DRC_RATIO                                   0x00010DB7
+
+/*
+* ID of the DTS enable dialog normalization parameter in the
+* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+
+* asm_dts_generic_param_t
+* ASM_PARAM_ID_DTS_ENABLE_DIALNORM
+*/
+#define ASM_PARAM_ID_DTS_ENABLE_DIALNORM                             0x00010DB8
+
+/*
+* ID of the DTS enable parse REV2AUX parameter in the
+* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+* asm_dts_generic_param_t
+* ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX
+*/
+#define ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX                         0x00010DB9
+
+struct asm_dts_generic_param {
+	int32_t		generic_parameter;
+	/*
+	* #ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT:
+	* - if enabled, mixes LFE channel to front
+	* while downmixing (if necessary)
+	* - Supported values: 1-> enable, 0-> disable
+	* - Default: disabled
+
+	* #ASM_PARAM_ID_DTS_DRC_RATIO:
+	* - percentage of DRC ratio.
+	* - Supported values: 0-100
+	* - Default: 0, DRC is disabled.
+
+	* #ASM_PARAM_ID_DTS_ENABLE_DIALNORM:
+	* - flag to enable dialog normalization post processing.
+	* - Supported values: 1-> enable, 0-> disable.
+	* - Default: enabled.
+
+	* #ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX:
+	* - flag to enable parsing of rev2aux chunk in the bitstream.
+	* This chunk contains broadcast metadata.
+	* - Supported values: 1-> enable, 0-> disable.
+	* - Default: disabled.
+	*/
+};
+
+struct asm_stream_cmd_dts_dec_param {
+	struct apr_hdr hdr;
+	u32            param_id;
+	u32            param_size;
+	struct asm_dts_generic_param generic_param;
+} __packed;
+
+
+#define ASM_STREAM_CMD_OPEN_READWRITE                    0x00010BCC
+
+struct asm_stream_cmd_open_read_write {
+	struct apr_hdr     hdr;
+	u32                uMode;
+	u32                post_proc_top;
+	u32                write_format;
+	u32                read_format;
+} __attribute__((packed));
+
+#define ASM_STREAM_CMD_OPEN_LOOPBACK	0x00010D6E
+struct asm_stream_cmd_open_loopback {
+	struct apr_hdr         hdr;
+	u32                    mode_flags;
+/* Mode flags.
+ * Bit 0-31: reserved; client should set these bits to 0
+ */
+	u16                    src_endpointype;
+	/* Endpoint type. 0 = Tx Matrix */
+	u16                    sink_endpointype;
+	/* Endpoint type. 0 = Rx Matrix */
+	u32                    postprocopo_id;
+/* Postprocessor topology ID. Specifies the topology of
+ * postprocessing algorithms.
+ */
+} __packed;
+
+#define ADM_CMD_CONNECT_AFE_PORT 0x00010320
+#define ADM_CMD_DISCONNECT_AFE_PORT 0x00010321
+
+struct adm_cmd_connect_afe_port {
+	struct apr_hdr     hdr;
+	u8	mode; /*mode represent the interface is for RX or TX*/
+	u8	session_id; /*ASM session ID*/
+	u16	afe_port_id;
+} __packed;
+
+#define ADM_CMD_CONNECT_AFE_PORT_V2 0x00010332
+
+struct adm_cmd_connect_afe_port_v2 {
+	struct apr_hdr     hdr;
+	u8	mode; /*mode represent the interface is for RX or TX*/
+	u8	session_id; /*ASM session ID*/
+	u16	afe_port_id;
+	u32	num_channels;
+	u32	sampling_rate;
+} __packed;
+
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM                  0x00010C10
+#define ASM_STREAM_CMD_GET_ENCDEC_PARAM                  0x00010C11
+#define ASM_ENCDEC_CFG_BLK_ID				 0x00010C2C
+#define ASM_ENABLE_SBR_PS				 0x00010C63
+#define ASM_CONFIGURE_DUAL_MONO			 0x00010C64
+struct asm_stream_cmd_encdec_cfg_blk{
+	struct apr_hdr              hdr;
+	u32                         param_id;
+	u32                         param_size;
+	struct asm_encode_cfg_blk   enc_blk;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_sbc_bitrate{
+	struct apr_hdr hdr;
+	u32            param_id;
+		struct asm_sbc_bitrate      sbc_bitrate;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_immed_decode{
+	struct apr_hdr hdr;
+	u32            param_id;
+	u32            param_size;
+	struct asm_immed_decode dec;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_sbr{
+	struct apr_hdr hdr;
+	u32            param_id;
+	u32            param_size;
+	struct asm_sbr_ps sbr_ps;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_dualmono {
+	struct apr_hdr hdr;
+	u32            param_id;
+	u32            param_size;
+	struct asm_dual_mono channel_map;
+} __packed;
+
+#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG        0x00010DD8
+
+/* Structure for AAC decoder stereo coefficient setting. */
+
+struct asm_aac_stereo_mix_coeff_selection_param {
+	struct apr_hdr				hdr;
+	u32					param_id;
+	u32					param_size;
+	u32					aac_stereo_mix_coeff_flag;
+} __packed;
+
+#define ASM_ENCDEC_DEC_CHAN_MAP				 0x00010D82
+struct asm_stream_cmd_encdec_channelmap {
+	struct apr_hdr hdr;
+	u32            param_id;
+	u32            param_size;
+	struct asm_dec_chan_map chan_map;
+} __packed;
+
+#define ASM_STREAM _CMD_ADJUST_SAMPLES                   0x00010C0A
+struct asm_stream_cmd_adjust_samples{
+	struct apr_hdr hdr;
+	u16            nsamples;
+	u16            reserved;
+} __attribute__((packed));
+
+#define ASM_STREAM_CMD_TAP_POPP_PCM                      0x00010BF9
+struct asm_stream_cmd_tap_popp_pcm{
+	struct apr_hdr hdr;
+	u16            enable;
+	u16            reserved;
+	u32            module_id;
+} __attribute__((packed));
+
+/*  Session Level commands */
+#define ASM_SESSION_CMD_MEMORY_MAP			0x00010C32
+struct asm_stream_cmd_memory_map{
+	struct apr_hdr	hdr;
+	u32		buf_add;
+	u32		buf_size;
+	u16		mempool_id;
+	u16		reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_UNMAP			0x00010C33
+struct asm_stream_cmd_memory_unmap{
+	struct apr_hdr	hdr;
+	u32		buf_add;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_MAP_REGIONS		0x00010C45
+struct asm_memory_map_regions{
+	u32		phys;
+	u32		buf_size;
+} __attribute__((packed));
+
+struct asm_stream_cmd_memory_map_regions{
+	struct apr_hdr	hdr;
+	u16		mempool_id;
+	u16		nregions;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_UNMAP_REGIONS		0x00010C46
+struct asm_memory_unmap_regions{
+	u32		phys;
+} __attribute__((packed));
+
+struct asm_stream_cmd_memory_unmap_regions{
+	struct apr_hdr	hdr;
+	u16		nregions;
+	u16		reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_RUN                              0x00010BD2
+struct asm_stream_cmd_run{
+	struct apr_hdr hdr;
+	u32            flags;
+	u32            msw_ts;
+	u32            lsw_ts;
+} __attribute__((packed));
+
+/* Session level events */
+#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
+struct asm_stream_cmd_reg_rx_underflow_event{
+	struct apr_hdr hdr;
+	u16            enable;
+	u16            reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_REGISTER_FOR_TX_OVERFLOW_EVENTS  0x00010BD6
+struct asm_stream_cmd_reg_tx_overflow_event{
+	struct apr_hdr hdr;
+	u16            enable;
+	u16            reserved;
+} __attribute__((packed));
+
+/* Data Path commands */
+#define ASM_DATA_CMD_WRITE                               0x00010BD9
+struct asm_stream_cmd_write{
+	struct apr_hdr     hdr;
+	u32	buf_add;
+	u32	avail_bytes;
+	u32	uid;
+	u32	msw_ts;
+	u32	lsw_ts;
+	u32	uflags;
+} __attribute__((packed));
+
+#define ASM_DATA_CMD_READ                                0x00010BDA
+struct asm_stream_cmd_read{
+	struct apr_hdr     hdr;
+	u32	buf_add;
+	u32	buf_size;
+	u32	uid;
+} __attribute__((packed));
+
+#define ASM_DATA_CMD_READ_COMPRESSED                     0x00010DBF
+struct asm_stream_cmd_read_compressed {
+	struct apr_hdr     hdr;
+	u32	buf_add;
+	u32	buf_size;
+	u32	uid;
+} __packed;
+
+#define ASM_DATA_CMD_MEDIA_FORMAT_UPDATE                 0x00010BDC
+#define ASM_DATA_EVENT_ENC_SR_CM_NOTIFY                  0x00010BDE
+struct asm_stream_media_format_update{
+	struct apr_hdr hdr;
+	u32            format;
+	u32            cfg_size;
+	union {
+		struct asm_pcm_cfg         pcm_cfg;
+		struct asm_adpcm_cfg       adpcm_cfg;
+		struct asm_yadpcm_cfg      yadpcm_cfg;
+		struct asm_midi_cfg        midi_cfg;
+		struct asm_wma_cfg         wma_cfg;
+		struct asm_wmapro_cfg      wmapro_cfg;
+		struct asm_aac_cfg         aac_cfg;
+		struct asm_flac_cfg        flac_cfg;
+		struct asm_vorbis_cfg      vorbis_cfg;
+		struct asm_multi_channel_pcm_fmt_blk multi_ch_pcm_cfg;
+		struct asm_amrwbplus_cfg   amrwbplus_cfg;
+	} __attribute__((packed)) write_cfg;
+} __attribute__((packed));
+
+
+/* Command Responses */
+#define ASM_STREAM_CMDRSP_GET_ENCDEC_PARAM               0x00010C12
+struct asm_stream_cmdrsp_get_readwrite_param{
+	struct apr_hdr hdr;
+	u32            status;
+	u32            param_id;
+	u16            param_size;
+	u16            padding;
+	union {
+		struct asm_sbc_bitrate      sbc_bitrate;
+		struct asm_immed_decode aac_dec;
+	} __attribute__((packed)) read_write_cfg;
+} __attribute__((packed));
+
+
+#define ASM_SESSION_CMDRSP_GET_SESSION_TIME              0x00010BD8
+struct asm_stream_cmdrsp_get_session_time{
+	struct apr_hdr hdr;
+	u32            status;
+	u32            msw_ts;
+	u32            lsw_ts;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_WRITE_DONE                        0x00010BDF
+struct asm_data_event_write_done{
+	u32	buf_add;
+	u32            status;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_READ_DONE                         0x00010BE0
+struct asm_data_event_read_done{
+	u32            status;
+	u32            buffer_add;
+	u32            enc_frame_size;
+	u32            offset;
+	u32            msw_ts;
+	u32            lsw_ts;
+	u32            flags;
+	u32            num_frames;
+	u32            id;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_READ_COMPRESSED_DONE              0x00010DC0
+struct asm_data_event_read_compressed_done {
+	u32            status;
+	u32            buffer_add;
+	u32            enc_frame_size;
+	u32            offset;
+	u32            msw_ts;
+	u32            lsw_ts;
+	u32            flags;
+	u32            num_frames;
+	u32            id;
+} __packed;
+
+#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY               0x00010C65
+struct asm_data_event_sr_cm_change_notify {
+	u32            sample_rate;
+	u16	           no_of_channels;
+	u16            reserved;
+	u8             channel_map[8];
+} __packed;
+
+/* service level events */
+
+#define ASM_SERVICE_CMDRSP_GET_STREAM_HANDLES            0x00010C1B
+struct asm_svc_cmdrsp_get_strm_handles{
+	struct apr_hdr hdr;
+	u32            num_handles;
+	u32            stream_handles;
+} __attribute__((packed));
+
+
+#define ASM_SERVICE_CMDRSP_GET_WALLCLOCK_TIME            0x00010C1A
+struct asm_svc_cmdrsp_get_wallclock_time{
+	struct apr_hdr hdr;
+	u32            status;
+	u32            msw_ts;
+	u32            lsw_ts;
+} __attribute__((packed));
+
+/*
+ * Error code
+*/
+#define ADSP_EOK          0x00000000 /* Success / completed / no errors. */
+#define ADSP_EFAILED      0x00000001 /* General failure. */
+#define ADSP_EBADPARAM    0x00000002 /* Bad operation parameter(s). */
+#define ADSP_EUNSUPPORTED 0x00000003 /* Unsupported routine/operation. */
+#define ADSP_EVERSION     0x00000004 /* Unsupported version. */
+#define ADSP_EUNEXPECTED  0x00000005 /* Unexpected problem encountered. */
+#define ADSP_EPANIC       0x00000006 /* Unhandled problem occurred. */
+#define ADSP_ENORESOURCE  0x00000007 /* Unable to allocate resource(s). */
+#define ADSP_EHANDLE      0x00000008 /* Invalid handle. */
+#define ADSP_EALREADY     0x00000009 /* Operation is already processed. */
+#define ADSP_ENOTREADY    0x0000000A /* Operation not ready to be processed*/
+#define ADSP_EPENDING     0x0000000B /* Operation is pending completion*/
+#define ADSP_EBUSY        0x0000000C /* Operation could not be accepted or
+					 processed. */
+#define ADSP_EABORTED     0x0000000D /* Operation aborted due to an error. */
+#define ADSP_EPREEMPTED   0x0000000E /* Operation preempted by higher priority*/
+#define ADSP_ECONTINUE    0x0000000F /* Operation requests intervention
+					to complete. */
+#define ADSP_EIMMEDIATE   0x00000010 /* Operation requests immediate
+					intervention to complete. */
+#define ADSP_ENOTIMPL     0x00000011 /* Operation is not implemented. */
+#define ADSP_ENEEDMORE    0x00000012 /* Operation needs more data or resources*/
+
+/* SRS TRUMEDIA GUIDS */
+#define SRS_TRUMEDIA_TOPOLOGY_ID    0x00010D90
+#define SRS_TRUMEDIA_MODULE_ID      0x10005010
+#define SRS_TRUMEDIA_PARAMS         0x10005011
+#define SRS_TRUMEDIA_PARAMS_WOWHD   0x10005012
+#define SRS_TRUMEDIA_PARAMS_CSHP    0x10005013
+#define SRS_TRUMEDIA_PARAMS_HPF     0x10005014
+#define SRS_TRUMEDIA_PARAMS_PEQ     0x10005015
+#define SRS_TRUMEDIA_PARAMS_HL      0x10005016
+
+/* SRS STUDIO SOUND 3D GUIDS */
+#define SRS_SS3D_TOPOLOGY_ID        0x00010720
+#define SRS_SS3D_MODULE_ID          0x10005020
+#define SRS_SS3D_PARAMS             0x10005021
+#define SRS_SS3D_PARAMS_CTRL        0x10005022
+#define SRS_SS3D_PARAMS_FILTER      0x10005023
+
+/* SRS ALSA CMD MASKS */
+#define SRS_CMD_UPLOAD              0x7FFF0000
+#define SRS_PARAM_INDEX_MASK        0x80000000
+#define SRS_PARAM_OFFSET_MASK       0x3FFF0000
+#define SRS_PARAM_VALUE_MASK        0x0000FFFF
+
+/* SRS TRUMEDIA start */
+#define SRS_ID_GLOBAL               0x00000001
+#define SRS_ID_WOWHD                0x00000002
+#define SRS_ID_CSHP                 0x00000003
+#define SRS_ID_HPF                  0x00000004
+#define SRS_ID_PEQ                  0x00000005
+#define SRS_ID_HL                   0x00000006
+
+struct srs_trumedia_params_GLOBAL {
+	uint8_t                  v1;
+	uint8_t                  v2;
+	uint8_t                  v3;
+	uint8_t                  v4;
+	uint8_t                  v5;
+	uint8_t                  v6;
+	uint8_t                  v7;
+	uint8_t                  v8;
+} __packed;
+
+struct srs_trumedia_params_WOWHD {
+	uint32_t				v1;
+	uint16_t				v2;
+	uint16_t				v3;
+	uint16_t				v4;
+	uint16_t				v5;
+	uint16_t				v6;
+	uint16_t				v7;
+	uint16_t				v8;
+	uint16_t				v____A1;
+	uint32_t				v9;
+	uint16_t				v10;
+	uint16_t				v11;
+	uint32_t				v12[16];
+} __packed;
+
+struct srs_trumedia_params_CSHP {
+	uint32_t				v1;
+	uint16_t				v2;
+	uint16_t				v3;
+	uint16_t				v4;
+	uint16_t				v5;
+	uint16_t				v6;
+	uint16_t				v____A1;
+	uint32_t				v7;
+	uint16_t				v8;
+	uint16_t				v9;
+	uint32_t				v10[16];
+} __packed;
+
+struct srs_trumedia_params_HPF {
+	uint32_t				v1;
+	uint32_t				v2[26];
+} __packed;
+
+struct srs_trumedia_params_PEQ {
+	uint32_t				v1;
+	uint16_t				v2;
+	uint16_t				v3;
+	uint16_t				v4;
+	uint16_t				v____A1;
+	uint32_t				v5[26];
+	uint32_t				v6[26];
+} __packed;
+
+struct srs_trumedia_params_HL {
+	uint16_t				v1;
+	uint16_t				v2;
+	uint16_t				v3;
+	uint16_t				v____A1;
+	int32_t					v4;
+	uint32_t				v5;
+	uint16_t				v6;
+	uint16_t				v____A2;
+	uint32_t				v7;
+} __packed;
+
+struct srs_trumedia_params {
+	struct srs_trumedia_params_GLOBAL	global;
+	struct srs_trumedia_params_WOWHD	wowhd;
+	struct srs_trumedia_params_CSHP		cshp;
+	struct srs_trumedia_params_HPF		hpf;
+	struct srs_trumedia_params_PEQ		peq;
+	struct srs_trumedia_params_HL		hl;
+} __packed;
+
+int srs_trumedia_open(int port_id, int srs_tech_id, void *srs_params);
+/* SRS TruMedia end */
+
+/* SRS Studio Sound 3D start */
+#define SRS_ID_SS3D_GLOBAL	0x00000001
+#define SRS_ID_SS3D_CTRL	0x00000002
+#define SRS_ID_SS3D_FILTER	0x00000003
+
+struct srs_SS3D_params_GLOBAL {
+	uint8_t                  v1;
+	uint8_t                  v2;
+	uint8_t                  v3;
+	uint8_t                  v4;
+	uint8_t                  v5;
+	uint8_t                  v6;
+	uint8_t                  v7;
+	uint8_t                  v8;
+} __packed;
+
+struct srs_SS3D_ctrl_params {
+	uint8_t				v[236];
+} __packed;
+
+struct srs_SS3D_filter_params {
+	uint8_t				v[28 + 2752];
+} __packed;
+
+struct srs_SS3D_params {
+	struct srs_SS3D_params_GLOBAL   global;
+	struct srs_SS3D_ctrl_params     ss3d;
+	struct srs_SS3D_filter_params   ss3d_f;
+} __packed;
+
+int srs_ss3d_open(int port_id, int srs_tech_id, void *srs_params);
+/* SRS Studio Sound 3D end */
+#endif /*_APR_AUDIO_H_*/
diff --git a/include/sound/audio_cal_utils.h b/include/sound/audio_cal_utils.h
new file mode 100644
index 000000000000..b28b3bdf4e83
--- /dev/null
+++ b/include/sound/audio_cal_utils.h
@@ -0,0 +1,102 @@
+/* Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef _AUDIO_CAL_UTILS_H
+#define _AUDIO_CAL_UTILS_H
+
+#include <linux/msm_ion.h>
+#include <linux/msm_audio_ion.h>
+#include <linux/msm_audio_calibration.h>
+#include "audio_calibration.h"
+
+struct cal_data {
+	size_t		size;
+	void		*kvaddr;
+	phys_addr_t	paddr;
+};
+
+struct mem_map_data {
+	size_t			map_size;
+	int32_t			q6map_handle;
+	int32_t			ion_map_handle;
+	struct ion_client	*ion_client;
+	struct ion_handle	*ion_handle;
+};
+
+struct cal_block_data {
+	size_t			client_info_size;
+	void			*client_info;
+	void			*cal_info;
+	struct list_head	list;
+	struct cal_data		cal_data;
+	struct mem_map_data	map_data;
+	int32_t			buffer_number;
+};
+
+struct cal_util_callbacks {
+	int (*map_cal)
+		(int32_t cal_type, struct cal_block_data *cal_block);
+	int (*unmap_cal)
+		(int32_t cal_type, struct cal_block_data *cal_block);
+	bool (*match_block)
+		(struct cal_block_data *cal_block, void *user_data);
+};
+
+struct cal_type_info {
+	struct audio_cal_reg		reg;
+	struct cal_util_callbacks	cal_util_callbacks;
+};
+
+struct cal_type_data {
+	struct cal_type_info		info;
+	struct mutex			lock;
+	struct list_head		cal_blocks;
+};
+
+
+/* to register & degregister with cal util driver */
+int cal_utils_create_cal_types(int num_cal_types,
+			struct cal_type_data **cal_type,
+			struct cal_type_info *info);
+void cal_utils_destroy_cal_types(int num_cal_types,
+			struct cal_type_data **cal_type);
+
+/* common functions for callbacks */
+int cal_utils_alloc_cal(size_t data_size, void *data,
+			struct cal_type_data *cal_type,
+			size_t client_info_size, void *client_info);
+int cal_utils_dealloc_cal(size_t data_size, void *data,
+			struct cal_type_data *cal_type);
+int cal_utils_set_cal(size_t data_size, void *data,
+			struct cal_type_data *cal_type,
+			size_t client_info_size, void *client_info);
+
+/* use for SSR */
+void cal_utils_clear_cal_block_q6maps(int num_cal_types,
+					struct cal_type_data **cal_type);
+
+
+/* common matching functions used to add blocks */
+bool cal_utils_match_buf_num(struct cal_block_data *cal_block,
+					void *user_data);
+
+/* common matching functions to find cal blocks */
+struct cal_block_data *cal_utils_get_only_cal_block(
+			struct cal_type_data *cal_type);
+
+/* Size of calibration specific data */
+size_t get_cal_info_size(int32_t cal_type);
+size_t get_user_cal_type_size(int32_t cal_type);
+
+/* Version of the cal type*/
+int32_t cal_utils_get_cal_type_version(void *cal_type_data);
+#endif
diff --git a/include/sound/audio_calibration.h b/include/sound/audio_calibration.h
new file mode 100644
index 000000000000..5f6b5e37aa5c
--- /dev/null
+++ b/include/sound/audio_calibration.h
@@ -0,0 +1,40 @@
+/* Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef _AUDIO_CALIBRATION_H
+#define _AUDIO_CALIBRATION_H
+
+#include <linux/msm_audio_calibration.h>
+
+/* Used by driver in buffer_number field to notify client
+ * To update all blocks, for example: freeing all memory */
+#define ALL_CAL_BLOCKS		-1
+
+
+struct audio_cal_callbacks {
+	int (*alloc) (int32_t cal_type, size_t data_size, void *data);
+	int (*dealloc) (int32_t cal_type, size_t data_size, void *data);
+	int (*pre_cal) (int32_t cal_type, size_t data_size, void *data);
+	int (*set_cal) (int32_t cal_type, size_t data_size, void *data);
+	int (*get_cal) (int32_t cal_type, size_t data_size, void *data);
+	int (*post_cal) (int32_t cal_type, size_t data_size, void *data);
+};
+
+struct audio_cal_reg {
+	int32_t				cal_type;
+	struct audio_cal_callbacks	callbacks;
+};
+
+int audio_cal_register(int num_cal_types, struct audio_cal_reg *reg_data);
+int audio_cal_deregister(int num_cal_types, struct audio_cal_reg *reg_data);
+
+#endif
diff --git a/include/sound/audio_slimslave.h b/include/sound/audio_slimslave.h
new file mode 100644
index 000000000000..316a5573f5b4
--- /dev/null
+++ b/include/sound/audio_slimslave.h
@@ -0,0 +1,18 @@
+#ifndef __AUDIO_SLIMSLAVE_H__
+#define __AUDIO_SLIMSLAVE_H__
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define AUDIO_SLIMSLAVE_IOCTL_NAME "audio_slimslave"
+#define AUDIO_SLIMSLAVE_MAGIC 'S'
+
+#define AUDIO_SLIMSLAVE_IOCTL_UNVOTE	_IO(AUDIO_SLIMSLAVE_MAGIC, 0x00)
+#define AUDIO_SLIMSLAVE_IOCTL_VOTE	_IO(AUDIO_SLIMSLAVE_MAGIC, 0x01)
+
+enum {
+	AUDIO_SLIMSLAVE_UNVOTE,
+	AUDIO_SLIMSLAVE_VOTE
+};
+
+#endif
diff --git a/include/sound/cpe_cmi.h b/include/sound/cpe_cmi.h
new file mode 100644
index 000000000000..5ce52ae912d4
--- /dev/null
+++ b/include/sound/cpe_cmi.h
@@ -0,0 +1,477 @@
+/*
+ * Copyright (c) 2014-2015, Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __CPE_CMI_H__
+#define __CPE_CMI_H__
+
+#include <linux/types.h>
+
+#define CPE_AFE_PORT_1_TX 1
+#define CPE_AFE_PORT_ID_2_OUT 0x02
+#define CMI_INBAND_MESSAGE_SIZE 127
+
+/*
+ * Multiple mad types can be supported at once.
+ * these values can be OR'ed to form the set of
+ * supported mad types
+ */
+#define MAD_TYPE_AUDIO (1 << 0)
+#define MAD_TYPE_BEACON (1 << 1)
+#define MAD_TYPE_ULTRASND (1 << 2)
+
+/* Core service command opcodes */
+#define CPE_CORE_SVC_CMD_SHARED_MEM_ALLOC	(0x3001)
+#define CPE_CORE_SVC_CMDRSP_SHARED_MEM_ALLOC	(0x3002)
+#define CPE_CORE_SVC_CMD_SHARED_MEM_DEALLOC	(0x3003)
+#define CPE_CORE_SVC_CMD_DRAM_ACCESS_REQ	(0x3004)
+#define CPE_CORE_SVC_EVENT_SYSTEM_BOOT		(0x3005)
+/* core service command opcodes for WCD9335 */
+#define CPE_CORE_SVC_CMD_CFG_CLK_PLAN		(0x3006)
+#define CPE_CORE_SVC_CMD_CLK_FREQ_REQUEST	(0x3007)
+
+#define CPE_BOOT_SUCCESS 0x00
+#define CPE_BOOT_FAILED 0x01
+
+#define CPE_CORE_VERSION_SYSTEM_BOOT_EVENT 0x01
+
+/* LSM Service command opcodes */
+#define CPE_LSM_SESSION_CMD_OPEN_TX		(0x2000)
+#define CPE_LSM_SESSION_CMD_SET_PARAMS		(0x2001)
+#define CPE_LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x2002)
+#define CPE_LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x2003)
+#define CPE_LSM_SESSION_CMD_START		(0x2004)
+#define CPE_LSM_SESSION_CMD_STOP		(0x2005)
+#define CPE_LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x2006)
+#define CPE_LSM_SESSION_CMD_CLOSE_TX		(0x2007)
+#define CPE_LSM_SESSION_CMD_SHARED_MEM_ALLOC	(0x2008)
+#define CPE_LSM_SESSION_CMDRSP_SHARED_MEM_ALLOC (0x2009)
+#define CPE_LSM_SESSION_CMD_SHARED_MEM_DEALLOC	(0x200A)
+#define CPE_LSM_SESSION_CMD_TX_BUFF_OUTPUT_CONFIG (0x200f)
+#define CPE_LSM_SESSION_CMD_OPEN_TX_V2		(0x200D)
+#define CPE_LSM_SESSION_CMD_SET_PARAMS_V2	(0x200E)
+
+/* LSM Service module and param IDs */
+#define CPE_LSM_MODULE_ID_VOICE_WAKEUP		(0x00012C00)
+#define CPE_LSM_MODULE_ID_VOICE_WAKEUP_V2	(0x00012C0D)
+#define CPE_LSM_MODULE_FRAMEWORK		(0x00012C0E)
+
+#define CPE_LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01)
+#define CPE_LSM_PARAM_ID_OPERATION_MODE		(0x00012C02)
+#define CPE_LSM_PARAM_ID_GAIN			(0x00012C03)
+#define CPE_LSM_PARAM_ID_CONNECT_TO_PORT	(0x00012C04)
+#define CPE_LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS	(0x00012C07)
+
+/* LSM LAB command opcodes */
+#define CPE_LSM_SESSION_CMD_EOB		0x0000200B
+#define CPE_LSM_MODULE_ID_LAB		0x00012C08
+/* used for enable/disable lab*/
+#define CPE_LSM_PARAM_ID_LAB_ENABLE	0x00012C09
+/* used for T in LAB config DSP internal buffer*/
+#define CPE_LSM_PARAM_ID_LAB_CONFIG	0x00012C0A
+#define CPE_LSM_PARAM_ID_REGISTER_SOUND_MODEL	(0x00012C14)
+#define CPE_LSM_PARAM_ID_DEREGISTER_SOUND_MODEL	(0x00012C15)
+
+/* AFE Service command opcodes */
+#define CPE_AFE_PORT_CMD_START			(0x1001)
+#define CPE_AFE_PORT_CMD_STOP			(0x1002)
+#define CPE_AFE_PORT_CMD_SUSPEND		(0x1003)
+#define CPE_AFE_PORT_CMD_RESUME			(0x1004)
+#define CPE_AFE_PORT_CMD_SHARED_MEM_ALLOC	(0x1005)
+#define CPE_AFE_PORT_CMDRSP_SHARED_MEM_ALLOC	(0x1006)
+#define CPE_AFE_PORT_CMD_SHARED_MEM_DEALLOC	(0x1007)
+#define CPE_AFE_PORT_CMD_GENERIC_CONFIG		(0x1008)
+#define CPE_AFE_SVC_CMD_LAB_MODE		(0x1009)
+
+/* AFE Service module and param IDs */
+#define CPE_AFE_CMD_SET_PARAM			(0x1000)
+#define CPE_AFE_MODULE_ID_SW_MAD		(0x0001022D)
+#define CPE_AFE_PARAM_ID_SW_MAD_CFG		(0x0001022E)
+#define CPE_AFE_PARAM_ID_SVM_MODEL		(0x0001022F)
+
+#define CPE_AFE_MODULE_HW_MAD			(0x00010230)
+#define CPE_AFE_PARAM_ID_HW_MAD_CTL		(0x00010232)
+#define CPE_AFE_PARAM_ID_HW_MAD_CFG		(0x00010231)
+
+#define CPE_AFE_MODULE_AUDIO_DEV_INTERFACE	(0x0001020C)
+#define CPE_AFE_PARAM_ID_GENERIC_PORT_CONFIG	(0x00010253)
+
+#define CPE_CMI_BASIC_RSP_OPCODE	(0x0001)
+#define CPE_HDR_MAX_PLD_SIZE	(0x7F)
+
+#define CMI_OBM_FLAG_IN_BAND	0
+#define CMI_OBM_FLAG_OUT_BAND	1
+
+#define CMI_SHMEM_ALLOC_FAILED 0xff
+
+/*
+ * Future Service ID's can be added one line
+ * before the CMI_CPE_SERVICE_ID_MAX
+ */
+enum {
+	CMI_CPE_SERVICE_ID_MIN = 0,
+	CMI_CPE_CORE_SERVICE_ID,
+	CMI_CPE_AFE_SERVICE_ID,
+	CMI_CPE_LSM_SERVICE_ID,
+	CMI_CPE_SERVICE_ID_MAX,
+};
+
+#define CPE_LSM_SESSION_ID_MAX 1
+
+#define IS_VALID_SESSION_ID(s_id) \
+	(s_id <= CPE_LSM_SESSION_ID_MAX)
+
+#define IS_VALID_SERVICE_ID(s_id) \
+	(s_id > CMI_CPE_SERVICE_ID_MIN && \
+	 s_id < CMI_CPE_SERVICE_ID_MAX)
+
+#define IS_VALID_PLD_SIZE(p_size) \
+	(p_size <= CPE_HDR_MAX_PLD_SIZE)
+
+#define CMI_HDR_SET_OPCODE(hdr, cmd) (hdr->opcode = cmd)
+
+
+#define CMI_HDR_SET(hdr_info, mask, shift, value) \
+		(hdr_info = (((hdr_info) & ~(mask)) | \
+			((value << shift) & mask)))
+
+#define SVC_ID_SHIFT 4
+#define SVC_ID_MASK (0x07 << SVC_ID_SHIFT)
+
+#define SESSION_ID_SHIFT 0
+#define SESSION_ID_MASK (0x0F << SESSION_ID_SHIFT)
+
+#define PAYLD_SIZE_SHIFT 0
+#define PAYLD_SIZE_MASK (0x7F << PAYLD_SIZE_SHIFT)
+
+#define OBM_FLAG_SHIFT 7
+#define OBM_FLAG_MASK (1 << OBM_FLAG_SHIFT)
+
+#define VERSION_SHIFT 7
+#define VERSION_MASK (1 << VERSION_SHIFT)
+
+#define CMI_HDR_SET_SERVICE(hdr, s_id) \
+		CMI_HDR_SET(hdr->hdr_info, SVC_ID_MASK,\
+			    SVC_ID_SHIFT, s_id)
+#define CMI_HDR_GET_SERVICE(hdr) \
+		((hdr->hdr_info >> SVC_ID_SHIFT) & \
+			(SVC_ID_MASK >> SVC_ID_SHIFT))
+
+
+#define CMI_HDR_SET_SESSION(hdr, s_id) \
+		CMI_HDR_SET(hdr->hdr_info, SESSION_ID_MASK,\
+			    SESSION_ID_SHIFT, s_id)
+
+#define CMI_HDR_GET_SESSION_ID(hdr) \
+		((hdr->hdr_info >> SESSION_ID_SHIFT) & \
+			 (SESSION_ID_MASK >> SESSION_ID_SHIFT))
+
+#define CMI_GET_HEADER(msg)	((struct cmi_hdr *)(msg))
+#define CMI_GET_PAYLOAD(msg)	((void *)(CMI_GET_HEADER(msg) + 1))
+#define CMI_GET_OPCODE(msg)	(CMI_GET_HEADER(msg)->opcode)
+
+#define CMI_HDR_SET_VERSION(hdr, ver) \
+		CMI_HDR_SET(hdr->hdr_info, VERSION_MASK, \
+				VERSION_SHIFT, ver)
+
+#define CMI_HDR_SET_PAYLOAD_SIZE(hdr, p_size) \
+		CMI_HDR_SET(hdr->pld_info, PAYLD_SIZE_MASK, \
+			    PAYLD_SIZE_SHIFT, p_size)
+
+#define CMI_HDR_GET_PAYLOAD_SIZE(hdr) \
+		((hdr->pld_info >> PAYLD_SIZE_SHIFT) & \
+			(PAYLD_SIZE_MASK >> PAYLD_SIZE_SHIFT))
+
+#define CMI_HDR_SET_OBM(hdr, obm_flag) \
+		CMI_HDR_SET(hdr->pld_info, OBM_FLAG_MASK, \
+			    OBM_FLAG_SHIFT, obm_flag)
+
+#define CMI_HDR_GET_OBM_FLAG(hdr) \
+	((hdr->pld_info >> OBM_FLAG_SHIFT) & \
+		(OBM_FLAG_MASK >> OBM_FLAG_SHIFT))
+
+struct cmi_hdr {
+	/*
+	 * bits 0:3 is session id
+	 * bits 4:6 is service id
+	 * bit 7 is the version flag
+	 */
+	u8 hdr_info;
+
+	/*
+	 * bits 0:6 is payload size in case of in-band message
+	 * bits 0:6 is size (OBM message size)
+	 * bit 7 is the OBM flag
+	 */
+	u8 pld_info;
+
+	/* 16 bit command opcode */
+	u16 opcode;
+} __packed;
+
+union cpe_addr {
+	u64 msw_lsw;
+	void *kvaddr;
+} __packed;
+
+struct cmi_obm {
+	u32 version;
+	u32 size;
+	union cpe_addr data_ptr;
+	u32 mem_handle;
+} __packed;
+
+struct cmi_obm_msg {
+	struct cmi_hdr hdr;
+	struct cmi_obm pld;
+} __packed;
+
+struct cmi_core_svc_event_system_boot {
+	u8 status;
+	u8 version;
+	u16 sfr_buff_size;
+	u32 sfr_buff_address;
+} __packed;
+
+struct cmi_core_svc_cmd_shared_mem_alloc {
+	u32 size;
+} __packed;
+
+struct cmi_core_svc_cmdrsp_shared_mem_alloc {
+	u32 addr;
+} __packed;
+
+struct cmi_core_svc_cmd_clk_freq_request {
+	u32 clk_freq;
+} __packed;
+
+struct cmi_msg_transport {
+	u32 size;
+	u32 addr;
+} __packed;
+
+struct cmi_basic_rsp_result {
+	u8 status;
+} __packed;
+
+struct cpe_lsm_cmd_open_tx {
+	struct cmi_hdr	hdr;
+	u16 app_id;
+	u16 reserved;
+	u32 sampling_rate;
+} __packed;
+
+struct cpe_lsm_cmd_open_tx_v2 {
+	struct cmi_hdr hdr;
+	u32 topology_id;
+} __packed;
+
+struct cpe_cmd_shmem_alloc {
+	struct cmi_hdr hdr;
+	u32 size;
+} __packed;
+
+struct cpe_cmdrsp_shmem_alloc {
+	struct cmi_hdr hdr;
+	u32 addr;
+} __packed;
+
+struct cpe_cmd_shmem_dealloc {
+	struct cmi_hdr	hdr;
+	u32 addr;
+} __packed;
+
+struct cpe_lsm_event_detect_v2 {
+	struct cmi_hdr	hdr;
+	u8 detection_status;
+	u8 size;
+	u8 payload[0];
+} __packed;
+
+struct cpe_lsm_psize_res {
+	u16 param_size;
+	u16 reserved;
+} __packed;
+
+union cpe_lsm_param_size {
+	u32 param_size;
+	struct cpe_lsm_psize_res sr;
+} __packed;
+
+struct cpe_param_data {
+	u32 module_id;
+	u32 param_id;
+	union cpe_lsm_param_size p_size;
+} __packed;
+
+struct cpe_lsm_param_epd_thres {
+	struct cmi_hdr hdr;
+	struct cpe_param_data param;
+	u32 minor_version;
+	u32 epd_begin;
+	u32 epd_end;
+} __packed;
+
+struct cpe_lsm_param_gain {
+	struct cmi_hdr hdr;
+	struct cpe_param_data param;
+	u32 minor_version;
+	u16 gain;
+	u16 reserved;
+} __packed;
+
+struct cpe_afe_hw_mad_ctrl {
+	struct cpe_param_data param;
+	u32 minor_version;
+	u16 mad_type;
+	u16 mad_enable;
+} __packed;
+
+struct cpe_afe_port_cfg {
+	struct cpe_param_data param;
+	u32 minor_version;
+	u16 bit_width;
+	u16 num_channels;
+	u32 sample_rate;
+} __packed;
+
+struct cpe_afe_cmd_port_cfg {
+	struct cmi_hdr hdr;
+	u8 bit_width;
+	u8 num_channels;
+	u16 buffer_size;
+	u32 sample_rate;
+} __packed;
+
+struct cpe_afe_params {
+	struct cmi_hdr hdr;
+	struct cpe_afe_hw_mad_ctrl hw_mad_ctrl;
+	struct cpe_afe_port_cfg port_cfg;
+} __packed;
+
+struct cpe_afe_svc_cmd_mode {
+	struct cmi_hdr hdr;
+	u8 mode;
+} __packed;
+
+struct cpe_lsm_param_opmode {
+	struct cmi_hdr hdr;
+	struct cpe_param_data param;
+	u32 minor_version;
+	u16 mode;
+	u16 reserved;
+} __packed;
+
+struct cpe_lsm_param_connectport {
+	struct cmi_hdr hdr;
+	struct cpe_param_data param;
+	u32 minor_version;
+	u16 afe_port_id;
+	u16 reserved;
+} __packed;
+
+/*
+ * This cannot be sent to CPE as is,
+ * need to append the conf_levels dynamically
+ */
+struct cpe_lsm_conf_level {
+	struct cmi_hdr hdr;
+	struct cpe_param_data param;
+	u8 num_active_models;
+} __packed;
+
+struct cpe_lsm_output_format_cfg {
+	struct cmi_hdr hdr;
+	u8 format;
+	u8 packing;
+	u8 data_path_events;
+} __packed;
+
+struct cpe_lsm_lab_enable {
+	struct cpe_param_data param;
+	u16 enable;
+	u16 reserved;
+} __packed;
+
+struct cpe_lsm_control_lab {
+	struct cmi_hdr hdr;
+	struct cpe_lsm_lab_enable lab_enable;
+} __packed;
+
+struct cpe_lsm_lab_config {
+	struct cpe_param_data param;
+	u32 minor_ver;
+	u32 latency;
+} __packed;
+
+struct cpe_lsm_lab_latency_config {
+	struct cmi_hdr hdr;
+	struct cpe_lsm_lab_config latency_cfg;
+} __packed;
+
+
+#define CPE_PARAM_LSM_LAB_LATENCY_SIZE (\
+				sizeof(struct cpe_lsm_lab_latency_config) - \
+				sizeof(struct cmi_hdr))
+#define PARAM_SIZE_LSM_LATENCY_SIZE (\
+					sizeof(struct cpe_lsm_lab_config) - \
+					sizeof(struct cpe_param_data))
+#define CPE_PARAM_SIZE_LSM_LAB_CONTROL (\
+				sizeof(struct cpe_lsm_control_lab) - \
+				sizeof(struct cmi_hdr))
+#define PARAM_SIZE_LSM_CONTROL_SIZE (sizeof(struct cpe_lsm_lab_enable) - \
+					sizeof(struct cpe_param_data))
+#define PARAM_SIZE_AFE_HW_MAD_CTRL (sizeof(struct cpe_afe_hw_mad_ctrl) - \
+				sizeof(struct cpe_param_data))
+#define PARAM_SIZE_AFE_PORT_CFG (sizeof(struct cpe_afe_port_cfg) - \
+				 sizeof(struct cpe_param_data))
+#define CPE_AFE_PARAM_PAYLOAD_SIZE (sizeof(struct cpe_afe_params) - \
+				sizeof(struct cmi_hdr))
+
+#define OPEN_CMD_PAYLOAD_SIZE (sizeof(struct cpe_lsm_cmd_open_tx) - \
+			       sizeof(struct cmi_hdr))
+#define OPEN_V2_CMD_PAYLOAD_SIZE (sizeof(struct cpe_lsm_cmd_open_tx_v2) - \
+			       sizeof(struct cmi_hdr))
+#define SHMEM_ALLOC_CMD_PLD_SIZE (sizeof(struct cpe_cmd_shmem_alloc) - \
+				      sizeof(struct cmi_hdr))
+
+#define SHMEM_DEALLOC_CMD_PLD_SIZE (sizeof(struct cpe_cmd_shmem_dealloc) - \
+				      sizeof(struct cmi_hdr))
+#define OUT_FMT_CFG_CMD_PAYLOAD_SIZE ( \
+		sizeof(struct cpe_lsm_output_format_cfg) - \
+		sizeof(struct cmi_hdr))
+
+#define CPE_AFE_CMD_PORT_CFG_PAYLOAD_SIZE \
+		(sizeof(struct cpe_afe_cmd_port_cfg) - \
+		 sizeof(struct cmi_hdr))
+
+#define CPE_AFE_CMD_MODE_PAYLOAD_SIZE \
+		(sizeof(struct cpe_afe_svc_cmd_mode) - \
+		 sizeof(struct cmi_hdr))
+#define CPE_CMD_EPD_THRES_PLD_SIZE (sizeof(struct cpe_lsm_param_epd_thres) - \
+				    sizeof(struct cmi_hdr))
+#define CPE_EPD_THRES_PARAM_SIZE ((CPE_CMD_EPD_THRES_PLD_SIZE) - \
+				  sizeof(struct cpe_param_data))
+#define CPE_CMD_OPMODE_PLD_SIZE (sizeof(struct cpe_lsm_param_opmode) - \
+				 sizeof(struct cmi_hdr))
+#define CPE_OPMODE_PARAM_SIZE ((CPE_CMD_OPMODE_PLD_SIZE) -\
+			       sizeof(struct cpe_param_data))
+#define CPE_CMD_CONNECTPORT_PLD_SIZE \
+	(sizeof(struct cpe_lsm_param_connectport) - \
+	 sizeof(struct cmi_hdr))
+#define CPE_CONNECTPORT_PARAM_SIZE ((CPE_CMD_CONNECTPORT_PLD_SIZE) - \
+				    sizeof(struct cpe_param_data))
+#define CPE_CMD_GAIN_PLD_SIZE (sizeof(struct cpe_lsm_param_gain) - \
+			       sizeof(struct cmi_hdr))
+#define CPE_GAIN_PARAM_SIZE ((CPE_CMD_GAIN_PLD_SIZE) - \
+			     sizeof(struct cpe_param_data))
+#endif /* __CPE_CMI_H__ */
diff --git a/include/sound/cpe_core.h b/include/sound/cpe_core.h
new file mode 100644
index 000000000000..3e38a27212a5
--- /dev/null
+++ b/include/sound/cpe_core.h
@@ -0,0 +1,167 @@
+/*
+ * Copyright (c) 2013-2015, Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __CPE_CORE_H__
+#define __CPE_CORE_H__
+
+#include <linux/types.h>
+#include <linux/wait.h>
+#include <linux/dma-mapping.h>
+#include <sound/lsm_params.h>
+
+enum {
+	CMD_INIT_STATE = 0,
+	CMD_SENT,
+	CMD_RESP_RCVD,
+};
+
+enum wcd_cpe_event {
+	WCD_CPE_PRE_ENABLE = 1,
+	WCD_CPE_POST_ENABLE,
+	WCD_CPE_PRE_DISABLE,
+	WCD_CPE_POST_DISABLE,
+};
+
+struct wcd_cpe_afe_port_cfg {
+	u8 port_id;
+	u16 bit_width;
+	u16 num_channels;
+	u32 sample_rate;
+};
+
+struct lsm_out_fmt_cfg {
+	u8 format;
+	u8 pack_mode;
+	u8 data_path_events;
+	u8 transfer_mode;
+};
+
+struct cpe_lsm_session {
+	/* sound model related */
+	void *snd_model_data;
+	u8 *conf_levels;
+	void *cmi_reg_handle;
+
+	/* Clients private data */
+	void *priv_d;
+
+	void (*event_cb) (void *priv_data,
+			  u8 detect_status,
+			  u8 size, u8 *payload);
+
+	struct completion cmd_comp;
+	struct wcd_cpe_afe_port_cfg afe_port_cfg;
+	struct wcd_cpe_afe_port_cfg afe_out_port_cfg;
+	struct mutex lsm_lock;
+
+	u32 snd_model_size;
+	u32 lsm_mem_handle;
+	u16 cmd_err_code;
+	u8 id;
+	u8 num_confidence_levels;
+	u16 afe_out_port_id;
+	struct task_struct *lsm_lab_thread;
+	bool started;
+
+	u32 lab_enable;
+	struct lsm_out_fmt_cfg out_fmt_cfg;
+
+	bool is_topology_used;
+};
+
+struct wcd_cpe_afe_ops {
+	int (*afe_set_params) (void *core_handle,
+			       struct wcd_cpe_afe_port_cfg *cfg);
+
+	int (*afe_port_start) (void *core_handle,
+			       struct wcd_cpe_afe_port_cfg *cfg);
+
+	int (*afe_port_stop) (void *core_handle,
+			       struct wcd_cpe_afe_port_cfg *cfg);
+
+	int (*afe_port_suspend) (void *core_handle,
+			       struct wcd_cpe_afe_port_cfg *cfg);
+
+	int (*afe_port_resume) (void *core_handle,
+			       struct wcd_cpe_afe_port_cfg *cfg);
+
+	int (*afe_port_cmd_cfg)(void *core_handle,
+				struct wcd_cpe_afe_port_cfg *cfg);
+};
+
+struct wcd_cpe_lsm_ops {
+
+	struct cpe_lsm_session *(*lsm_alloc_session)
+			(void *core_handle, void *lsm_priv_d,
+			 void (*event_cb) (void *priv_data,
+					   u8 detect_status,
+					   u8 size, u8 *payload));
+
+	int (*lsm_dealloc_session)
+		(void *core_handle, struct cpe_lsm_session *);
+
+	int (*lsm_open_tx) (void *core_handle,
+			    struct cpe_lsm_session *, u16, u16);
+
+	int (*lsm_close_tx) (void *core_handle,
+			     struct cpe_lsm_session *);
+
+	int (*lsm_shmem_alloc) (void *core_handle,
+				struct cpe_lsm_session *, u32 size);
+
+	int (*lsm_shmem_dealloc) (void *core_handle,
+				  struct cpe_lsm_session *);
+
+	int (*lsm_register_snd_model) (void *core_handle,
+				       struct cpe_lsm_session *,
+				       enum lsm_detection_mode, bool);
+
+	int (*lsm_deregister_snd_model) (void *core_handle,
+					 struct cpe_lsm_session *);
+
+	int (*lsm_get_afe_out_port_id)(void *core_handle,
+			       struct cpe_lsm_session *session);
+
+	int (*lsm_start) (void *core_handle,
+			  struct cpe_lsm_session *);
+
+	int (*lsm_stop) (void *core_handle,
+			 struct cpe_lsm_session *);
+
+	int (*lsm_lab_control)(void *core_handle,
+			       struct cpe_lsm_session *session,
+			       bool enable);
+
+	int (*lab_ch_setup)(void *core_handle,
+				   struct cpe_lsm_session *session,
+				   enum wcd_cpe_event event);
+
+	int (*lsm_set_data) (void *core_handle,
+			struct cpe_lsm_session *session,
+			enum lsm_detection_mode detect_mode,
+			bool detect_failure);
+	int (*lsm_set_fmt_cfg)(void *core_handle,
+			struct cpe_lsm_session *session);
+	int (*lsm_set_one_param)(void *core_handle,
+			struct cpe_lsm_session *session,
+			struct lsm_params_info *p_info,
+			void *data, enum LSM_PARAM_TYPE param_type);
+	void (*lsm_get_snd_model_offset)
+		(void *core_handle, struct cpe_lsm_session *,
+		 size_t *offset);
+};
+
+int wcd_cpe_get_lsm_ops(struct wcd_cpe_lsm_ops *);
+int wcd_cpe_get_afe_ops(struct wcd_cpe_afe_ops *);
+void *wcd_cpe_get_core_handle(struct snd_soc_codec *);
+#endif
diff --git a/include/sound/cpe_err.h b/include/sound/cpe_err.h
new file mode 100644
index 000000000000..e31fa757aa4d
--- /dev/null
+++ b/include/sound/cpe_err.h
@@ -0,0 +1,166 @@
+/*
+ * Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __CPE_ERR__
+#define __CPE_ERR__
+
+#include <linux/errno.h>
+
+/* ERROR CODES */
+/* Success. The operation completed with no errors. */
+#define CPE_EOK          0x00000000
+/* General failure. */
+#define CPE_EFAILED      0x00000001
+/* Bad operation parameter. */
+#define CPE_EBADPARAM    0x00000002
+/* Unsupported routine or operation. */
+#define CPE_EUNSUPPORTED 0x00000003
+/* Unsupported version. */
+#define CPE_EVERSION     0x00000004
+/* Unexpected problem encountered. */
+#define CPE_EUNEXPECTED  0x00000005
+/* Unhandled problem occurred. */
+#define CPE_EPANIC       0x00000006
+/* Unable to allocate resource. */
+#define CPE_ENORESOURCE  0x00000007
+/* Invalid handle. */
+#define CPE_EHANDLE      0x00000008
+/* Operation is already processed. */
+#define CPE_EALREADY     0x00000009
+/* Operation is not ready to be processed. */
+#define CPE_ENOTREADY    0x0000000A
+/* Operation is pending completion. */
+#define CPE_EPENDING     0x0000000B
+/* Operation could not be accepted or processed. */
+#define CPE_EBUSY        0x0000000C
+/* Operation aborted due to an error. */
+#define CPE_EABORTED     0x0000000D
+/* Operation preempted by a higher priority. */
+#define CPE_EPREEMPTED   0x0000000E
+/* Operation requests intervention to complete. */
+#define CPE_ECONTINUE    0x0000000F
+/* Operation requests immediate intervention to complete. */
+#define CPE_EIMMEDIATE   0x00000010
+/* Operation is not implemented. */
+#define CPE_ENOTIMPL     0x00000011
+/* Operation needs more data or resources. */
+#define CPE_ENEEDMORE    0x00000012
+/* Operation does not have memory. */
+#define CPE_ENOMEMORY    0x00000014
+/* Item does not exist. */
+#define CPE_ENOTEXIST    0x00000015
+/* Operation is finished. */
+#define CPE_ETERMINATED  0x00000016
+/* Max count for adsp error code sent to HLOS*/
+#define CPE_ERR_MAX      (CPE_ETERMINATED + 1)
+
+
+/* ERROR STRING */
+/* Success. The operation completed with no errors. */
+#define CPE_EOK_STR          "CPE_EOK"
+/* General failure. */
+#define CPE_EFAILED_STR      "CPE_EFAILED"
+/* Bad operation parameter. */
+#define CPE_EBADPARAM_STR    "CPE_EBADPARAM"
+/* Unsupported routine or operation. */
+#define CPE_EUNSUPPORTED_STR "CPE_EUNSUPPORTED"
+/* Unsupported version. */
+#define CPE_EVERSION_STR     "CPE_EVERSION"
+/* Unexpected problem encountered. */
+#define CPE_EUNEXPECTED_STR  "CPE_EUNEXPECTED"
+/* Unhandled problem occurred. */
+#define CPE_EPANIC_STR       "CPE_EPANIC"
+/* Unable to allocate resource. */
+#define CPE_ENORESOURCE_STR  "CPE_ENORESOURCE"
+/* Invalid handle. */
+#define CPE_EHANDLE_STR      "CPE_EHANDLE"
+/* Operation is already processed. */
+#define CPE_EALREADY_STR     "CPE_EALREADY"
+/* Operation is not ready to be processed. */
+#define CPE_ENOTREADY_STR    "CPE_ENOTREADY"
+/* Operation is pending completion. */
+#define CPE_EPENDING_STR     "CPE_EPENDING"
+/* Operation could not be accepted or processed. */
+#define CPE_EBUSY_STR        "CPE_EBUSY"
+/* Operation aborted due to an error. */
+#define CPE_EABORTED_STR     "CPE_EABORTED"
+/* Operation preempted by a higher priority. */
+#define CPE_EPREEMPTED_STR   "CPE_EPREEMPTED"
+/* Operation requests intervention to complete. */
+#define CPE_ECONTINUE_STR    "CPE_ECONTINUE"
+/* Operation requests immediate intervention to complete. */
+#define CPE_EIMMEDIATE_STR   "CPE_EIMMEDIATE"
+/* Operation is not implemented. */
+#define CPE_ENOTIMPL_STR     "CPE_ENOTIMPL"
+/* Operation needs more data or resources. */
+#define CPE_ENEEDMORE_STR    "CPE_ENEEDMORE"
+/* Operation does not have memory. */
+#define CPE_ENOMEMORY_STR    "CPE_ENOMEMORY"
+/* Item does not exist. */
+#define CPE_ENOTEXIST_STR    "CPE_ENOTEXIST"
+/* Operation is finished. */
+#define CPE_ETERMINATED_STR  "CPE_ETERMINATED"
+/* Unexpected error code. */
+#define CPE_ERR_MAX_STR      "CPE_ERR_MAX"
+
+
+struct cpe_err_code {
+	int     lnx_err_code;
+	char    *cpe_err_str;
+};
+
+
+static struct cpe_err_code cpe_err_code_info[CPE_ERR_MAX+1] = {
+	{ 0, CPE_EOK_STR},
+	{ -ENOTRECOVERABLE, CPE_EFAILED_STR},
+	{ -EINVAL, CPE_EBADPARAM_STR},
+	{ -ENOSYS, CPE_EUNSUPPORTED_STR},
+	{ -ENOPROTOOPT, CPE_EVERSION_STR},
+	{ -ENOTRECOVERABLE, CPE_EUNEXPECTED_STR},
+	{ -ENOTRECOVERABLE, CPE_EPANIC_STR},
+	{ -ENOSPC, CPE_ENORESOURCE_STR},
+	{ -EBADR, CPE_EHANDLE_STR},
+	{ -EALREADY, CPE_EALREADY_STR},
+	{ -EPERM, CPE_ENOTREADY_STR},
+	{ -EINPROGRESS, CPE_EPENDING_STR},
+	{ -EBUSY, CPE_EBUSY_STR},
+	{ -ECANCELED, CPE_EABORTED_STR},
+	{ -EAGAIN, CPE_EPREEMPTED_STR},
+	{ -EAGAIN, CPE_ECONTINUE_STR},
+	{ -EAGAIN, CPE_EIMMEDIATE_STR},
+	{ -EAGAIN, CPE_ENOTIMPL_STR},
+	{ -ENODATA, CPE_ENEEDMORE_STR},
+	{ -EADV, CPE_ERR_MAX_STR},
+	{ -ENOMEM, CPE_ENOMEMORY_STR},
+	{ -ENODEV, CPE_ENOTEXIST_STR},
+	{ -EADV, CPE_ETERMINATED_STR},
+	{ -EADV, CPE_ERR_MAX_STR},
+};
+
+static inline int cpe_err_get_lnx_err_code(u32 cpe_error)
+{
+	if (cpe_error > CPE_ERR_MAX)
+		return cpe_err_code_info[CPE_ERR_MAX].lnx_err_code;
+	else
+		return cpe_err_code_info[cpe_error].lnx_err_code;
+}
+
+static inline char *cpe_err_get_err_str(u32 cpe_error)
+{
+	if (cpe_error > CPE_ERR_MAX)
+		return cpe_err_code_info[CPE_ERR_MAX].cpe_err_str;
+	else
+		return cpe_err_code_info[cpe_error].cpe_err_str;
+}
+
+#endif
diff --git a/include/sound/msm-audio-effects-q6-v2.h b/include/sound/msm-audio-effects-q6-v2.h
new file mode 100644
index 000000000000..cbdea328d46c
--- /dev/null
+++ b/include/sound/msm-audio-effects-q6-v2.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef _MSM_AUDIO_EFFECTS_H
+#define _MSM_AUDIO_EFFECTS_H
+
+#include <sound/audio_effects.h>
+
+bool msm_audio_effects_is_effmodule_supp_in_top(int effect_module,
+						int topology);
+
+int msm_audio_effects_enable_extn(struct audio_client *ac,
+			     struct msm_nt_eff_all_config *effects,
+			     bool flag);
+
+int msm_audio_effects_reverb_handler(struct audio_client *ac,
+				     struct reverb_params *reverb,
+				     long *values);
+
+int msm_audio_effects_bass_boost_handler(struct audio_client *ac,
+					struct bass_boost_params *bass_boost,
+					long *values);
+
+int msm_audio_effects_pbe_handler(struct audio_client *ac,
+					struct pbe_params *pbe,
+					long *values);
+
+int msm_audio_effects_virtualizer_handler(struct audio_client *ac,
+				struct virtualizer_params *virtualizer,
+				long *values);
+
+int msm_audio_effects_popless_eq_handler(struct audio_client *ac,
+					 struct eq_params *eq,
+					 long *values);
+
+int msm_audio_effects_volume_handler(struct audio_client *ac,
+				     struct soft_volume_params *vol,
+				     long *values);
+
+int msm_audio_effects_volume_handler_v2(struct audio_client *ac,
+					struct soft_volume_params *vol,
+					long *values, int instance);
+#endif /*_MSM_AUDIO_EFFECTS_H*/
diff --git a/include/sound/msm-dai-q6-v2.h b/include/sound/msm-dai-q6-v2.h
new file mode 100644
index 000000000000..810339fc0704
--- /dev/null
+++ b/include/sound/msm-dai-q6-v2.h
@@ -0,0 +1,92 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __MSM_DAI_Q6_PDATA_H__
+
+#define __MSM_DAI_Q6_PDATA_H__
+
+#define MSM_MI2S_SD0 (1 << 0)
+#define MSM_MI2S_SD1 (1 << 1)
+#define MSM_MI2S_SD2 (1 << 2)
+#define MSM_MI2S_SD3 (1 << 3)
+#define MSM_MI2S_CAP_RX 0
+#define MSM_MI2S_CAP_TX 1
+
+#define MSM_PRIM_MI2S 0
+#define MSM_SEC_MI2S  1
+#define MSM_TERT_MI2S 2
+#define MSM_QUAT_MI2S  3
+#define MSM_SEC_MI2S_SD1  4
+#define MSM_QUIN_MI2S  5
+#define MSM_SENARY_MI2S  6
+#define MSM_MI2S_MIN MSM_PRIM_MI2S
+#define MSM_MI2S_MAX MSM_SENARY_MI2S
+
+struct msm_dai_auxpcm_config {
+	u16 mode;
+	u16 sync;
+	u16 frame;
+	u16 quant;
+	u16 num_slots;
+	u16 *slot_mapping;
+	u16 data;
+	u32 pcm_clk_rate;
+};
+
+struct msm_dai_auxpcm_pdata {
+	struct msm_dai_auxpcm_config mode_8k;
+	struct msm_dai_auxpcm_config mode_16k;
+};
+
+struct msm_mi2s_pdata {
+	u16 rx_sd_lines;
+	u16 tx_sd_lines;
+	u16 intf_id;
+};
+
+struct msm_i2s_data {
+	u32 capability; /* RX or TX */
+	u16 sd_lines;
+};
+
+struct msm_dai_tdm_group_config {
+	u16 group_id;
+	u16 num_ports;
+	u16 *port_id;
+	u16 nslots_per_frame;
+	u16 slot_width;
+	u32 slot_mask;
+	u32 clk_rate;
+};
+
+struct msm_dai_tdm_config {
+	u16 sync_mode;
+	u16 sync_src;
+	u16 data_out;
+	u16 invert_sync;
+	u16 data_delay;
+	u32 offset_data_align;
+	u16 num_offset;
+	u16 *offset;
+	u16  header_start_offset;
+	u16  header_width;
+	u16  header_num_frame_repeat;
+};
+
+#define MSM_DAI_TDM_MAX_CH	8
+
+struct msm_dai_tdm_pdata {
+	struct msm_dai_tdm_group_config group_config;
+	struct msm_dai_tdm_config config[MSM_DAI_TDM_MAX_CH];
+};
+
+#endif
diff --git a/include/sound/msm-dts-eagle.h b/include/sound/msm-dts-eagle.h
new file mode 100644
index 000000000000..2ef01136b7ce
--- /dev/null
+++ b/include/sound/msm-dts-eagle.h
@@ -0,0 +1,148 @@
+/* Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __MSM_DTS_EAGLE_H__
+#define __MSM_DTS_EAGLE_H__
+
+#include <linux/compat.h>
+#include <sound/soc.h>
+#include <sound/devdep_params.h>
+#include <sound/q6asm-v2.h>
+
+#ifdef CONFIG_COMPAT
+enum {
+	DTS_EAGLE_IOCTL_GET_CACHE_SIZE32 = _IOR(0xF2, 0, __s32),
+	DTS_EAGLE_IOCTL_SET_CACHE_SIZE32 = _IOW(0xF2, 1, __s32),
+	DTS_EAGLE_IOCTL_GET_PARAM32 = _IOR(0xF2, 2, compat_uptr_t),
+	DTS_EAGLE_IOCTL_SET_PARAM32 = _IOW(0xF2, 3, compat_uptr_t),
+	DTS_EAGLE_IOCTL_SET_CACHE_BLOCK32 =
+				_IOW(0xF2, 4, compat_uptr_t),
+	DTS_EAGLE_IOCTL_SET_ACTIVE_DEVICE32 =
+				_IOW(0xF2, 5, compat_uptr_t),
+	DTS_EAGLE_IOCTL_GET_LICENSE32 =
+				_IOR(0xF2, 6, compat_uptr_t),
+	DTS_EAGLE_IOCTL_SET_LICENSE32 =
+				 _IOW(0xF2, 7, compat_uptr_t),
+	DTS_EAGLE_IOCTL_SEND_LICENSE32 = _IOW(0xF2, 8, __s32),
+	DTS_EAGLE_IOCTL_SET_VOLUME_COMMANDS32 = _IOW(0xF2, 9,
+						     compat_uptr_t),
+};
+#endif
+
+#ifdef CONFIG_DTS_EAGLE
+void msm_dts_ion_memmap(struct param_outband *po_);
+int msm_dts_eagle_enable_asm(struct audio_client *ac, u32 enable, int module);
+int msm_dts_eagle_enable_adm(int port_id, int copp_idx, u32 enable);
+void msm_dts_eagle_add_controls(struct snd_soc_platform *platform);
+int msm_dts_eagle_set_stream_gain(struct audio_client *ac,
+				  int lgain, int rgain);
+int msm_dts_eagle_handle_asm(struct dts_eagle_param_desc *depd, char *buf,
+			     bool for_pre, bool get, struct audio_client *ac,
+			     struct param_outband *po);
+int msm_dts_eagle_handle_adm(struct dts_eagle_param_desc *depd, char *buf,
+			     bool for_pre, bool get);
+int msm_dts_eagle_ioctl(unsigned int cmd, unsigned long arg);
+int msm_dts_eagle_is_hpx_on(void);
+int msm_dts_eagle_init_pre(struct audio_client *ac);
+int msm_dts_eagle_deinit_pre(struct audio_client *ac);
+int msm_dts_eagle_init_post(int port_id, int copp_id);
+int msm_dts_eagle_deinit_post(int port_id, int topology);
+int msm_dts_eagle_init_master_module(struct audio_client *ac);
+int msm_dts_eagle_deinit_master_module(struct audio_client *ac);
+int msm_dts_eagle_pcm_new(struct snd_soc_pcm_runtime *runtime);
+void msm_dts_eagle_pcm_free(struct snd_pcm *pcm);
+int msm_dts_eagle_compat_ioctl(unsigned int cmd, unsigned long arg);
+#else
+static inline void msm_dts_ion_memmap(struct param_outband *po_)
+{
+	pr_debug("%s\n", __func__);
+}
+static inline int msm_dts_eagle_enable_asm(struct audio_client *ac,
+					   u32 enable, int module)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_enable_adm(int port_id, int copp_idx,
+					   u32 enable)
+{
+	return 0;
+}
+static inline void msm_dts_eagle_add_controls(struct snd_soc_platform *platform)
+{
+}
+static inline int msm_dts_eagle_set_stream_gain(struct audio_client *ac,
+						int lgain, int rgain)
+{
+	pr_debug("%s\n", __func__);
+	return 0;
+}
+static inline int msm_dts_eagle_handle_asm(struct dts_eagle_param_desc *depd,
+					   char *buf, bool for_pre, bool get,
+					   struct audio_client *ac,
+					   struct param_outband *po)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_handle_adm(struct dts_eagle_param_desc *depd,
+					   char *buf, bool for_pre, bool get)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_ioctl(unsigned int cmd, unsigned long arg)
+{
+	return -EPERM;
+}
+static inline int msm_dts_eagle_is_hpx_on(void)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_init_pre(struct audio_client *ac)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_deinit_pre(struct audio_client *ac)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_init_post(int port_id, int coppid)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_deinit_post(int port_id, int topology)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_init_master_module(struct audio_client *ac)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_deinit_master_module(struct audio_client *ac)
+{
+	return 0;
+}
+static inline int msm_dts_eagle_pcm_new(struct snd_soc_pcm_runtime *runtime)
+{
+	pr_debug("%s\n", __func__);
+	return 0;
+}
+static inline void msm_dts_eagle_pcm_free(struct snd_pcm *pcm)
+{
+	pr_debug("%s\n", __func__);
+}
+static inline int msm_dts_eagle_compat_ioctl(unsigned int cmd,
+					unsigned long arg)
+{
+	return 0;
+}
+#endif
+
+#endif
diff --git a/include/sound/msm-slim-dma.h b/include/sound/msm-slim-dma.h
new file mode 100644
index 000000000000..6bbdbe563edc
--- /dev/null
+++ b/include/sound/msm-slim-dma.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ */
+#ifndef _MSM_SLIMBUS_DMA_H
+#define _MSM_SLIMBUS_DMA_H
+
+#include <linux/slimbus/slimbus.h>
+
+/*
+ * struct msm_slim_dma_data - DMA data for slimbus data transfer
+ *
+ * @sdev: Handle to the slim_device instance associated with the
+ *	  data transfer.
+ * @ph:	Port handle for the slimbus ports.
+ * @dai_channel_ctl: callback function into the CPU dai driver
+ *		     to setup the data path.
+ *
+ * This structure is used to share the slimbus port handles and
+ * other data path setup related handles with other drivers.
+ */
+struct msm_slim_dma_data {
+
+	/* Handle to slimbus device */
+	struct slim_device *sdev;
+
+	/* Port Handle */
+	u32 ph;
+
+	/* Callback for data channel control */
+	int (*dai_channel_ctl) (struct msm_slim_dma_data *dma_data,
+				struct snd_soc_dai *dai, bool enable);
+};
+
+#endif
diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h
new file mode 100644
index 000000000000..1f4c6ab97f05
--- /dev/null
+++ b/include/sound/q6adm-v2.h
@@ -0,0 +1,154 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+#ifndef __Q6_ADM_V2_H__
+#define __Q6_ADM_V2_H__
+
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define ADM_PATH_NONLIVE_REC 0x3
+#define ADM_PATH_COMPRESSED_RX 0x5
+#include <linux/qdsp6v2/rtac.h>
+#include <sound/q6afe-v2.h>
+#include <sound/q6audio-v2.h>
+
+#define MAX_MODULES_IN_TOPO 16
+#define ADM_GET_TOPO_MODULE_LIST_LENGTH\
+		((MAX_MODULES_IN_TOPO + 1) * sizeof(uint32_t))
+#define AUD_PROC_BLOCK_SIZE	4096
+#define AUD_VOL_BLOCK_SIZE	4096
+#define AUDIO_RX_CALIBRATION_SIZE	(AUD_PROC_BLOCK_SIZE + \
+						AUD_VOL_BLOCK_SIZE)
+enum {
+	ADM_CUSTOM_TOP_CAL = 0,
+	ADM_AUDPROC_CAL,
+	ADM_AUDVOL_CAL,
+	ADM_RTAC_INFO_CAL,
+	ADM_RTAC_APR_CAL,
+	ADM_DTS_EAGLE,
+	ADM_SRS_TRUMEDIA,
+	ADM_RTAC_AUDVOL_CAL,
+	ADM_MAX_CAL_TYPES
+};
+
+enum {
+	ADM_MEM_MAP_INDEX_SOURCE_TRACKING = ADM_MAX_CAL_TYPES,
+	ADM_MEM_MAP_INDEX_MAX
+};
+
+enum {
+	ADM_CLIENT_ID_DEFAULT = 0,
+	ADM_CLIENT_ID_SOURCE_TRACKING,
+	ADM_CLIENT_ID_MAX,
+};
+
+#define MAX_COPPS_PER_PORT 0x8
+#define ADM_MAX_CHANNELS 8
+
+/* multiple copp per stream. */
+struct route_payload {
+	unsigned int copp_idx[MAX_COPPS_PER_PORT];
+	unsigned int port_id[MAX_COPPS_PER_PORT];
+	int app_type;
+	int acdb_dev_id;
+	int sample_rate;
+	unsigned short num_copps;
+	unsigned int session_id;
+};
+
+int srs_trumedia_open(int port_id, int copp_idx, __s32 srs_tech_id,
+		      void *srs_params);
+
+int adm_dts_eagle_set(int port_id, int copp_idx, int param_id,
+		      void *data, uint32_t size);
+
+int adm_dts_eagle_get(int port_id, int copp_idx, int param_id,
+		      void *data, uint32_t size);
+
+int adm_get_params(int port_id, int copp_idx, uint32_t module_id,
+		   uint32_t param_id, uint32_t params_length, char *params);
+
+int adm_send_params_v5(int port_id, int copp_idx, char *params,
+			      uint32_t params_length);
+
+int adm_dolby_dap_send_params(int port_id, int copp_idx, char *params,
+			      uint32_t params_length);
+
+int adm_open(int port, int path, int rate, int mode, int topology,
+			   int perf_mode, uint16_t bits_per_sample,
+			   int app_type, int acdbdev_id);
+
+int adm_map_rtac_block(struct rtac_cal_block_data *cal_block);
+
+int adm_unmap_rtac_block(uint32_t *mem_map_handle);
+
+int adm_close(int port, int topology, int perf_mode);
+
+int adm_matrix_map(int path, struct route_payload payload_map,
+		   int perf_mode);
+
+int adm_connect_afe_port(int mode, int session_id, int port_id);
+
+void adm_ec_ref_rx_id(int  port_id);
+
+int adm_get_lowlatency_copp_id(int port_id);
+
+int adm_set_multi_ch_map(char *channel_map, int path);
+
+int adm_get_multi_ch_map(char *channel_map, int path);
+
+int adm_validate_and_get_port_index(int port_id);
+
+int adm_get_default_copp_idx(int port_id);
+
+int adm_get_topology_for_port_from_copp_id(int port_id, int copp_id);
+
+int adm_get_topology_for_port_copp_idx(int port_id, int copp_idx);
+
+int adm_get_indexes_from_copp_id(int copp_id, int *port_idx, int *copp_idx);
+
+int adm_set_stereo_to_custom_stereo(int port_id, int copp_idx,
+				    unsigned int session_id,
+				    char *params, uint32_t params_length);
+
+int adm_get_pp_topo_module_list(int port_id, int copp_idx, int32_t param_length,
+				char *params);
+
+int adm_set_volume(int port_id, int copp_idx, int volume);
+
+int adm_set_softvolume(int port_id, int copp_idx,
+		       struct audproc_softvolume_params *softvol_param);
+
+int adm_param_enable(int port_id, int copp_idx, int module_id,  int enable);
+
+int adm_send_calibration(int port_id, int copp_idx, int path, int perf_mode,
+			 int cal_type, char *params, int size);
+
+int adm_set_wait_parameters(int port_id, int copp_idx);
+
+int adm_reset_wait_parameters(int port_id, int copp_idx);
+
+int adm_wait_timeout(int port_id, int copp_idx, int wait_time);
+
+int adm_store_cal_data(int port_id, int copp_idx, int path, int perf_mode,
+		       int cal_type, char *params, int *size);
+
+int adm_send_compressed_device_mute(int port_id, int copp_idx, bool mute_on);
+
+int adm_send_compressed_device_latency(int port_id, int copp_idx, int latency);
+int adm_set_sound_focus(int port_id, int copp_idx,
+			struct sound_focus_param soundFocusData);
+int adm_get_sound_focus(int port_id, int copp_idx,
+			struct sound_focus_param *soundFocusData);
+int adm_get_source_tracking(int port_id, int copp_idx,
+			    struct source_tracking_param *sourceTrackingData);
+#endif /* __Q6_ADM_V2_H__ */
diff --git a/include/sound/q6afe-v2.h b/include/sound/q6afe-v2.h
new file mode 100644
index 000000000000..c76709b55e6a
--- /dev/null
+++ b/include/sound/q6afe-v2.h
@@ -0,0 +1,330 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+#ifndef __Q6AFE_V2_H__
+#define __Q6AFE_V2_H__
+#include <sound/apr_audio-v2.h>
+#include <linux/qdsp6v2/rtac.h>
+
+#define IN			0x000
+#define OUT			0x001
+#define MSM_AFE_MONO        0
+#define MSM_AFE_CH_STEREO   1
+#define MSM_AFE_MONO_RIGHT  1
+#define MSM_AFE_MONO_LEFT   2
+#define MSM_AFE_STEREO      3
+#define MSM_AFE_4CHANNELS   4
+#define MSM_AFE_6CHANNELS   6
+#define MSM_AFE_8CHANNELS   8
+
+#define MSM_AFE_I2S_FORMAT_LPCM		0
+#define MSM_AFE_I2S_FORMAT_COMPR		1
+#define MSM_AFE_I2S_FORMAT_IEC60958_LPCM	2
+#define MSM_AFE_I2S_FORMAT_IEC60958_COMPR	3
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+
+#define RT_PROXY_DAI_001_RX	0xE0
+#define RT_PROXY_DAI_001_TX	0xF0
+#define RT_PROXY_DAI_002_RX	0xF1
+#define RT_PROXY_DAI_002_TX	0xE1
+#define VIRTUAL_ID_TO_PORTID(val) ((val & 0xF) | 0x2000)
+
+enum {
+	/* IDX 0->4 */
+	IDX_PRIMARY_I2S_RX,
+	IDX_PRIMARY_I2S_TX,
+	IDX_AFE_PORT_ID_PRIMARY_PCM_RX,
+	IDX_AFE_PORT_ID_PRIMARY_PCM_TX,
+	IDX_SECONDARY_I2S_RX,
+	/* IDX 5->9 */
+	IDX_SECONDARY_I2S_TX,
+	IDX_MI2S_RX,
+	IDX_MI2S_TX,
+	IDX_HDMI_RX,
+	IDX_RSVD_2,
+	/* IDX 10->14 */
+	IDX_RSVD_3,
+	IDX_DIGI_MIC_TX,
+	IDX_VOICE_RECORD_RX,
+	IDX_VOICE_RECORD_TX,
+	IDX_VOICE_PLAYBACK_TX,
+	/* IDX 15->19 */
+	IDX_SLIMBUS_0_RX,
+	IDX_SLIMBUS_0_TX,
+	IDX_SLIMBUS_1_RX,
+	IDX_SLIMBUS_1_TX,
+	IDX_SLIMBUS_2_RX,
+	/* IDX 20->24 */
+	IDX_SLIMBUS_2_TX,
+	IDX_SLIMBUS_3_RX,
+	IDX_SLIMBUS_3_TX,
+	IDX_SLIMBUS_4_RX,
+	IDX_SLIMBUS_4_TX,
+	/* IDX 25->29 */
+	IDX_SLIMBUS_5_RX,
+	IDX_SLIMBUS_5_TX,
+	IDX_INT_BT_SCO_RX,
+	IDX_INT_BT_SCO_TX,
+	IDX_INT_BT_A2DP_RX,
+	/* IDX 30->34 */
+	IDX_INT_FM_RX,
+	IDX_INT_FM_TX,
+	IDX_RT_PROXY_PORT_001_RX,
+	IDX_RT_PROXY_PORT_001_TX,
+	IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX,
+	/* IDX 35->39 */
+	IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX,
+	IDX_AFE_PORT_ID_SECONDARY_MI2S_RX,
+	IDX_AFE_PORT_ID_SECONDARY_MI2S_TX,
+	IDX_AFE_PORT_ID_TERTIARY_MI2S_RX,
+	IDX_AFE_PORT_ID_TERTIARY_MI2S_TX,
+	/* IDX 40->44 */
+	IDX_AFE_PORT_ID_PRIMARY_MI2S_RX,
+	IDX_AFE_PORT_ID_PRIMARY_MI2S_TX,
+	IDX_AFE_PORT_ID_SECONDARY_PCM_RX,
+	IDX_AFE_PORT_ID_SECONDARY_PCM_TX,
+	IDX_VOICE2_PLAYBACK_TX,
+	/* IDX 45->49 */
+	IDX_SLIMBUS_6_RX,
+	IDX_SLIMBUS_6_TX,
+	IDX_SPDIF_RX,
+	IDX_GLOBAL_CFG,
+	IDX_AUDIO_PORT_ID_I2S_RX,
+	/* IDX 50->53 */
+	IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_SD1,
+	IDX_AFE_PORT_ID_QUINARY_MI2S_RX,
+	IDX_AFE_PORT_ID_QUINARY_MI2S_TX,
+	IDX_AFE_PORT_ID_SENARY_MI2S_TX,
+	/* IDX 54-> 118 */
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_0,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_0,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_1,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_1,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_2,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_2,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_3,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_3,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_4,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_4,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_5,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_5,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_6,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_6,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_RX_7,
+	IDX_AFE_PORT_ID_PRIMARY_TDM_TX_7,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_0,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_0,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_1,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_1,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_2,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_2,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_3,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_3,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_4,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_4,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_5,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_5,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_6,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_6,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_RX_7,
+	IDX_AFE_PORT_ID_SECONDARY_TDM_TX_7,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_0,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_0,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_1,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_1,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_2,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_2,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_3,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_3,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_4,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_4,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_5,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_5,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_6,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_6,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_RX_7,
+	IDX_AFE_PORT_ID_TERTIARY_TDM_TX_7,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_0,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_0,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_1,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_1,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_2,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_2,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_3,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_3,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_4,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_4,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_5,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_5,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_6,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_6,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_7,
+	IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_7,
+	AFE_MAX_PORTS
+};
+
+enum afe_mad_type {
+	MAD_HW_NONE = 0x00,
+	MAD_HW_AUDIO = 0x01,
+	MAD_HW_BEACON = 0x02,
+	MAD_HW_ULTRASOUND = 0x04,
+	MAD_SW_AUDIO = 0x05,
+};
+
+enum afe_cal_mode {
+	AFE_CAL_MODE_DEFAULT = 0x00,
+	AFE_CAL_MODE_NONE,
+};
+
+struct afe_audio_buffer {
+	dma_addr_t phys;
+	void       *data;
+	uint32_t   used;
+	uint32_t   size;/* size of buffer */
+	uint32_t   actual_size; /* actual number of bytes read by DSP */
+	struct      ion_handle *handle;
+	struct      ion_client *client;
+};
+
+struct afe_audio_port_data {
+	struct afe_audio_buffer *buf;
+	uint32_t	    max_buf_cnt;
+	uint32_t	    dsp_buf;
+	uint32_t	    cpu_buf;
+	struct list_head    mem_map_handle;
+	uint32_t	    tmp_hdl;
+	/* read or write locks */
+	struct mutex	    lock;
+	spinlock_t	    dsp_lock;
+};
+
+struct afe_audio_client {
+	atomic_t	       cmd_state;
+	/* Relative or absolute TS */
+	uint32_t	       time_flag;
+	void		       *priv;
+	uint64_t	       time_stamp;
+	struct mutex	       cmd_lock;
+	/* idx:1 out port, 0: in port*/
+	struct afe_audio_port_data port[2];
+	wait_queue_head_t      cmd_wait;
+	uint32_t               mem_map_handle;
+};
+
+struct aanc_data {
+	bool aanc_active;
+	uint16_t aanc_rx_port;
+	uint16_t aanc_tx_port;
+	uint32_t aanc_rx_port_sample_rate;
+	uint32_t aanc_tx_port_sample_rate;
+};
+
+int afe_open(u16 port_id, union afe_port_config *afe_config, int rate);
+int afe_close(int port_id);
+int afe_loopback(u16 enable, u16 rx_port, u16 tx_port);
+int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain);
+int afe_loopback_gain(u16 port_id, u16 volume);
+int afe_validate_port(u16 port_id);
+int afe_get_port_index(u16 port_id);
+int afe_get_topology(int port_id);
+int afe_start_pseudo_port(u16 port_id);
+int afe_stop_pseudo_port(u16 port_id);
+uint32_t afe_req_mmap_handle(struct afe_audio_client *ac);
+int afe_memory_map(phys_addr_t dma_addr_p, u32 dma_buf_sz,
+		struct afe_audio_client *ac);
+int afe_cmd_memory_map(phys_addr_t dma_addr_p, u32 dma_buf_sz);
+int afe_cmd_memory_map_nowait(int port_id, phys_addr_t dma_addr_p,
+			u32 dma_buf_sz);
+int afe_cmd_memory_unmap(u32 dma_addr_p);
+int afe_cmd_memory_unmap_nowait(u32 dma_addr_p);
+void afe_set_dtmf_gen_rx_portid(u16 rx_port_id, int set);
+int afe_dtmf_generate_rx(int64_t duration_in_ms,
+			 uint16_t high_freq,
+			 uint16_t low_freq, uint16_t gain);
+int afe_register_get_events(u16 port_id,
+		void (*cb) (uint32_t opcode,
+		uint32_t token, uint32_t *payload, void *priv),
+		void *private_data);
+int afe_unregister_get_events(u16 port_id);
+int afe_rt_proxy_port_write(phys_addr_t buf_addr_p,
+			u32 mem_map_handle, int bytes);
+int afe_rt_proxy_port_read(phys_addr_t buf_addr_p,
+			u32 mem_map_handle, int bytes);
+void afe_set_cal_mode(u16 port_id, enum afe_cal_mode afe_cal_mode);
+int afe_port_start(u16 port_id, union afe_port_config *afe_config,
+	u32 rate);
+int afe_spk_prot_feed_back_cfg(int src_port, int dst_port,
+	int l_ch, int r_ch, u32 enable);
+int afe_spk_prot_get_calib_data(struct afe_spkr_prot_get_vi_calib *calib);
+int afe_port_stop_nowait(int port_id);
+int afe_apply_gain(u16 port_id, u16 gain);
+int afe_q6_interface_prepare(void);
+int afe_get_port_type(u16 port_id);
+int q6afe_audio_client_buf_alloc_contiguous(unsigned int dir,
+			struct afe_audio_client *ac,
+			unsigned int bufsz,
+			unsigned int bufcnt);
+struct afe_audio_client *q6afe_audio_client_alloc(void *priv);
+int q6afe_audio_client_buf_free_contiguous(unsigned int dir,
+			struct afe_audio_client *ac);
+void q6afe_audio_client_free(struct afe_audio_client *ac);
+/* if port_id is virtual, convert to physical..
+ * if port_id is already physical, return physical
+ */
+int afe_convert_virtual_to_portid(u16 port_id);
+
+int afe_pseudo_port_start_nowait(u16 port_id);
+int afe_pseudo_port_stop_nowait(u16 port_id);
+int afe_set_lpass_clock(u16 port_id, struct afe_clk_cfg *cfg);
+int afe_set_lpass_clock_v2(u16 port_id, struct afe_clk_set *cfg);
+int afe_set_lpass_clk_cfg(int index, struct afe_clk_set *cfg);
+int afe_set_digital_codec_core_clock(u16 port_id,
+			struct afe_digital_clk_cfg *cfg);
+int afe_set_lpass_internal_digital_codec_clock(u16 port_id,
+				struct afe_digital_clk_cfg *cfg);
+int afe_enable_lpass_core_shared_clock(u16 port_id, u32 enable);
+
+int q6afe_check_osr_clk_freq(u32 freq);
+
+int afe_send_spdif_clk_cfg(struct afe_param_id_spdif_clk_cfg *cfg,
+		u16 port_id);
+int afe_send_spdif_ch_status_cfg(struct afe_param_id_spdif_ch_status_cfg
+		*ch_status_cfg,	u16 port_id);
+
+int afe_spdif_port_start(u16 port_id, struct afe_spdif_port_config *spdif_port,
+		u32 rate);
+
+int afe_turn_onoff_hw_mad(u16 mad_type, u16 mad_enable);
+int afe_port_set_mad_type(u16 port_id, enum afe_mad_type mad_type);
+enum afe_mad_type afe_port_get_mad_type(u16 port_id);
+int afe_set_config(enum afe_config_type config_type, void *config_data,
+		   int arg);
+void afe_clear_config(enum afe_config_type config);
+bool afe_has_config(enum afe_config_type config);
+
+void afe_set_aanc_info(struct aanc_data *aanc_info);
+int afe_port_group_set_param(u16 group_id,
+	union afe_port_group_config *afe_group_config);
+int afe_port_group_enable(u16 group_id,
+	union afe_port_group_config *afe_group_config, u16 enable);
+int afe_unmap_rtac_block(uint32_t *mem_map_handle);
+int afe_map_rtac_block(struct rtac_cal_block_data *cal_block);
+int afe_send_slot_mapping_cfg(
+	struct afe_param_id_slot_mapping_cfg *slot_mapping_cfg,
+	u16 port_id);
+int afe_send_custom_tdm_header_cfg(
+	struct afe_param_id_custom_tdm_header_cfg *custom_tdm_header_cfg,
+	u16 port_id);
+int afe_tdm_port_start(u16 port_id, struct afe_tdm_port_config *tdm_port,
+		u32 rate);
+#endif /* __Q6AFE_V2_H__ */
diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h
new file mode 100644
index 000000000000..44c216907006
--- /dev/null
+++ b/include/sound/q6asm-v2.h
@@ -0,0 +1,490 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+#ifndef __Q6_ASM_V2_H__
+#define __Q6_ASM_V2_H__
+
+#include <linux/qdsp6v2/apr.h>
+#include <linux/qdsp6v2/rtac.h>
+#include <sound/apr_audio-v2.h>
+#include <linux/list.h>
+#include <linux/msm_ion.h>
+#include <linux/spinlock.h>
+
+#define IN                      0x000
+#define OUT                     0x001
+#define CH_MODE_MONO            0x001
+#define CH_MODE_STEREO          0x002
+
+#define FORMAT_LINEAR_PCM   0x0000
+#define FORMAT_DTMF         0x0001
+#define FORMAT_ADPCM	    0x0002
+#define FORMAT_YADPCM       0x0003
+#define FORMAT_MP3          0x0004
+#define FORMAT_MPEG4_AAC    0x0005
+#define FORMAT_AMRNB	    0x0006
+#define FORMAT_AMRWB	    0x0007
+#define FORMAT_V13K	    0x0008
+#define FORMAT_EVRC	    0x0009
+#define FORMAT_EVRCB	    0x000a
+#define FORMAT_EVRCWB	    0x000b
+#define FORMAT_MIDI	    0x000c
+#define FORMAT_SBC	    0x000d
+#define FORMAT_WMA_V10PRO   0x000e
+#define FORMAT_WMA_V9	    0x000f
+#define FORMAT_AMR_WB_PLUS  0x0010
+#define FORMAT_MPEG4_MULTI_AAC 0x0011
+#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012
+#define FORMAT_AC3          0x0013
+#define FORMAT_EAC3         0x0014
+#define FORMAT_MP2          0x0015
+#define FORMAT_FLAC         0x0016
+#define FORMAT_ALAC         0x0017
+#define FORMAT_VORBIS       0x0018
+#define FORMAT_APE          0x0019
+
+#define ENCDEC_SBCBITRATE   0x0001
+#define ENCDEC_IMMEDIATE_DECODE 0x0002
+#define ENCDEC_CFG_BLK          0x0003
+
+#define CMD_PAUSE          0x0001
+#define CMD_FLUSH          0x0002
+#define CMD_EOS            0x0003
+#define CMD_CLOSE          0x0004
+#define CMD_OUT_FLUSH      0x0005
+#define CMD_SUSPEND        0x0006
+
+/* bit 0:1 represents priority of stream */
+#define STREAM_PRIORITY_NORMAL	0x0000
+#define STREAM_PRIORITY_LOW	0x0001
+#define STREAM_PRIORITY_HIGH	0x0002
+
+/* bit 4 represents META enable of encoded data buffer */
+#define BUFFER_META_ENABLE	0x0010
+
+/* Enable Sample_Rate/Channel_Mode notification event from Decoder */
+#define SR_CM_NOTIFY_ENABLE	0x0004
+
+#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+#define TUN_READ_IO_MODE  0x0004 /* tunnel read write mode */
+#define SYNC_IO_MODE	0x0001
+#define ASYNC_IO_MODE	0x0002
+#define COMPRESSED_IO	0x0040
+#define COMPRESSED_STREAM_IO	0x0080
+#define NT_MODE        0x0400
+
+#define NO_TIMESTAMP    0xFF00
+#define SET_TIMESTAMP   0x0000
+
+#define SOFT_PAUSE_ENABLE	1
+#define SOFT_PAUSE_DISABLE	0
+
+#define SESSION_MAX		0x08
+#define ASM_CONTROL_SESSION	0x0F
+
+#define ASM_SHIFT_GAPLESS_MODE_FLAG	31
+#define ASM_SHIFT_LAST_BUFFER_FLAG	30
+
+/* payload structure bytes */
+#define READDONE_IDX_STATUS 0
+#define READDONE_IDX_BUFADD_LSW 1
+#define READDONE_IDX_BUFADD_MSW 2
+#define READDONE_IDX_MEMMAP_HDL 3
+#define READDONE_IDX_SIZE 4
+#define READDONE_IDX_OFFSET 5
+#define READDONE_IDX_LSW_TS 6
+#define READDONE_IDX_MSW_TS 7
+#define READDONE_IDX_FLAGS 8
+#define READDONE_IDX_NUMFRAMES 9
+#define READDONE_IDX_SEQ_ID 10
+
+#define SOFT_PAUSE_PERIOD       30   /* ramp up/down for 30ms    */
+#define SOFT_PAUSE_STEP         0 /* Step value 0ms or 0us */
+enum {
+	SOFT_PAUSE_CURVE_LINEAR = 0,
+	SOFT_PAUSE_CURVE_EXP,
+	SOFT_PAUSE_CURVE_LOG,
+};
+
+#define SOFT_VOLUME_PERIOD       30   /* ramp up/down for 30ms    */
+#define SOFT_VOLUME_STEP         0 /* Step value 0ms or 0us */
+enum {
+	SOFT_VOLUME_CURVE_LINEAR = 0,
+	SOFT_VOLUME_CURVE_EXP,
+	SOFT_VOLUME_CURVE_LOG,
+};
+
+#define SOFT_VOLUME_INSTANCE_1	1
+#define SOFT_VOLUME_INSTANCE_2	2
+
+typedef void (*app_cb)(uint32_t opcode, uint32_t token,
+			uint32_t *payload, void *priv);
+
+struct audio_buffer {
+	dma_addr_t phys;
+	void       *data;
+	uint32_t   used;
+	uint32_t   size;/* size of buffer */
+	uint32_t   actual_size; /* actual number of bytes read by DSP */
+	struct      ion_handle *handle;
+	struct      ion_client *client;
+};
+
+struct audio_aio_write_param {
+	phys_addr_t   paddr;
+	uint32_t      len;
+	uint32_t      uid;
+	uint32_t      lsw_ts;
+	uint32_t      msw_ts;
+	uint32_t      flags;
+	uint32_t      metadata_len;
+	uint32_t      last_buffer;
+};
+
+struct audio_aio_read_param {
+	phys_addr_t   paddr;
+	uint32_t      len;
+	uint32_t      uid;
+};
+
+struct audio_port_data {
+	struct audio_buffer *buf;
+	uint32_t	    max_buf_cnt;
+	uint32_t	    dsp_buf;
+	uint32_t	    cpu_buf;
+	struct list_head    mem_map_handle;
+	uint32_t	    tmp_hdl;
+	/* read or write locks */
+	struct mutex	    lock;
+	spinlock_t	    dsp_lock;
+};
+
+struct audio_client {
+	int                    session;
+	app_cb		       cb;
+	atomic_t	       cmd_state;
+	/* Relative or absolute TS */
+	atomic_t	       time_flag;
+	atomic_t	       nowait_cmd_cnt;
+	struct list_head       no_wait_que;
+	spinlock_t             no_wait_que_spinlock;
+	atomic_t               mem_state;
+	void		       *priv;
+	uint32_t               io_mode;
+	uint64_t	       time_stamp;
+	struct apr_svc         *apr;
+	struct apr_svc         *mmap_apr;
+	struct apr_svc         *apr2;
+	struct mutex	       cmd_lock;
+	/* idx:1 out port, 0: in port*/
+	struct audio_port_data port[2];
+	wait_queue_head_t      cmd_wait;
+	wait_queue_head_t      time_wait;
+	wait_queue_head_t      mem_wait;
+	int                    perf_mode;
+	int					   stream_id;
+	struct device *dev;
+	int		       topology;
+	int		       app_type;
+	/* audio cache operations fptr*/
+	int (*fptr_cache_ops)(struct audio_buffer *abuff, int cache_op);
+	atomic_t               unmap_cb_success;
+	atomic_t               reset;
+	/* holds latest DSP pipeline delay */
+	uint32_t               path_delay;
+};
+
+void q6asm_audio_client_free(struct audio_client *ac);
+
+struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv);
+
+struct audio_client *q6asm_get_audio_client(int session_id);
+
+int q6asm_audio_client_buf_alloc(unsigned int dir/* 1:Out,0:In */,
+				struct audio_client *ac,
+				unsigned int bufsz,
+				unsigned int bufcnt);
+int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir
+				/* 1:Out,0:In */,
+				struct audio_client *ac,
+				unsigned int bufsz,
+				unsigned int bufcnt);
+
+int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
+			struct audio_client *ac);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format
+		/*, uint16_t bits_per_sample*/);
+
+int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,
+			uint16_t bits_per_sample);
+
+int q6asm_open_write(struct audio_client *ac, uint32_t format
+		/*, uint16_t bits_per_sample*/);
+
+int q6asm_open_write_v2(struct audio_client *ac, uint32_t format,
+			uint16_t bits_per_sample);
+
+int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
+				uint16_t bits_per_sample, int32_t stream_id,
+				bool is_gapless_mode);
+
+int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format,
+				uint32_t passthrough_flag);
+
+int q6asm_open_read_write(struct audio_client *ac,
+			uint32_t rd_format,
+			uint32_t wr_format);
+
+int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format,
+			     uint32_t wr_format, bool is_meta_data_mode,
+			     uint32_t bits_per_sample, bool overwrite_topology,
+			     int topology);
+
+int q6asm_open_loopback_v2(struct audio_client *ac,
+			   uint16_t bits_per_sample);
+
+int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+				uint32_t lsw_ts, uint32_t flags);
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+				uint32_t lsw_ts, uint32_t flags);
+
+int q6asm_async_write(struct audio_client *ac,
+					  struct audio_aio_write_param *param);
+
+int q6asm_async_read(struct audio_client *ac,
+					  struct audio_aio_read_param *param);
+
+int q6asm_read(struct audio_client *ac);
+int q6asm_read_v2(struct audio_client *ac, uint32_t len);
+int q6asm_read_nolock(struct audio_client *ac);
+
+int q6asm_memory_map(struct audio_client *ac, phys_addr_t buf_add,
+			int dir, uint32_t bufsz, uint32_t bufcnt);
+
+int q6asm_memory_unmap(struct audio_client *ac, phys_addr_t buf_add,
+							int dir);
+
+int q6asm_map_rtac_block(struct rtac_cal_block_data *cal_block);
+
+int q6asm_unmap_rtac_block(uint32_t *mem_map_handle);
+
+int q6asm_send_cal(struct audio_client *ac);
+
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+		uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+		uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_stream_run_nowait(struct audio_client *ac, uint32_t flags,
+		uint32_t msw_ts, uint32_t lsw_ts, uint32_t stream_id);
+
+int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable);
+
+int q6asm_reg_rx_underflow(struct audio_client *ac, uint16_t enable);
+
+int q6asm_cmd(struct audio_client *ac, int cmd);
+
+int q6asm_stream_cmd(struct audio_client *ac, int cmd, uint32_t stream_id);
+
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
+
+int q6asm_stream_cmd_nowait(struct audio_client *ac, int cmd,
+			    uint32_t stream_id);
+
+void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac,
+				uint32_t *size, uint32_t *idx);
+
+void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
+					uint32_t *size, uint32_t *idx);
+
+int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac);
+
+/* File format specific configurations to be added below */
+
+int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
+			 uint32_t frames_per_buf,
+			uint32_t sample_rate, uint32_t channels,
+			 uint32_t bit_rate,
+			 uint32_t mode, uint32_t format);
+
+int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
+			uint32_t rate, uint32_t channels);
+
+int q6asm_enc_cfg_blk_pcm_v2(struct audio_client *ac,
+			uint32_t rate, uint32_t channels,
+			uint16_t bits_per_sample,
+			bool use_default_chmap, bool use_back_flavor,
+			u8 *channel_map);
+
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+			uint32_t rate, uint32_t channels,
+			uint16_t bits_per_sample);
+
+int q6asm_set_encdec_chan_map(struct audio_client *ac,
+		uint32_t num_channels);
+
+int q6asm_enc_cfg_blk_pcm_native(struct audio_client *ac,
+			uint32_t rate, uint32_t channels);
+
+int q6asm_enable_sbrps(struct audio_client *ac,
+			uint32_t sbr_ps);
+
+int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
+			uint16_t sce_left, uint16_t sce_right);
+
+int q6asm_cfg_aac_sel_mix_coef(struct audio_client *ac, uint32_t mix_coeff);
+
+int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
+		uint16_t min_rate, uint16_t max_rate,
+		uint16_t reduced_rate_level, uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
+		uint16_t min_rate, uint16_t max_rate,
+		uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
+		uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
+		uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_media_format_block_pcm(struct audio_client *ac,
+			uint32_t rate, uint32_t channels);
+
+int q6asm_media_format_block_pcm_format_support(struct audio_client *ac,
+			uint32_t rate, uint32_t channels,
+			uint16_t bits_per_sample);
+
+int q6asm_media_format_block_pcm_format_support_v2(struct audio_client *ac,
+				uint32_t rate, uint32_t channels,
+				uint16_t bits_per_sample, int stream_id,
+				bool use_default_chmap, char *channel_map);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+			uint32_t rate, uint32_t channels,
+			bool use_default_chmap, char *channel_map);
+
+int q6asm_media_format_block_multi_ch_pcm_v2(
+			struct audio_client *ac,
+			uint32_t rate, uint32_t channels,
+			bool use_default_chmap, char *channel_map,
+			uint16_t bits_per_sample);
+
+int q6asm_media_format_block_aac(struct audio_client *ac,
+			struct asm_aac_cfg *cfg);
+
+int q6asm_stream_media_format_block_aac(struct audio_client *ac,
+			struct asm_aac_cfg *cfg, int stream_id);
+
+int q6asm_media_format_block_multi_aac(struct audio_client *ac,
+			struct asm_aac_cfg *cfg);
+
+int q6asm_media_format_block_wma(struct audio_client *ac,
+			void *cfg, int stream_id);
+
+int q6asm_media_format_block_wmapro(struct audio_client *ac,
+			void *cfg, int stream_id);
+
+int q6asm_media_format_block_amrwbplus(struct audio_client *ac,
+			struct asm_amrwbplus_cfg *cfg);
+
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+			struct asm_flac_cfg *cfg, int stream_id);
+
+int q6asm_media_format_block_alac(struct audio_client *ac,
+			struct asm_alac_cfg *cfg, int stream_id);
+
+int q6asm_stream_media_format_block_vorbis(struct audio_client *ac,
+			struct asm_vorbis_cfg *cfg, int stream_id);
+
+int q6asm_media_format_block_ape(struct audio_client *ac,
+			struct asm_ape_cfg *cfg, int stream_id);
+
+int q6asm_ds1_set_endp_params(struct audio_client *ac,
+				int param_id, int param_value);
+
+/* Send stream based end params */
+int q6asm_ds1_set_stream_endp_params(struct audio_client *ac, int param_id,
+				     int param_value, int stream_id);
+
+/* PP specific */
+int q6asm_equalizer(struct audio_client *ac, void *eq);
+
+/* Send Volume Command */
+int q6asm_set_volume(struct audio_client *ac, int volume);
+
+/* Send Volume Command */
+int q6asm_set_volume_v2(struct audio_client *ac, int volume, int instance);
+
+/* DTS Eagle Params */
+int q6asm_dts_eagle_set(struct audio_client *ac, int param_id, uint32_t size,
+			void *data, struct param_outband *po, int m_id);
+int q6asm_dts_eagle_get(struct audio_client *ac, int param_id, uint32_t size,
+			void *data, struct param_outband *po, int m_id);
+
+/* Set SoftPause Params */
+int q6asm_set_softpause(struct audio_client *ac,
+			struct asm_softpause_params *param);
+
+/* Set Softvolume Params */
+int q6asm_set_softvolume(struct audio_client *ac,
+			struct asm_softvolume_params *param);
+
+/* Set Softvolume Params */
+int q6asm_set_softvolume_v2(struct audio_client *ac,
+			    struct asm_softvolume_params *param, int instance);
+
+/* Send left-right channel gain */
+int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain);
+
+/* Send multi channel gain */
+int q6asm_set_multich_gain(struct audio_client *ac, uint32_t channels,
+			   uint32_t *gains, uint8_t *ch_map, bool use_default);
+
+/* Enable Mute/unmute flag */
+int q6asm_set_mute(struct audio_client *ac, int muteflag);
+
+int q6asm_get_session_time(struct audio_client *ac, uint64_t *tstamp);
+
+int q6asm_send_audio_effects_params(struct audio_client *ac, char *params,
+				    uint32_t params_length);
+
+/* Client can set the IO mode to either AIO/SIO mode */
+int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode);
+
+/* Get Service ID for APR communication */
+int q6asm_get_apr_service_id(int session_id);
+
+/* Common format block without any payload
+*/
+int q6asm_media_format_block(struct audio_client *ac, uint32_t format);
+
+/* Send the meta data to remove initial and trailing silence */
+int q6asm_send_meta_data(struct audio_client *ac, uint32_t initial_samples,
+		uint32_t trailing_samples);
+
+/* Send the stream meta data to remove initial and trailing silence */
+int q6asm_stream_send_meta_data(struct audio_client *ac, uint32_t stream_id,
+		uint32_t initial_samples, uint32_t trailing_samples);
+
+int q6asm_get_asm_topology(int session_id);
+int q6asm_get_asm_app_type(int session_id);
+
+int q6asm_send_mtmx_strtr_window(struct audio_client *ac,
+		struct asm_session_mtmx_strtr_param_window_v2_t *window_param,
+		uint32_t param_id);
+
+/* Retrieve the current DSP path delay */
+int q6asm_get_path_delay(struct audio_client *ac);
+
+#endif /* __Q6_ASM_H__ */
diff --git a/include/sound/q6audio-v2.h b/include/sound/q6audio-v2.h
new file mode 100644
index 000000000000..fd14f330d1d5
--- /dev/null
+++ b/include/sound/q6audio-v2.h
@@ -0,0 +1,36 @@
+/* Copyright (c) 2012-2013, 2015 The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef _Q6_AUDIO_H_
+#define _Q6_AUDIO_H_
+
+#include <linux/qdsp6v2/apr.h>
+
+enum {
+	LEGACY_PCM_MODE = 0,
+	LOW_LATENCY_PCM_MODE,
+	ULTRA_LOW_LATENCY_PCM_MODE,
+	ULL_POST_PROCESSING_PCM_MODE,
+};
+
+
+int q6audio_get_port_index(u16 port_id);
+
+int q6audio_convert_virtual_to_portid(u16 port_id);
+
+int q6audio_validate_port(u16 port_id);
+
+int q6audio_is_digital_pcm_interface(u16 port_id);
+
+int q6audio_get_port_id(u16 port_id);
+
+#endif
diff --git a/include/sound/q6core.h b/include/sound/q6core.h
new file mode 100644
index 000000000000..2c4e09a86c0b
--- /dev/null
+++ b/include/sound/q6core.h
@@ -0,0 +1,156 @@
+/* Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef __Q6CORE_H__
+#define __Q6CORE_H__
+#include <linux/qdsp6v2/apr.h>
+
+
+
+#define AVCS_CMD_ADSP_EVENT_GET_STATE		0x0001290C
+#define AVCS_CMDRSP_ADSP_EVENT_GET_STATE	0x0001290D
+
+bool q6core_is_adsp_ready(void);
+
+#define ADSP_CMD_SET_DTS_EAGLE_DATA_ID 0x00012919
+#define DTS_EAGLE_LICENSE_ID           0x00028346
+struct adsp_dts_eagle {
+	struct apr_hdr hdr;
+	uint32_t id;
+	uint32_t overwrite;
+	uint32_t size;
+	char data[];
+};
+int core_dts_eagle_set(int size, char *data);
+int core_dts_eagle_get(int id, int size, char *data);
+
+#define ADSP_CMD_SET_DOLBY_MANUFACTURER_ID 0x00012918
+
+struct adsp_dolby_manufacturer_id {
+	struct apr_hdr hdr;
+	int manufacturer_id;
+};
+
+uint32_t core_set_dolby_manufacturer_id(int manufacturer_id);
+
+/* Dolby Surround1 Module License ID. This ID is used as an identifier
+   for DS1 license via ADSP generic license mechanism.
+   Please refer AVCS_CMD_SET_LICENSE for more details.
+*/
+#define DOLBY_DS1_LICENSE_ID	0x00000001
+
+#define AVCS_CMD_SET_LICENSE	0x00012919
+struct avcs_cmd_set_license {
+	struct apr_hdr hdr;
+	uint32_t id; /**< A unique ID used to refer to this license */
+	uint32_t overwrite;
+	/**< 0 = do not overwrite an existing license with this id.
+		1 = overwrite an existing license with this id. */
+	uint32_t size;
+	/**< Size in bytes of the license data following this header. */
+	/* uint8_t* data ,  data and padding follows this structure
+		total packet size needs to be multiple of 4 Bytes*/
+
+};
+
+#define AVCS_CMD_GET_LICENSE_VALIDATION_RESULT        0x0001291A
+struct avcs_cmd_get_license_validation_result {
+	struct apr_hdr hdr;
+	uint32_t id; /**< A unique ID used to refer to this license */
+};
+
+#define AVCS_CMDRSP_GET_LICENSE_VALIDATION_RESULT        0x0001291B
+struct avcs_cmdrsp_get_license_validation_result {
+	uint32_t result;
+	/* ADSP_EOK if the license validation result was successfully retrieved.
+		ADSP_ENOTEXIST if there is no license with the given id.
+		ADSP_ENOTIMPL if there is no validation function for a license
+		with this id. */
+	uint32_t size;
+	/* Length in bytes of the result that follows this structure*/
+};
+
+/* Set Q6 topologies */
+/*
+ *	Registers custom topologies in the aDSP for
+ *	use in audio, voice, AFE and LSM.
+ */
+
+
+#define AVCS_CMD_SHARED_MEM_MAP_REGIONS                             0x00012924
+#define AVCS_CMDRSP_SHARED_MEM_MAP_REGIONS                          0x00012925
+#define AVCS_CMD_SHARED_MEM_UNMAP_REGIONS                           0x00012926
+
+
+#define AVCS_CMD_REGISTER_TOPOLOGIES                                0x00012923
+
+/* The payload for the AVCS_CMD_REGISTER_TOPOLOGIES command */
+struct avcs_cmd_register_topologies {
+	struct apr_hdr hdr;
+	uint32_t                  payload_addr_lsw;
+	/* Lower 32 bits of the topology buffer address. */
+
+	uint32_t                  payload_addr_msw;
+	/* Upper 32 bits of the topology buffer address. */
+
+	uint32_t                  mem_map_handle;
+	/* Unique identifier for an address.
+	 * -This memory map handle is returned by the aDSP through the
+	 * memory map command.
+	 * -NULL mem_map_handle is interpreted as in-band parameter
+	 * passing.
+	 * -Client has the flexibility to choose in-band or out-of-band.
+	 * -Out-of-band is recommended in this case.
+	 */
+
+	uint32_t                  payload_size;
+	/* Size in bytes of the valid data in the topology buffer. */
+} __packed;
+
+
+#define AVCS_CMD_DEREGISTER_TOPOLOGIES                                0x0001292a
+
+/* The payload for the AVCS_CMD_DEREGISTER_TOPOLOGIES command */
+struct avcs_cmd_deregister_topologies {
+	struct apr_hdr hdr;
+	uint32_t                  payload_addr_lsw;
+	/* Lower 32 bits of the topology buffer address. */
+
+	uint32_t                  payload_addr_msw;
+	/* Upper 32 bits of the topology buffer address. */
+
+	uint32_t                  mem_map_handle;
+	/* Unique identifier for an address.
+	 * -This memory map handle is returned by the aDSP through the
+	 * memory map command.
+	 * -NULL mem_map_handle is interpreted as in-band parameter
+	 * passing.
+	 * -Client has the flexibility to choose in-band or out-of-band.
+	 * -Out-of-band is recommended in this case.
+	 */
+
+	uint32_t                  payload_size;
+	/* Size in bytes of the valid data in the topology buffer. */
+
+	uint32_t                  mode;
+	/* 0: Deregister selected topologies
+	 * 1: Deregister all topologies
+	 */
+} __packed;
+
+#define AVCS_MODE_DEREGISTER_ALL_CUSTOM_TOPOLOGIES	1
+
+
+int32_t core_set_license(uint32_t key, uint32_t module_id);
+int32_t core_get_license_status(uint32_t module_id);
+
+#endif /* __Q6CORE_H__ */
diff --git a/include/sound/q6lsm.h b/include/sound/q6lsm.h
new file mode 100644
index 000000000000..7cb7e15941cb
--- /dev/null
+++ b/include/sound/q6lsm.h
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2013-2015, Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+#ifndef __Q6LSM_H__
+#define __Q6LSM_H__
+
+#include <linux/list.h>
+#include <linux/msm_ion.h>
+#include <sound/apr_audio-v2.h>
+#include <sound/lsm_params.h>
+#include <linux/qdsp6v2/apr.h>
+
+#define MAX_NUM_CONFIDENCE 20
+
+typedef void (*lsm_app_cb)(uint32_t opcode, uint32_t token,
+		       uint32_t *payload, void *priv);
+
+struct lsm_sound_model {
+	dma_addr_t      phys;
+	void		*data;
+	size_t		size; /* size of buffer */
+	uint32_t	actual_size; /* actual number of bytes read by DSP */
+	struct ion_handle *handle;
+	struct ion_client *client;
+	uint32_t	mem_map_handle;
+};
+
+struct snd_lsm_event_status_v2 {
+	uint16_t status;
+	uint16_t payload_size;
+	uint8_t  confidence_value[0];
+};
+
+struct lsm_lab_buffer {
+	dma_addr_t phys;
+	void *data;
+	size_t size;
+	struct ion_handle *handle;
+	struct ion_client *client;
+	uint32_t mem_map_handle;
+};
+
+struct lsm_lab_hw_params {
+	u16 sample_rate;
+	u16 sample_size;
+	u32 buf_sz;
+	u32 period_count;
+};
+
+struct lsm_client {
+	int		session;
+	lsm_app_cb	cb;
+	atomic_t	cmd_state;
+	void		*priv;
+	struct apr_svc  *apr;
+	struct apr_svc  *mmap_apr;
+	struct mutex    cmd_lock;
+	struct lsm_sound_model sound_model;
+	wait_queue_head_t cmd_wait;
+	uint32_t	cmd_err_code;
+	uint16_t	mode;
+	uint16_t	connect_to_port;
+	uint8_t		num_confidence_levels;
+	uint8_t		*confidence_levels;
+	bool		started;
+	dma_addr_t	lsm_cal_phy_addr;
+	uint32_t	lsm_cal_size;
+	uint32_t	app_id;
+	bool		lab_enable;
+	bool		lab_started;
+	struct lsm_lab_buffer *lab_buffer;
+	struct lsm_lab_hw_params hw_params;
+	bool		use_topology;
+};
+
+struct lsm_stream_cmd_open_tx {
+	struct apr_hdr  hdr;
+	uint16_t	app_id;
+	uint16_t	reserved;
+	uint32_t	sampling_rate;
+} __packed;
+
+struct lsm_stream_cmd_open_tx_v2 {
+	struct apr_hdr hdr;
+	uint32_t	topology_id;
+} __packed;
+
+struct lsm_custom_topologies {
+	struct apr_hdr hdr;
+	uint32_t data_payload_addr_lsw;
+	uint32_t data_payload_addr_msw;
+	uint32_t mem_map_handle;
+	uint32_t buffer_size;
+} __packed;
+
+struct lsm_param_size_reserved {
+	uint16_t param_size;
+	uint16_t reserved;
+} __packed;
+
+union lsm_param_size {
+	uint32_t param_size;
+	struct lsm_param_size_reserved sr;
+} __packed;
+
+struct lsm_param_payload_common {
+	uint32_t	module_id;
+	uint32_t	param_id;
+	union lsm_param_size p_size;
+} __packed;
+
+struct lsm_param_op_mode {
+	struct lsm_param_payload_common common;
+	uint32_t	minor_version;
+	uint16_t	mode;
+	uint16_t	reserved;
+} __packed;
+
+struct lsm_param_connect_to_port {
+	struct lsm_param_payload_common common;
+	uint32_t	minor_version;
+	/* AFE port id that receives voice wake up data */
+	uint16_t	port_id;
+	uint16_t	reserved;
+} __packed;
+
+
+/*
+ * This param cannot be sent in this format.
+ * The actual number of confidence level values
+ * need to appended to this param payload.
+ */
+struct lsm_param_min_confidence_levels {
+	struct lsm_param_payload_common common;
+	uint8_t		num_confidence_levels;
+} __packed;
+
+struct lsm_set_params_hdr {
+	uint32_t	data_payload_size;
+	uint32_t	data_payload_addr_lsw;
+	uint32_t	data_payload_addr_msw;
+	uint32_t	mem_map_handle;
+} __packed;
+
+struct lsm_cmd_set_params {
+	struct apr_hdr  msg_hdr;
+	struct lsm_set_params_hdr param_hdr;
+} __packed;
+
+struct lsm_cmd_set_params_conf {
+	struct apr_hdr  msg_hdr;
+	struct lsm_set_params_hdr params_hdr;
+	struct lsm_param_min_confidence_levels	conf_payload;
+} __packed;
+
+struct lsm_cmd_set_opmode_connectport {
+	struct apr_hdr  msg_hdr;
+	struct lsm_set_params_hdr params_hdr;
+	struct lsm_param_connect_to_port	connect_to_port;
+	struct lsm_param_op_mode		op_mode;
+} __packed;
+
+struct lsm_param_epd_thres {
+	struct lsm_param_payload_common common;
+	uint32_t	minor_version;
+	uint32_t	epd_begin;
+	uint32_t	epd_end;
+} __packed;
+
+struct lsm_cmd_set_epd_threshold {
+	struct apr_hdr msg_hdr;
+	struct lsm_set_params_hdr param_hdr;
+	struct lsm_param_epd_thres epd_thres;
+} __packed;
+
+struct lsm_param_gain {
+	struct lsm_param_payload_common common;
+	uint32_t	minor_version;
+	uint16_t	gain;
+	uint16_t	reserved;
+} __packed;
+
+struct lsm_cmd_set_gain {
+	struct apr_hdr msg_hdr;
+	struct lsm_set_params_hdr param_hdr;
+	struct lsm_param_gain lsm_gain;
+} __packed;
+
+struct lsm_cmd_reg_snd_model {
+	struct apr_hdr	hdr;
+	uint32_t	model_size;
+	uint32_t	model_addr_lsw;
+	uint32_t	model_addr_msw;
+	uint32_t	mem_map_handle;
+} __packed;
+
+struct lsm_lab_enable {
+	struct lsm_param_payload_common common;
+	uint16_t enable;
+	uint16_t reserved;
+} __packed;
+
+struct lsm_params_lab_enable {
+	struct apr_hdr msg_hdr;
+	struct lsm_set_params_hdr params_hdr;
+	struct lsm_lab_enable lab_enable;
+} __packed;
+
+struct lsm_lab_config {
+	struct lsm_param_payload_common common;
+	uint32_t minor_version;
+	uint32_t wake_up_latency_ms;
+} __packed;
+
+
+struct lsm_params_lab_config {
+	struct apr_hdr  msg_hdr;
+	struct lsm_set_params_hdr params_hdr;
+	struct lsm_lab_config lab_config;
+} __packed;
+
+struct lsm_cmd_read {
+	struct apr_hdr hdr;
+	uint32_t buf_addr_lsw;
+	uint32_t buf_addr_msw;
+	uint32_t mem_map_handle;
+	uint32_t buf_size;
+} __packed;
+
+struct lsm_cmd_read_done {
+	struct apr_hdr hdr;
+	uint32_t status;
+	uint32_t buf_addr_lsw;
+	uint32_t buf_addr_msw;
+	uint32_t mem_map_handle;
+	uint32_t total_size;
+	uint32_t offset;
+	uint32_t timestamp_lsw;
+	uint32_t timestamp_msw;
+	uint32_t flags;
+} __packed;
+
+struct lsm_client *q6lsm_client_alloc(lsm_app_cb cb, void *priv);
+void q6lsm_client_free(struct lsm_client *client);
+int q6lsm_open(struct lsm_client *client, uint16_t app_id);
+int q6lsm_start(struct lsm_client *client, bool wait);
+int q6lsm_stop(struct lsm_client *client, bool wait);
+int q6lsm_snd_model_buf_alloc(struct lsm_client *client, size_t len,
+			      bool allocate_module_data);
+int q6lsm_snd_model_buf_free(struct lsm_client *client);
+int q6lsm_close(struct lsm_client *client);
+int q6lsm_register_sound_model(struct lsm_client *client,
+			       enum lsm_detection_mode mode,
+			       bool detectfailure);
+int q6lsm_set_data(struct lsm_client *client,
+		   enum lsm_detection_mode mode,
+		   bool detectfailure);
+int q6lsm_deregister_sound_model(struct lsm_client *client);
+void set_lsm_port(int);
+int get_lsm_port(void);
+int q6lsm_lab_control(struct lsm_client *client, u32 enable);
+int q6lsm_stop_lab(struct lsm_client *client);
+int q6lsm_read(struct lsm_client *client, struct lsm_cmd_read *read);
+int q6lsm_lab_buffer_alloc(struct lsm_client *client, bool alloc);
+int q6lsm_set_one_param(struct lsm_client *client,
+			struct lsm_params_info *p_info, void *data,
+			enum LSM_PARAM_TYPE param_type);
+void q6lsm_sm_set_param_data(struct lsm_client *client,
+		struct lsm_params_info *p_info,
+		size_t *offset);
+#endif /* __Q6LSM_H__ */
diff --git a/include/sound/voice_params.h b/include/sound/voice_params.h
new file mode 100644
index 000000000000..43e3b9d0aa49
--- /dev/null
+++ b/include/sound/voice_params.h
@@ -0,0 +1,14 @@
+#ifndef __VOICE_PARAMS_H__
+#define __VOICE_PARAMS_H__
+
+#include <linux/types.h>
+#include <sound/asound.h>
+
+enum voice_lch_mode {
+	VOICE_LCH_START = 1,
+	VOICE_LCH_STOP
+};
+
+#define SNDRV_VOICE_IOCTL_LCH _IOW('U', 0x00, enum voice_lch_mode)
+
+#endif
diff --git a/include/sound/voice_svc.h b/include/sound/voice_svc.h
new file mode 100644
index 000000000000..035053f091ef
--- /dev/null
+++ b/include/sound/voice_svc.h
@@ -0,0 +1,47 @@
+#ifndef __VOICE_SVC_H__
+#define __VOICE_SVC_H__
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define VOICE_SVC_DRIVER_NAME "voice_svc"
+
+#define VOICE_SVC_MVM_STR "MVM"
+#define VOICE_SVC_CVS_STR "CVS"
+#define MAX_APR_SERVICE_NAME_LEN  64
+
+#define MSG_REGISTER 0x1
+#define MSG_REQUEST  0x2
+#define MSG_RESPONSE 0x3
+
+struct voice_svc_write_msg {
+	__u32 msg_type;
+	__u8 payload[0];
+};
+
+struct voice_svc_register {
+	char svc_name[MAX_APR_SERVICE_NAME_LEN];
+	__u32 src_port;
+	__u8 reg_flag;
+};
+
+struct voice_svc_cmd_response {
+	__u32 src_port;
+	__u32 dest_port;
+	__u32 token;
+	__u32 opcode;
+	__u32 payload_size;
+	__u8 payload[0];
+};
+
+struct voice_svc_cmd_request {
+	char svc_name[MAX_APR_SERVICE_NAME_LEN];
+	__u32 src_port;
+	__u32 dest_port;
+	__u32 token;
+	__u32 opcode;
+	__u32 payload_size;
+	__u8 payload[0];
+};
+
+#endif
diff --git a/include/uapi/linux/Kbuild b/include/uapi/linux/Kbuild
index 1e9d279b9536..da52f0b7dbaf 100644
--- a/include/uapi/linux/Kbuild
+++ b/include/uapi/linux/Kbuild
@@ -63,6 +63,7 @@ header-y += audit.h
 header-y += auto_fs4.h
 header-y += auto_fs.h
 header-y += auxvec.h
+header-y += avtimer.h
 header-y += ax25.h
 header-y += b1lli.h
 header-y += baycom.h
@@ -277,6 +278,22 @@ header-y += mroute6.h
 header-y += mroute.h
 header-y += msdos_fs.h
 header-y += msg.h
+header-y += msm_adsp.h
+header-y += msm_audio.h
+header-y += msm_audio_aac.h
+header-y += msm_audio_ac3.h
+header-y += msm_audio_amrnb.h
+header-y += msm_audio_amrwb.h
+header-y += msm_audio_amrwbplus.h
+header-y += msm_audio_calibration.h
+header-y += msm_audio_mvs.h
+header-y += msm_audio_qcp.h
+header-y += msm_audio_sbc.h
+header-y += msm_audio_voicememo.h
+header-y += msm_audio_wma.h
+header-y += msm_audio_wmapro.h
+header-y += msm_audio_alac.h
+header-y += msm_audio_ape.h
 header-y += msm_ion.h
 header-y += msm_kgsl.h
 header-y += msm_rmnet.h
diff --git a/include/uapi/linux/avtimer.h b/include/uapi/linux/avtimer.h
new file mode 100644
index 000000000000..96b5483fbf2e
--- /dev/null
+++ b/include/uapi/linux/avtimer.h
@@ -0,0 +1,10 @@
+#ifndef _UAPI_AVTIMER_H
+#define _UAPI_AVTIMER_H
+
+#include <linux/ioctl.h>
+
+#define MAJOR_NUM 100
+
+#define IOCTL_GET_AVTIMER_TICK _IOR(MAJOR_NUM, 0, uint64_t)
+
+#endif
diff --git a/include/uapi/linux/msm_adsp.h b/include/uapi/linux/msm_adsp.h
new file mode 100644
index 000000000000..65c31ac98334
--- /dev/null
+++ b/include/uapi/linux/msm_adsp.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2007 Google, Inc.
+ * Author: Iliyan Malchev <ibm@android.com>
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * 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.
+ *
+ */
+#ifndef _UAPI_LINUX_MSM_ADSP_H
+#define _UAPI_LINUX_MSM_ADSP_H
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define ADSP_IOCTL_MAGIC 'q'
+
+/* ADSP_IOCTL_WRITE_COMMAND */
+struct adsp_command_t {
+	uint16_t queue;
+	uint32_t len;		/* bytes */
+	uint8_t *data;
+};
+
+/* ADSP_IOCTL_GET_EVENT */
+struct adsp_event_t {
+	uint16_t type;		/* 1 == event (RPC), 0 == message (adsp) */
+	uint32_t timeout_ms;	/* -1 for infinite, 0 for immediate return */
+	uint16_t msg_id;
+	uint16_t flags;		/* 1 == 16--bit event, 0 == 32-bit event */
+	uint32_t len;		/* size in, number of bytes out */
+	uint8_t *data;
+};
+
+#define ADSP_IOCTL_ENABLE \
+	_IOR(ADSP_IOCTL_MAGIC, 1, unsigned)
+
+#define ADSP_IOCTL_DISABLE \
+	_IOR(ADSP_IOCTL_MAGIC, 2, unsigned)
+
+#define ADSP_IOCTL_DISABLE_ACK \
+	_IOR(ADSP_IOCTL_MAGIC, 3, unsigned)
+
+#define ADSP_IOCTL_WRITE_COMMAND \
+	_IOR(ADSP_IOCTL_MAGIC, 4, struct adsp_command_t *)
+
+#define ADSP_IOCTL_GET_EVENT \
+	_IOWR(ADSP_IOCTL_MAGIC, 5, struct adsp_event_data_t *)
+
+#define ADSP_IOCTL_SET_CLKRATE \
+	_IOR(ADSP_IOCTL_MAGIC, 6, unsigned)
+
+#define ADSP_IOCTL_DISABLE_EVENT_RSP \
+	_IOR(ADSP_IOCTL_MAGIC, 10, unsigned)
+
+#define ADSP_IOCTL_REGISTER_PMEM \
+	_IOW(ADSP_IOCTL_MAGIC, 13, unsigned)
+
+#define ADSP_IOCTL_UNREGISTER_PMEM \
+	_IOW(ADSP_IOCTL_MAGIC, 14, unsigned)
+
+/* Cause any further GET_EVENT ioctls to fail (-ENODEV)
+ * until the device is closed and reopened.  Useful for
+ * terminating event dispatch threads
+ */
+#define ADSP_IOCTL_ABORT_EVENT_READ \
+	_IOW(ADSP_IOCTL_MAGIC, 15, unsigned)
+
+#define ADSP_IOCTL_LINK_TASK \
+	_IOW(ADSP_IOCTL_MAGIC, 16, unsigned)
+
+#endif
diff --git a/include/uapi/linux/msm_audio.h b/include/uapi/linux/msm_audio.h
new file mode 100644
index 000000000000..36b66c7cde76
--- /dev/null
+++ b/include/uapi/linux/msm_audio.h
@@ -0,0 +1,463 @@
+/* include/linux/msm_audio.h
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (c) 2012, 2014 The Linux Foundation. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * 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.
+ *
+ */
+
+#ifndef _UAPI_LINUX_MSM_AUDIO_H
+#define _UAPI_LINUX_MSM_AUDIO_H
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+/* PCM Audio */
+
+#define AUDIO_IOCTL_MAGIC 'a'
+
+#define AUDIO_START        _IOW(AUDIO_IOCTL_MAGIC, 0, unsigned)
+#define AUDIO_STOP         _IOW(AUDIO_IOCTL_MAGIC, 1, unsigned)
+#define AUDIO_FLUSH        _IOW(AUDIO_IOCTL_MAGIC, 2, unsigned)
+#define AUDIO_GET_CONFIG   _IOR(AUDIO_IOCTL_MAGIC, 3, \
+		struct msm_audio_config)
+#define AUDIO_SET_CONFIG   _IOW(AUDIO_IOCTL_MAGIC, 4, \
+		struct msm_audio_config)
+#define AUDIO_GET_STATS    _IOR(AUDIO_IOCTL_MAGIC, 5, \
+		struct msm_audio_stats)
+#define AUDIO_ENABLE_AUDPP _IOW(AUDIO_IOCTL_MAGIC, 6, unsigned)
+#define AUDIO_SET_ADRC     _IOW(AUDIO_IOCTL_MAGIC, 7, unsigned)
+#define AUDIO_SET_EQ       _IOW(AUDIO_IOCTL_MAGIC, 8, unsigned)
+#define AUDIO_SET_RX_IIR   _IOW(AUDIO_IOCTL_MAGIC, 9, unsigned)
+#define AUDIO_SET_VOLUME   _IOW(AUDIO_IOCTL_MAGIC, 10, unsigned)
+#define AUDIO_PAUSE        _IOW(AUDIO_IOCTL_MAGIC, 11, unsigned)
+#define AUDIO_PLAY_DTMF    _IOW(AUDIO_IOCTL_MAGIC, 12, unsigned)
+#define AUDIO_GET_EVENT    _IOR(AUDIO_IOCTL_MAGIC, 13, \
+		struct msm_audio_event)
+#define AUDIO_ABORT_GET_EVENT _IOW(AUDIO_IOCTL_MAGIC, 14, unsigned)
+#define AUDIO_REGISTER_PMEM _IOW(AUDIO_IOCTL_MAGIC, 15, unsigned)
+#define AUDIO_DEREGISTER_PMEM _IOW(AUDIO_IOCTL_MAGIC, 16, unsigned)
+#define AUDIO_ASYNC_WRITE _IOW(AUDIO_IOCTL_MAGIC, 17, \
+		struct msm_audio_aio_buf)
+#define AUDIO_ASYNC_READ _IOW(AUDIO_IOCTL_MAGIC, 18, \
+		struct msm_audio_aio_buf)
+#define AUDIO_SET_INCALL _IOW(AUDIO_IOCTL_MAGIC, 19, struct msm_voicerec_mode)
+#define AUDIO_GET_NUM_SND_DEVICE _IOR(AUDIO_IOCTL_MAGIC, 20, unsigned)
+#define AUDIO_GET_SND_DEVICES _IOWR(AUDIO_IOCTL_MAGIC, 21, \
+				struct msm_snd_device_list)
+#define AUDIO_ENABLE_SND_DEVICE _IOW(AUDIO_IOCTL_MAGIC, 22, unsigned)
+#define AUDIO_DISABLE_SND_DEVICE _IOW(AUDIO_IOCTL_MAGIC, 23, unsigned)
+#define AUDIO_ROUTE_STREAM _IOW(AUDIO_IOCTL_MAGIC, 24, \
+				struct msm_audio_route_config)
+#define AUDIO_GET_PCM_CONFIG _IOR(AUDIO_IOCTL_MAGIC, 30, unsigned)
+#define AUDIO_SET_PCM_CONFIG _IOW(AUDIO_IOCTL_MAGIC, 31, unsigned)
+#define AUDIO_SWITCH_DEVICE  _IOW(AUDIO_IOCTL_MAGIC, 32, unsigned)
+#define AUDIO_SET_MUTE       _IOW(AUDIO_IOCTL_MAGIC, 33, unsigned)
+#define AUDIO_UPDATE_ACDB    _IOW(AUDIO_IOCTL_MAGIC, 34, unsigned)
+#define AUDIO_START_VOICE    _IOW(AUDIO_IOCTL_MAGIC, 35, unsigned)
+#define AUDIO_STOP_VOICE     _IOW(AUDIO_IOCTL_MAGIC, 36, unsigned)
+#define AUDIO_REINIT_ACDB    _IOW(AUDIO_IOCTL_MAGIC, 39, unsigned)
+#define AUDIO_OUTPORT_FLUSH  _IOW(AUDIO_IOCTL_MAGIC, 40, unsigned short)
+#define AUDIO_SET_ERR_THRESHOLD_VALUE _IOW(AUDIO_IOCTL_MAGIC, 41, \
+					unsigned short)
+#define AUDIO_GET_BITSTREAM_ERROR_INFO _IOR(AUDIO_IOCTL_MAGIC, 42, \
+			       struct msm_audio_bitstream_error_info)
+
+#define AUDIO_SET_SRS_TRUMEDIA_PARAM _IOW(AUDIO_IOCTL_MAGIC, 43, unsigned)
+
+/* Qualcomm extensions */
+#define AUDIO_SET_STREAM_CONFIG   _IOW(AUDIO_IOCTL_MAGIC, 80, \
+				struct msm_audio_stream_config)
+#define AUDIO_GET_STREAM_CONFIG   _IOR(AUDIO_IOCTL_MAGIC, 81, \
+				struct msm_audio_stream_config)
+#define AUDIO_GET_SESSION_ID _IOR(AUDIO_IOCTL_MAGIC, 82, unsigned short)
+#define AUDIO_GET_STREAM_INFO   _IOR(AUDIO_IOCTL_MAGIC, 83, \
+			       struct msm_audio_bitstream_info)
+#define AUDIO_SET_PAN       _IOW(AUDIO_IOCTL_MAGIC, 84, unsigned)
+#define AUDIO_SET_QCONCERT_PLUS       _IOW(AUDIO_IOCTL_MAGIC, 85, unsigned)
+#define AUDIO_SET_MBADRC       _IOW(AUDIO_IOCTL_MAGIC, 86, unsigned)
+#define AUDIO_SET_VOLUME_PATH   _IOW(AUDIO_IOCTL_MAGIC, 87, \
+				     struct msm_vol_info)
+#define AUDIO_SET_MAX_VOL_ALL _IOW(AUDIO_IOCTL_MAGIC, 88, unsigned)
+#define AUDIO_ENABLE_AUDPRE  _IOW(AUDIO_IOCTL_MAGIC, 89, unsigned)
+#define AUDIO_SET_AGC        _IOW(AUDIO_IOCTL_MAGIC, 90, unsigned)
+#define AUDIO_SET_NS         _IOW(AUDIO_IOCTL_MAGIC, 91, unsigned)
+#define AUDIO_SET_TX_IIR     _IOW(AUDIO_IOCTL_MAGIC, 92, unsigned)
+#define AUDIO_GET_BUF_CFG    _IOW(AUDIO_IOCTL_MAGIC, 93, \
+					struct msm_audio_buf_cfg)
+#define AUDIO_SET_BUF_CFG    _IOW(AUDIO_IOCTL_MAGIC, 94, \
+					struct msm_audio_buf_cfg)
+#define AUDIO_SET_ACDB_BLK _IOW(AUDIO_IOCTL_MAGIC, 95,  \
+					struct msm_acdb_cmd_device)
+#define AUDIO_GET_ACDB_BLK _IOW(AUDIO_IOCTL_MAGIC, 96,  \
+					struct msm_acdb_cmd_device)
+
+#define AUDIO_REGISTER_ION _IOW(AUDIO_IOCTL_MAGIC, 97, \
+		struct msm_audio_ion_info)
+#define AUDIO_DEREGISTER_ION _IOW(AUDIO_IOCTL_MAGIC, 98, \
+		struct msm_audio_ion_info)
+#define AUDIO_SET_EFFECTS_CONFIG   _IOW(AUDIO_IOCTL_MAGIC, 99, \
+				struct msm_hwacc_effects_config)
+#define AUDIO_EFFECTS_SET_BUF_LEN _IOW(AUDIO_IOCTL_MAGIC, 100, \
+				struct msm_hwacc_buf_cfg)
+#define AUDIO_EFFECTS_GET_BUF_AVAIL _IOW(AUDIO_IOCTL_MAGIC, 101, \
+				struct msm_hwacc_buf_avail)
+#define AUDIO_EFFECTS_WRITE _IOW(AUDIO_IOCTL_MAGIC, 102, void *)
+#define AUDIO_EFFECTS_READ _IOWR(AUDIO_IOCTL_MAGIC, 103, void *)
+#define AUDIO_EFFECTS_SET_PP_PARAMS _IOW(AUDIO_IOCTL_MAGIC, 104, void *)
+
+#define AUDIO_PM_AWAKE      _IOW(AUDIO_IOCTL_MAGIC, 105, unsigned)
+#define AUDIO_PM_RELAX      _IOW(AUDIO_IOCTL_MAGIC, 106, unsigned)
+
+#define	AUDIO_MAX_COMMON_IOCTL_NUM	107
+
+
+#define HANDSET_MIC			0x01
+#define HANDSET_SPKR			0x02
+#define HEADSET_MIC			0x03
+#define HEADSET_SPKR_MONO		0x04
+#define HEADSET_SPKR_STEREO		0x05
+#define SPKR_PHONE_MIC			0x06
+#define SPKR_PHONE_MONO			0x07
+#define SPKR_PHONE_STEREO		0x08
+#define BT_SCO_MIC			0x09
+#define BT_SCO_SPKR			0x0A
+#define BT_A2DP_SPKR			0x0B
+#define TTY_HEADSET_MIC			0x0C
+#define TTY_HEADSET_SPKR		0x0D
+
+/* Default devices are not supported in a */
+/* device switching context. Only supported */
+/* for stream devices. */
+/* DO NOT USE */
+#define DEFAULT_TX			0x0E
+#define DEFAULT_RX			0x0F
+
+#define BT_A2DP_TX			0x10
+
+#define HEADSET_MONO_PLUS_SPKR_MONO_RX         0x11
+#define HEADSET_MONO_PLUS_SPKR_STEREO_RX       0x12
+#define HEADSET_STEREO_PLUS_SPKR_MONO_RX       0x13
+#define HEADSET_STEREO_PLUS_SPKR_STEREO_RX     0x14
+
+#define I2S_RX				0x20
+#define I2S_TX				0x21
+
+#define ADRC_ENABLE		0x0001
+#define EQUALIZER_ENABLE	0x0002
+#define IIR_ENABLE		0x0004
+#define QCONCERT_PLUS_ENABLE	0x0008
+#define MBADRC_ENABLE		0x0010
+#define SRS_ENABLE		0x0020
+#define SRS_DISABLE	0x0040
+
+#define AGC_ENABLE		0x0001
+#define NS_ENABLE		0x0002
+#define TX_IIR_ENABLE		0x0004
+#define FLUENCE_ENABLE		0x0008
+
+#define VOC_REC_UPLINK		0x00
+#define VOC_REC_DOWNLINK	0x01
+#define VOC_REC_BOTH		0x02
+
+struct msm_audio_config {
+	uint32_t buffer_size;
+	uint32_t buffer_count;
+	uint32_t channel_count;
+	uint32_t sample_rate;
+	uint32_t type;
+	uint32_t meta_field;
+	uint32_t bits;
+	uint32_t unused[3];
+};
+
+struct msm_audio_stream_config {
+	uint32_t buffer_size;
+	uint32_t buffer_count;
+};
+
+struct msm_audio_buf_cfg{
+	uint32_t meta_info_enable;
+	uint32_t frames_per_buf;
+};
+
+struct msm_audio_stats {
+	uint32_t byte_count;
+	uint32_t sample_count;
+	uint32_t unused[2];
+};
+
+struct msm_audio_ion_info {
+	int fd;
+	void *vaddr;
+};
+
+struct msm_audio_pmem_info {
+	int fd;
+	void *vaddr;
+};
+
+struct msm_audio_aio_buf {
+	void *buf_addr;
+	uint32_t buf_len;
+	uint32_t data_len;
+	void *private_data;
+	unsigned short mfield_sz; /*only useful for data has meta field */
+};
+
+/* Audio routing */
+
+#define SND_IOCTL_MAGIC 's'
+
+#define SND_MUTE_UNMUTED 0
+#define SND_MUTE_MUTED   1
+
+struct msm_mute_info {
+	uint32_t mute;
+	uint32_t path;
+};
+
+struct msm_vol_info {
+	uint32_t vol;
+	uint32_t path;
+};
+
+struct msm_voicerec_mode {
+	uint32_t rec_mode;
+};
+
+struct msm_snd_device_config {
+	uint32_t device;
+	uint32_t ear_mute;
+	uint32_t mic_mute;
+};
+
+#define SND_SET_DEVICE _IOW(SND_IOCTL_MAGIC, 2, struct msm_device_config *)
+
+enum cad_device_path_type {
+	CAD_DEVICE_PATH_RX,	/*For Decoding session*/
+	CAD_DEVICE_PATH_TX,	/* For Encoding session*/
+	CAD_DEVICE_PATH_RX_TX, /* For Voice call */
+	CAD_DEVICE_PATH_LB,	/* For loopback (FM Analog)*/
+	CAD_DEVICE_PATH_MAX
+};
+
+struct cad_devices_type {
+	uint32_t rx_device;
+	uint32_t tx_device;
+	enum cad_device_path_type pathtype;
+};
+
+struct msm_cad_device_config {
+	struct cad_devices_type device;
+	uint32_t ear_mute;
+	uint32_t mic_mute;
+};
+
+#define CAD_SET_DEVICE _IOW(SND_IOCTL_MAGIC, 2, struct msm_cad_device_config *)
+
+#define SND_METHOD_VOICE 0
+#define SND_METHOD_MIDI 4
+
+struct msm_snd_volume_config {
+	uint32_t device;
+	uint32_t method;
+	uint32_t volume;
+};
+
+#define SND_SET_VOLUME _IOW(SND_IOCTL_MAGIC, 3, struct msm_snd_volume_config *)
+
+struct msm_cad_volume_config {
+	struct cad_devices_type device;
+	uint32_t method;
+	uint32_t volume;
+};
+
+#define CAD_SET_VOLUME _IOW(SND_IOCTL_MAGIC, 3, struct msm_cad_volume_config *)
+
+/* Returns the number of SND endpoints supported. */
+
+#define SND_GET_NUM_ENDPOINTS _IOR(SND_IOCTL_MAGIC, 4, unsigned *)
+
+struct msm_snd_endpoint {
+	int id; /* input and output */
+	char name[64]; /* output only */
+};
+
+/* Takes an index between 0 and one less than the number returned by
+ * SND_GET_NUM_ENDPOINTS, and returns the SND index and name of a
+ * SND endpoint.  On input, the .id field contains the number of the
+ * endpoint, and on exit it contains the SND index, while .name contains
+ * the description of the endpoint.
+ */
+
+#define SND_GET_ENDPOINT _IOWR(SND_IOCTL_MAGIC, 5, struct msm_snd_endpoint *)
+
+
+#define SND_AVC_CTL _IOW(SND_IOCTL_MAGIC, 6, unsigned *)
+#define SND_AGC_CTL _IOW(SND_IOCTL_MAGIC, 7, unsigned *)
+
+/*return the number of CAD endpoints supported. */
+
+#define CAD_GET_NUM_ENDPOINTS _IOR(SND_IOCTL_MAGIC, 4, unsigned *)
+
+struct msm_cad_endpoint {
+	int id; /* input and output */
+	char name[64]; /* output only */
+};
+
+/* Takes an index between 0 and one less than the number returned by
+ * SND_GET_NUM_ENDPOINTS, and returns the CAD index and name of a
+ * CAD endpoint.  On input, the .id field contains the number of the
+ * endpoint, and on exit it contains the SND index, while .name contains
+ * the description of the endpoint.
+ */
+
+#define CAD_GET_ENDPOINT _IOWR(SND_IOCTL_MAGIC, 5, struct msm_cad_endpoint *)
+
+struct msm_audio_pcm_config {
+	uint32_t pcm_feedback;	/* 0 - disable > 0 - enable */
+	uint32_t buffer_count;	/* Number of buffers to allocate */
+	uint32_t buffer_size;	/* Size of buffer for capturing of
+				   PCM samples */
+};
+
+#define AUDIO_EVENT_SUSPEND 0
+#define AUDIO_EVENT_RESUME 1
+#define AUDIO_EVENT_WRITE_DONE 2
+#define AUDIO_EVENT_READ_DONE   3
+#define AUDIO_EVENT_STREAM_INFO 4
+#define AUDIO_EVENT_BITSTREAM_ERROR_INFO 5
+
+#define AUDIO_CODEC_TYPE_MP3 0
+#define AUDIO_CODEC_TYPE_AAC 1
+
+struct msm_audio_bitstream_info {
+	uint32_t codec_type;
+	uint32_t chan_info;
+	uint32_t sample_rate;
+	uint32_t bit_stream_info;
+	uint32_t bit_rate;
+	uint32_t unused[3];
+};
+
+struct msm_audio_bitstream_error_info {
+	uint32_t dec_id;
+	uint32_t err_msg_indicator;
+	uint32_t err_type;
+};
+
+union msm_audio_event_payload {
+	struct msm_audio_aio_buf aio_buf;
+	struct msm_audio_bitstream_info stream_info;
+	struct msm_audio_bitstream_error_info error_info;
+	int reserved;
+};
+
+struct msm_audio_event {
+	int event_type;
+	int timeout_ms;
+	union msm_audio_event_payload event_payload;
+};
+
+#define MSM_SNDDEV_CAP_RX 0x1
+#define MSM_SNDDEV_CAP_TX 0x2
+#define MSM_SNDDEV_CAP_VOICE 0x4
+
+struct msm_snd_device_info {
+	uint32_t dev_id;
+	uint32_t dev_cap; /* bitmask describe capability of device */
+	char dev_name[64];
+};
+
+struct msm_snd_device_list {
+	uint32_t  num_dev; /* Indicate number of device info to be retrieved */
+	struct msm_snd_device_info *list;
+};
+
+struct msm_dtmf_config {
+	uint16_t path;
+	uint16_t dtmf_hi;
+	uint16_t dtmf_low;
+	uint16_t duration;
+	uint16_t tx_gain;
+	uint16_t rx_gain;
+	uint16_t mixing;
+};
+
+#define AUDIO_ROUTE_STREAM_VOICE_RX 0
+#define AUDIO_ROUTE_STREAM_VOICE_TX 1
+#define AUDIO_ROUTE_STREAM_PLAYBACK 2
+#define AUDIO_ROUTE_STREAM_REC      3
+
+struct msm_audio_route_config {
+	uint32_t stream_type;
+	uint32_t stream_id;
+	uint32_t dev_id;
+};
+
+#define AUDIO_MAX_EQ_BANDS 12
+
+struct msm_audio_eq_band {
+	uint16_t     band_idx; /* The band index, 0 .. 11 */
+	uint32_t     filter_type; /* Filter band type */
+	uint32_t     center_freq_hz; /* Filter band center frequency */
+	uint32_t     filter_gain; /* Filter band initial gain (dB) */
+			/* Range is +12 dB to -12 dB with 1dB increments. */
+	uint32_t     q_factor;
+} __attribute__ ((packed));
+
+struct msm_audio_eq_stream_config {
+	uint32_t	enable; /* Number of consequtive bands specified */
+	uint32_t	num_bands;
+	struct msm_audio_eq_band	eq_bands[AUDIO_MAX_EQ_BANDS];
+} __attribute__ ((packed));
+
+struct msm_acdb_cmd_device {
+	uint32_t     command_id;
+	uint32_t     device_id;
+	uint32_t     network_id;
+	uint32_t     sample_rate_id;      /* Actual sample rate value */
+	uint32_t     interface_id;        /* See interface id's above */
+	uint32_t     algorithm_block_id;  /* See enumerations above */
+	uint32_t     total_bytes;         /* Length in bytes used by buffer */
+	uint32_t     *phys_buf;           /* Physical Address of data */
+};
+
+struct msm_hwacc_data_config {
+	__u32 buf_size;
+	__u32 num_buf;
+	__u32 num_channels;
+	__u8 channel_map[8];
+	__u32 sample_rate;
+	__u32 bits_per_sample;
+};
+
+struct msm_hwacc_buf_cfg {
+	__u32 input_len;
+	__u32 output_len;
+};
+
+struct msm_hwacc_buf_avail {
+	__u32 input_num_avail;
+	__u32 output_num_avail;
+};
+
+struct msm_hwacc_effects_config {
+	struct msm_hwacc_data_config input;
+	struct msm_hwacc_data_config output;
+	struct msm_hwacc_buf_cfg buf_cfg;
+	__u32 meta_mode_enabled;
+	__u32 overwrite_topology;
+	__s32 topology;
+};
+
+#endif
diff --git a/include/uapi/linux/msm_audio_aac.h b/include/uapi/linux/msm_audio_aac.h
new file mode 100644
index 000000000000..fadeb74c2ea7
--- /dev/null
+++ b/include/uapi/linux/msm_audio_aac.h
@@ -0,0 +1,76 @@
+#ifndef _UAPI_MSM_AUDIO_AAC_H
+#define _UAPI_MSM_AUDIO_AAC_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_SET_AAC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+0), struct msm_audio_aac_config)
+#define AUDIO_GET_AAC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+1), struct msm_audio_aac_config)
+
+#define AUDIO_SET_AAC_ENC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+3), struct msm_audio_aac_enc_config)
+
+#define AUDIO_GET_AAC_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+4), struct msm_audio_aac_enc_config)
+
+#define AUDIO_SET_AAC_MIX_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+5), uint32_t)
+
+#define AUDIO_AAC_FORMAT_ADTS		-1
+#define	AUDIO_AAC_FORMAT_RAW		0x0000
+#define	AUDIO_AAC_FORMAT_PSUEDO_RAW	0x0001
+#define AUDIO_AAC_FORMAT_LOAS		0x0002
+#define AUDIO_AAC_FORMAT_ADIF		0x0003
+
+#define AUDIO_AAC_OBJECT_LC            	0x0002
+#define AUDIO_AAC_OBJECT_LTP		0x0004
+#define AUDIO_AAC_OBJECT_ERLC  		0x0011
+#define AUDIO_AAC_OBJECT_BSAC  		0x0016
+
+#define AUDIO_AAC_SEC_DATA_RES_ON       0x0001
+#define AUDIO_AAC_SEC_DATA_RES_OFF      0x0000
+
+#define AUDIO_AAC_SCA_DATA_RES_ON       0x0001
+#define AUDIO_AAC_SCA_DATA_RES_OFF      0x0000
+
+#define AUDIO_AAC_SPEC_DATA_RES_ON      0x0001
+#define AUDIO_AAC_SPEC_DATA_RES_OFF     0x0000
+
+#define AUDIO_AAC_SBR_ON_FLAG_ON	0x0001
+#define AUDIO_AAC_SBR_ON_FLAG_OFF	0x0000
+
+#define AUDIO_AAC_SBR_PS_ON_FLAG_ON	0x0001
+#define AUDIO_AAC_SBR_PS_ON_FLAG_OFF	0x0000
+
+/* Primary channel on both left and right channels */
+#define AUDIO_AAC_DUAL_MONO_PL_PR  0
+/* Secondary channel on both left and right channels */
+#define AUDIO_AAC_DUAL_MONO_SL_SR  1
+/* Primary channel on right channel and 2nd on left channel */
+#define AUDIO_AAC_DUAL_MONO_SL_PR  2
+/* 2nd channel on right channel and primary on left channel */
+#define AUDIO_AAC_DUAL_MONO_PL_SR  3
+
+struct msm_audio_aac_config {
+	signed short format;
+	unsigned short audio_object;
+	unsigned short ep_config;	/* 0 ~ 3 useful only obj = ERLC */
+	unsigned short aac_section_data_resilience_flag;
+	unsigned short aac_scalefactor_data_resilience_flag;
+	unsigned short aac_spectral_data_resilience_flag;
+	unsigned short sbr_on_flag;
+	unsigned short sbr_ps_on_flag;
+	unsigned short dual_mono_mode;
+	unsigned short channel_configuration;
+	unsigned short sample_rate;
+};
+
+struct msm_audio_aac_enc_config {
+	uint32_t channels;
+	uint32_t sample_rate;
+	uint32_t bit_rate;
+	uint32_t stream_format;
+};
+
+#endif /* _UAPI_MSM_AUDIO_AAC_H */
diff --git a/include/uapi/linux/msm_audio_ac3.h b/include/uapi/linux/msm_audio_ac3.h
new file mode 100644
index 000000000000..e314f3a48351
--- /dev/null
+++ b/include/uapi/linux/msm_audio_ac3.h
@@ -0,0 +1,41 @@
+#ifndef _UAPI_MSM_AUDIO_AC3_H
+#define _UAPI_MSM_AUDIO_AC3_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_SET_AC3_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+0), unsigned)
+#define AUDIO_GET_AC3_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+1), unsigned)
+
+#define AUDAC3_DEF_WORDSIZE 0
+#define AUDAC3_DEF_USER_DOWNMIX_FLAG 0x0
+#define AUDAC3_DEF_USER_KARAOKE_FLAG 0x0
+#define AUDAC3_DEF_ERROR_CONCEALMENT 0
+#define AUDAC3_DEF_MAX_REPEAT_COUNT  0
+
+struct msm_audio_ac3_config {
+	unsigned short		numChans;
+	unsigned short		wordSize;
+	unsigned short		kCapableMode;
+	unsigned short		compMode;
+	unsigned short		outLfeOn;
+	unsigned short		outputMode;
+	unsigned short		stereoMode;
+	unsigned short		dualMonoMode;
+	unsigned short		fsCod;
+	unsigned short		pcmScaleFac;
+	unsigned short		dynRngScaleHi;
+	unsigned short		dynRngScaleLow;
+	unsigned short		user_downmix_flag;
+	unsigned short		user_karaoke_flag;
+	unsigned short		dm_address_high;
+	unsigned short		dm_address_low;
+	unsigned short		ko_address_high;
+	unsigned short		ko_address_low;
+	unsigned short		error_concealment;
+	unsigned short		max_rep_count;
+	unsigned short		channel_routing_mode[6];
+};
+
+#endif /* _UAPI_MSM_AUDIO_AC3_H */
diff --git a/include/uapi/linux/msm_audio_alac.h b/include/uapi/linux/msm_audio_alac.h
new file mode 100644
index 000000000000..5476e96d06fc
--- /dev/null
+++ b/include/uapi/linux/msm_audio_alac.h
@@ -0,0 +1,24 @@
+#ifndef _UAPI_MSM_AUDIO_ALAC_H
+#define _UAPI_MSM_AUDIO_ALAC_H
+
+#define AUDIO_GET_ALAC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+0), struct msm_audio_alac_config)
+#define AUDIO_SET_ALAC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+1), struct msm_audio_alac_config)
+
+struct msm_audio_alac_config {
+	uint32_t frameLength;
+	uint8_t compatVersion;
+	uint8_t bitDepth;
+	uint8_t pb; /* currently unused */
+	uint8_t mb; /* currently unused */
+	uint8_t kb; /* currently unused */
+	uint8_t channelCount;
+	uint16_t maxRun; /* currently unused */
+	uint32_t maxSize;
+	uint32_t averageBitRate;
+	uint32_t sampleRate;
+	uint32_t channelLayout;
+};
+
+#endif /* _UAPI_MSM_AUDIO_ALAC_H */
diff --git a/include/uapi/linux/msm_audio_amrnb.h b/include/uapi/linux/msm_audio_amrnb.h
new file mode 100644
index 000000000000..f995069b9e38
--- /dev/null
+++ b/include/uapi/linux/msm_audio_amrnb.h
@@ -0,0 +1,33 @@
+#ifndef _UAPI_MSM_AUDIO_AMRNB_H
+#define _UAPI_MSM_AUDIO_AMRNB_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_GET_AMRNB_ENC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+0), unsigned)
+#define AUDIO_SET_AMRNB_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+1), unsigned)
+#define AUDIO_GET_AMRNB_ENC_CONFIG_V2  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+2), \
+	struct msm_audio_amrnb_enc_config_v2)
+#define AUDIO_SET_AMRNB_ENC_CONFIG_V2  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+3), \
+	struct msm_audio_amrnb_enc_config_v2)
+
+struct msm_audio_amrnb_enc_config {
+	unsigned short voicememoencweight1;
+	unsigned short voicememoencweight2;
+	unsigned short voicememoencweight3;
+	unsigned short voicememoencweight4;
+	unsigned short dtx_mode_enable; /* 0xFFFF - enable, 0- disable */
+	unsigned short test_mode_enable; /* 0xFFFF - enable, 0- disable */
+	unsigned short enc_mode; /* 0-MR475,1-MR515,2-MR59,3-MR67,4-MR74
+				5-MR795, 6- MR102, 7- MR122(default) */
+};
+
+struct msm_audio_amrnb_enc_config_v2 {
+	uint32_t band_mode;
+	uint32_t dtx_enable;
+	uint32_t frame_format;
+};
+#endif /* _UAPI_MSM_AUDIO_AMRNB_H */
diff --git a/include/uapi/linux/msm_audio_amrwb.h b/include/uapi/linux/msm_audio_amrwb.h
new file mode 100644
index 000000000000..51240389988f
--- /dev/null
+++ b/include/uapi/linux/msm_audio_amrwb.h
@@ -0,0 +1,18 @@
+#ifndef _UAPI_MSM_AUDIO_AMRWB_H
+#define _UAPI_MSM_AUDIO_AMRWB_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_GET_AMRWB_ENC_CONFIG _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+0), \
+	struct msm_audio_amrwb_enc_config)
+#define AUDIO_SET_AMRWB_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+1), \
+	struct msm_audio_amrwb_enc_config)
+
+struct msm_audio_amrwb_enc_config {
+	uint32_t band_mode;
+	uint32_t dtx_enable;
+	uint32_t frame_format;
+};
+#endif /* _UAPI_MSM_AUDIO_AMRWB_H */
diff --git a/include/uapi/linux/msm_audio_amrwbplus.h b/include/uapi/linux/msm_audio_amrwbplus.h
new file mode 100644
index 000000000000..ba2d06e99aa1
--- /dev/null
+++ b/include/uapi/linux/msm_audio_amrwbplus.h
@@ -0,0 +1,18 @@
+#ifndef _UAPI_MSM_AUDIO_AMR_WB_PLUS_H
+#define _UAPI_MSM_AUDIO_AMR_WB_PLUS_H
+
+#define AUDIO_GET_AMRWBPLUS_CONFIG_V2  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+2), struct msm_audio_amrwbplus_config_v2)
+#define AUDIO_SET_AMRWBPLUS_CONFIG_V2  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+3), struct msm_audio_amrwbplus_config_v2)
+
+struct msm_audio_amrwbplus_config_v2 {
+	unsigned int size_bytes;
+	unsigned int version;
+	unsigned int num_channels;
+	unsigned int amr_band_mode;
+	unsigned int amr_dtx_mode;
+	unsigned int amr_frame_fmt;
+	unsigned int amr_lsf_idx;
+};
+#endif /* _UAPI_MSM_AUDIO_AMR_WB_PLUS_H */
diff --git a/include/uapi/linux/msm_audio_ape.h b/include/uapi/linux/msm_audio_ape.h
new file mode 100644
index 000000000000..397cdbf09a54
--- /dev/null
+++ b/include/uapi/linux/msm_audio_ape.h
@@ -0,0 +1,25 @@
+/*The following structure has been taken
+from Monkey's Audio SDK with permission*/
+
+#ifndef _UAPI_MSM_AUDIO_APE_H
+#define _UAPI_MSM_AUDIO_APE_H
+
+#define AUDIO_GET_APE_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+0), struct msm_audio_ape_config)
+#define AUDIO_SET_APE_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+1), struct msm_audio_ape_config)
+
+struct msm_audio_ape_config {
+	uint16_t compatibleVersion;
+	uint16_t compressionLevel;
+	uint32_t formatFlags;
+	uint32_t blocksPerFrame;
+	uint32_t finalFrameBlocks;
+	uint32_t totalFrames;
+	uint16_t bitsPerSample;
+	uint16_t numChannels;
+	uint32_t sampleRate;
+	uint32_t seekTablePresent;
+};
+
+#endif /* _UAPI_MSM_AUDIO_APE_H */
diff --git a/include/uapi/linux/msm_audio_calibration.h b/include/uapi/linux/msm_audio_calibration.h
new file mode 100644
index 000000000000..eb6c692b394c
--- /dev/null
+++ b/include/uapi/linux/msm_audio_calibration.h
@@ -0,0 +1,607 @@
+#ifndef _UAPI_MSM_AUDIO_CALIBRATION_H
+#define _UAPI_MSM_AUDIO_CALIBRATION_H
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define CAL_IOCTL_MAGIC 'a'
+
+#define AUDIO_ALLOCATE_CALIBRATION	_IOWR(CAL_IOCTL_MAGIC, \
+							200, void *)
+#define AUDIO_DEALLOCATE_CALIBRATION	_IOWR(CAL_IOCTL_MAGIC, \
+							201, void *)
+#define AUDIO_PREPARE_CALIBRATION	_IOWR(CAL_IOCTL_MAGIC, \
+							202, void *)
+#define AUDIO_SET_CALIBRATION		_IOWR(CAL_IOCTL_MAGIC, \
+							203, void *)
+#define AUDIO_GET_CALIBRATION		_IOWR(CAL_IOCTL_MAGIC, \
+							204, void *)
+#define AUDIO_POST_CALIBRATION		_IOWR(CAL_IOCTL_MAGIC, \
+							205, void *)
+
+/* For Real-Time Audio Calibration */
+#define AUDIO_GET_RTAC_ADM_INFO		_IOR(CAL_IOCTL_MAGIC, \
+							207, void *)
+#define AUDIO_GET_RTAC_VOICE_INFO	_IOR(CAL_IOCTL_MAGIC, \
+							208, void *)
+#define AUDIO_GET_RTAC_ADM_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							209, void *)
+#define AUDIO_SET_RTAC_ADM_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							210, void *)
+#define AUDIO_GET_RTAC_ASM_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							211, void *)
+#define AUDIO_SET_RTAC_ASM_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							212, void *)
+#define AUDIO_GET_RTAC_CVS_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							213, void *)
+#define AUDIO_SET_RTAC_CVS_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							214, void *)
+#define AUDIO_GET_RTAC_CVP_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							215, void *)
+#define AUDIO_SET_RTAC_CVP_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							216, void *)
+#define AUDIO_GET_RTAC_AFE_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							217, void *)
+#define AUDIO_SET_RTAC_AFE_CAL		_IOWR(CAL_IOCTL_MAGIC, \
+							218, void *)
+enum {
+	CVP_VOC_RX_TOPOLOGY_CAL_TYPE = 0,
+	CVP_VOC_TX_TOPOLOGY_CAL_TYPE,
+	CVP_VOCPROC_STATIC_CAL_TYPE,
+	CVP_VOCPROC_DYNAMIC_CAL_TYPE,
+	CVS_VOCSTRM_STATIC_CAL_TYPE,
+	CVP_VOCDEV_CFG_CAL_TYPE,
+	CVP_VOCPROC_STATIC_COL_CAL_TYPE,
+	CVP_VOCPROC_DYNAMIC_COL_CAL_TYPE,
+	CVS_VOCSTRM_STATIC_COL_CAL_TYPE,
+
+	ADM_TOPOLOGY_CAL_TYPE,
+	ADM_CUST_TOPOLOGY_CAL_TYPE,
+	ADM_AUDPROC_CAL_TYPE,
+	ADM_AUDVOL_CAL_TYPE,
+
+	ASM_TOPOLOGY_CAL_TYPE,
+	ASM_CUST_TOPOLOGY_CAL_TYPE,
+	ASM_AUDSTRM_CAL_TYPE,
+
+	AFE_COMMON_RX_CAL_TYPE,
+	AFE_COMMON_TX_CAL_TYPE,
+	AFE_ANC_CAL_TYPE,
+	AFE_AANC_CAL_TYPE,
+	AFE_FB_SPKR_PROT_CAL_TYPE,
+	AFE_HW_DELAY_CAL_TYPE,
+	AFE_SIDETONE_CAL_TYPE,
+	AFE_TOPOLOGY_CAL_TYPE,
+	AFE_CUST_TOPOLOGY_CAL_TYPE,
+
+	LSM_CUST_TOPOLOGY_CAL_TYPE,
+	LSM_TOPOLOGY_CAL_TYPE,
+	LSM_CAL_TYPE,
+
+	ADM_RTAC_INFO_CAL_TYPE,
+	VOICE_RTAC_INFO_CAL_TYPE,
+	ADM_RTAC_APR_CAL_TYPE,
+	ASM_RTAC_APR_CAL_TYPE,
+	VOICE_RTAC_APR_CAL_TYPE,
+
+	MAD_CAL_TYPE,
+	ULP_AFE_CAL_TYPE,
+	ULP_LSM_CAL_TYPE,
+
+	DTS_EAGLE_CAL_TYPE,
+	AUDIO_CORE_METAINFO_CAL_TYPE,
+	SRS_TRUMEDIA_CAL_TYPE,
+
+	CORE_CUSTOM_TOPOLOGIES_CAL_TYPE,
+	ADM_RTAC_AUDVOL_CAL_TYPE,
+
+	ULP_LSM_TOPOLOGY_ID_CAL_TYPE,
+	MAX_CAL_TYPES,
+};
+
+enum {
+	VERSION_0_0,
+};
+
+enum {
+	PER_VOCODER_CAL_BIT_MASK = 0x10000,
+};
+
+#define MAX_IOCTL_CMD_SIZE	512
+
+/* common structures */
+
+struct audio_cal_header {
+	int32_t		data_size;
+	int32_t		version;
+	int32_t		cal_type;
+	int32_t		cal_type_size;
+};
+
+struct audio_cal_type_header {
+	int32_t		version;
+	int32_t		buffer_number;
+};
+
+struct audio_cal_data {
+	/* Size of cal data at mem_handle allocation or at vaddr */
+	int32_t		cal_size;
+	/* If mem_handle if shared memory is used*/
+	int32_t		mem_handle;
+	/* size of virtual memory if shared memory not used */
+};
+
+
+/* AUDIO_ALLOCATE_CALIBRATION */
+struct audio_cal_type_alloc {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+};
+
+struct audio_cal_alloc {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_alloc	cal_type;
+};
+
+
+/* AUDIO_DEALLOCATE_CALIBRATION */
+struct audio_cal_type_dealloc {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+};
+
+struct audio_cal_dealloc {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_dealloc	cal_type;
+};
+
+
+/* AUDIO_PREPARE_CALIBRATION */
+struct audio_cal_type_prepare {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+};
+
+struct audio_cal_prepare {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_prepare	cal_type;
+};
+
+
+/* AUDIO_POST_CALIBRATION */
+struct audio_cal_type_post {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+};
+
+struct audio_cal_post {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_post	cal_type;
+};
+
+/*AUDIO_CORE_META_INFO */
+
+struct audio_cal_info_metainfo {
+	uint32_t nKey;
+};
+
+/* Cal info types */
+enum {
+	RX_DEVICE,
+	TX_DEVICE,
+	MAX_PATH_TYPE
+};
+
+struct audio_cal_info_adm_top {
+	int32_t		topology;
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		app_type;
+	int32_t		sample_rate;
+};
+
+struct audio_cal_info_audproc {
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		app_type;
+	int32_t		sample_rate;
+};
+
+struct audio_cal_info_audvol {
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		app_type;
+	int32_t		vol_index;
+};
+
+struct audio_cal_info_afe {
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		sample_rate;
+};
+
+struct audio_cal_info_afe_top {
+	int32_t		topology;
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		sample_rate;
+};
+
+struct audio_cal_info_asm_top {
+	int32_t		topology;
+	int32_t		app_type;
+};
+
+struct audio_cal_info_audstrm {
+	int32_t		app_type;
+};
+
+struct audio_cal_info_aanc {
+	int32_t		acdb_id;
+};
+
+#define MAX_HW_DELAY_ENTRIES	25
+
+struct audio_cal_hw_delay_entry {
+	uint32_t sample_rate;
+	uint32_t delay_usec;
+};
+
+struct audio_cal_hw_delay_data {
+	uint32_t				num_entries;
+	struct audio_cal_hw_delay_entry		entry[MAX_HW_DELAY_ENTRIES];
+};
+
+struct audio_cal_info_hw_delay {
+	int32_t					acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t					path;
+	int32_t					property_type;
+	struct audio_cal_hw_delay_data		data;
+};
+
+enum msm_spkr_prot_states {
+	MSM_SPKR_PROT_CALIBRATED,
+	MSM_SPKR_PROT_CALIBRATION_IN_PROGRESS,
+	MSM_SPKR_PROT_DISABLED,
+	MSM_SPKR_PROT_NOT_CALIBRATED,
+	MSM_SPKR_PROT_PRE_CALIBRATED,
+};
+
+enum msm_spkr_count {
+	SP_V2_SPKR_1,
+	SP_V2_SPKR_2,
+	SP_V2_NUM_MAX_SPKRS
+};
+
+struct audio_cal_info_spk_prot_cfg {
+	int32_t		r0[SP_V2_NUM_MAX_SPKRS];
+	int32_t		t0[SP_V2_NUM_MAX_SPKRS];
+	uint32_t	quick_calib_flag;
+	uint32_t	mode; /*0 - Start spk prot
+	1 - Start calib
+	2 - Disable spk prot*/
+};
+
+struct audio_cal_info_msm_spk_prot_status {
+	int32_t		r0[SP_V2_NUM_MAX_SPKRS];
+	int32_t		status;
+};
+
+struct audio_cal_info_sidetone {
+	uint16_t	enable;
+	uint16_t	gain;
+	int32_t		tx_acdb_id;
+	int32_t		rx_acdb_id;
+	int32_t		mid;
+	int32_t		pid;
+};
+
+struct audio_cal_info_lsm_top {
+	int32_t		topology;
+	int32_t		acdb_id;
+	int32_t		app_type;
+};
+
+
+struct audio_cal_info_lsm {
+	int32_t		acdb_id;
+	/* RX_DEVICE or TX_DEVICE */
+	int32_t		path;
+	int32_t		app_type;
+};
+
+struct audio_cal_info_voc_top {
+	int32_t		topology;
+	int32_t		acdb_id;
+};
+
+struct audio_cal_info_vocproc {
+	int32_t		tx_acdb_id;
+	int32_t		rx_acdb_id;
+	int32_t		tx_sample_rate;
+	int32_t		rx_sample_rate;
+};
+
+enum {
+	DEFAULT_FEATURE_SET,
+	VOL_BOOST_FEATURE_SET,
+};
+
+struct audio_cal_info_vocvol {
+	int32_t		tx_acdb_id;
+	int32_t		rx_acdb_id;
+	/* DEFUALT_ or VOL_BOOST_FEATURE_SET */
+	int32_t		feature_set;
+};
+
+struct audio_cal_info_vocdev_cfg {
+	int32_t		tx_acdb_id;
+	int32_t		rx_acdb_id;
+};
+
+#define MAX_VOICE_COLUMNS	20
+
+union audio_cal_col_na {
+	uint8_t		val8;
+	uint16_t	val16;
+	uint32_t	val32;
+	uint64_t	val64;
+} __packed;
+
+struct audio_cal_col {
+	uint32_t		id;
+	uint32_t		type;
+	union audio_cal_col_na	na_value;
+} __packed;
+
+struct audio_cal_col_data {
+	uint32_t		num_columns;
+	struct audio_cal_col	column[MAX_VOICE_COLUMNS];
+} __packed;
+
+struct audio_cal_info_voc_col {
+	int32_t				table_id;
+	int32_t				tx_acdb_id;
+	int32_t				rx_acdb_id;
+	struct audio_cal_col_data	data;
+};
+
+/* AUDIO_SET_CALIBRATION & */
+struct audio_cal_type_basic {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+};
+
+struct audio_cal_basic {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_basic	cal_type;
+};
+
+struct audio_cal_type_adm_top {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_adm_top	cal_info;
+};
+
+struct audio_cal_adm_top {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_adm_top	cal_type;
+};
+
+struct audio_cal_type_metainfo {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_metainfo	cal_info;
+};
+
+struct audio_core_metainfo {
+	struct audio_cal_header	  hdr;
+	struct audio_cal_type_metainfo cal_type;
+};
+
+struct audio_cal_type_audproc {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_audproc	cal_info;
+};
+
+struct audio_cal_audproc {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_audproc	cal_type;
+};
+
+struct audio_cal_type_audvol {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_audvol	cal_info;
+};
+
+struct audio_cal_audvol {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_audvol	cal_type;
+};
+
+struct audio_cal_type_asm_top {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_asm_top	cal_info;
+};
+
+struct audio_cal_asm_top {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_asm_top	cal_type;
+};
+
+struct audio_cal_type_audstrm {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_audstrm	cal_info;
+};
+
+struct audio_cal_audstrm {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_audstrm	cal_type;
+};
+
+struct audio_cal_type_afe {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_afe	cal_info;
+};
+
+struct audio_cal_afe {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_afe	cal_type;
+};
+
+struct audio_cal_type_afe_top {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_afe_top	cal_info;
+};
+
+struct audio_cal_afe_top {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_afe_top	cal_type;
+};
+
+struct audio_cal_type_aanc {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_aanc	cal_info;
+};
+
+struct audio_cal_aanc {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_aanc	cal_type;
+};
+
+struct audio_cal_type_fb_spk_prot_cfg {
+	struct audio_cal_type_header		cal_hdr;
+	struct audio_cal_data			cal_data;
+	struct audio_cal_info_spk_prot_cfg	cal_info;
+};
+
+struct audio_cal_fb_spk_prot_cfg {
+	struct audio_cal_header			hdr;
+	struct audio_cal_type_fb_spk_prot_cfg	cal_type;
+};
+
+struct audio_cal_type_hw_delay {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_hw_delay	cal_info;
+};
+
+struct audio_cal_hw_delay {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_hw_delay	cal_type;
+};
+
+struct audio_cal_type_sidetone {
+	struct audio_cal_type_header		cal_hdr;
+	struct audio_cal_data			cal_data;
+	struct audio_cal_info_sidetone		cal_info;
+};
+
+struct audio_cal_sidetone {
+	struct audio_cal_header			hdr;
+	struct audio_cal_type_sidetone		cal_type;
+};
+
+struct audio_cal_type_lsm_top {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_lsm_top	cal_info;
+};
+
+struct audio_cal_lsm_top {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_lsm_top	cal_type;
+};
+
+struct audio_cal_type_lsm {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_lsm	cal_info;
+};
+
+struct audio_cal_lsm {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_lsm	cal_type;
+};
+
+struct audio_cal_type_voc_top {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_voc_top	cal_info;
+};
+
+struct audio_cal_voc_top {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_voc_top	cal_type;
+};
+
+struct audio_cal_type_vocproc {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_vocproc	cal_info;
+};
+
+struct audio_cal_vocproc {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_vocproc	cal_type;
+};
+
+struct audio_cal_type_vocvol {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_vocvol	cal_info;
+};
+
+struct audio_cal_vocvol {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_vocvol	cal_type;
+};
+
+struct audio_cal_type_vocdev_cfg {
+	struct audio_cal_type_header		cal_hdr;
+	struct audio_cal_data			cal_data;
+	struct audio_cal_info_vocdev_cfg	cal_info;
+};
+
+struct audio_cal_vocdev_cfg {
+	struct audio_cal_header			hdr;
+	struct audio_cal_type_vocdev_cfg	cal_type;
+};
+
+struct audio_cal_type_voc_col {
+	struct audio_cal_type_header	cal_hdr;
+	struct audio_cal_data		cal_data;
+	struct audio_cal_info_voc_col	cal_info;
+};
+
+struct audio_cal_voc_col {
+	struct audio_cal_header		hdr;
+	struct audio_cal_type_voc_col	cal_type;
+};
+
+/* AUDIO_GET_CALIBRATION */
+struct audio_cal_type_fb_spk_prot_status {
+	struct audio_cal_type_header			cal_hdr;
+	struct audio_cal_data				cal_data;
+	struct audio_cal_info_msm_spk_prot_status	cal_info;
+};
+
+struct audio_cal_fb_spk_prot_status {
+	struct audio_cal_header				hdr;
+	struct audio_cal_type_fb_spk_prot_status	cal_type;
+};
+
+#endif /* _UAPI_MSM_AUDIO_CALIBRATION_H */
diff --git a/include/uapi/linux/msm_audio_mvs.h b/include/uapi/linux/msm_audio_mvs.h
new file mode 100644
index 000000000000..5a71b26c8097
--- /dev/null
+++ b/include/uapi/linux/msm_audio_mvs.h
@@ -0,0 +1,154 @@
+#ifndef _UAPI_MSM_AUDIO_MVS_H
+#define _UAPI_MSM_AUDIO_MVS_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_GET_MVS_CONFIG _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM + 0), unsigned)
+#define AUDIO_SET_MVS_CONFIG _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM + 1), unsigned)
+
+/* MVS modes */
+#define MVS_MODE_IS733 0x1 /*QCELP 13K*/
+#define MVS_MODE_IS127 0x2 /*EVRC-8k*/
+#define MVS_MODE_4GV_NB 0x3 /*EVRC-B*/
+#define MVS_MODE_4GV_WB 0x4 /*EVRC-WB*/
+#define MVS_MODE_AMR 0x5
+#define MVS_MODE_EFR 0x6
+#define MVS_MODE_FR 0x7
+#define MVS_MODE_HR 0x8
+#define MVS_MODE_LINEAR_PCM 0x9
+#define MVS_MODE_G711 0xA
+#define MVS_MODE_PCM 0xC
+#define MVS_MODE_AMR_WB 0xD
+#define MVS_MODE_G729A 0xE
+#define MVS_MODE_G711A 0xF
+#define MVS_MODE_G722 0x10
+#define MVS_MODE_PCM_WB 0x12
+
+enum msm_audio_amr_mode {
+	MVS_AMR_MODE_0475, /* AMR 4.75 kbps */
+	MVS_AMR_MODE_0515, /* AMR 5.15 kbps */
+	MVS_AMR_MODE_0590, /* AMR 5.90 kbps */
+	MVS_AMR_MODE_0670, /* AMR 6.70 kbps */
+	MVS_AMR_MODE_0740, /* AMR 7.40 kbps */
+	MVS_AMR_MODE_0795, /* AMR 7.95 kbps */
+	MVS_AMR_MODE_1020, /* AMR 10.20 kbps */
+	MVS_AMR_MODE_1220, /* AMR 12.20 kbps */
+	MVS_AMR_MODE_0660, /* AMR-WB 6.60 kbps */
+	MVS_AMR_MODE_0885, /* AMR-WB 8.85 kbps */
+	MVS_AMR_MODE_1265, /* AMR-WB 12.65 kbps */
+	MVS_AMR_MODE_1425, /* AMR-WB 14.25 kbps */
+	MVS_AMR_MODE_1585, /* AMR-WB 15.85 kbps */
+	MVS_AMR_MODE_1825, /* AMR-WB 18.25 kbps */
+	MVS_AMR_MODE_1985, /* AMR-WB 19.85 kbps */
+	MVS_AMR_MODE_2305, /* AMR-WB 23.05 kbps */
+	MVS_AMR_MODE_2385, /* AMR-WB 23.85 kbps */
+	MVS_AMR_MODE_UNDEF
+};
+
+/*The MVS VOC rate type is used to identify the rate of QCELP 13K(IS733),
+EVRC(IS127), 4GV, or 4GV-WB frame.*/
+enum msm_audio_voc_rate {
+		MVS_VOC_0_RATE, /* Blank frame */
+		MVS_VOC_8_RATE, /* 1/8 rate    */
+		MVS_VOC_4_RATE, /* 1/4 rate    */
+		MVS_VOC_2_RATE, /* 1/2 rate    */
+		MVS_VOC_1_RATE,/* Full rate   */
+		MVS_VOC_ERASURE, /* erasure frame */
+		MVS_VOC_RATE_MAX,
+		MVS_VOC_RATE_UNDEF = MVS_VOC_RATE_MAX
+};
+
+enum msm_audio_amr_frame_type {
+	MVS_AMR_SPEECH_GOOD,	      /* Good speech frame              */
+	MVS_AMR_SPEECH_DEGRADED,      /* Speech degraded                */
+	MVS_AMR_ONSET,		      /* Onset                          */
+	MVS_AMR_SPEECH_BAD,	      /* Corrupt speech frame (bad CRC) */
+	MVS_AMR_SID_FIRST,	      /* First silence descriptor       */
+	MVS_AMR_SID_UPDATE,	      /* Comfort noise frame            */
+	MVS_AMR_SID_BAD,	      /* Corrupt SID frame (bad CRC)    */
+	MVS_AMR_NO_DATA,	      /* Nothing to transmit            */
+	MVS_AMR_SPEECH_LOST	      /* Downlink speech lost           */
+};
+
+enum msm_audio_g711a_mode {
+	MVS_G711A_MODE_MULAW,
+	MVS_G711A_MODE_ALAW
+};
+
+enum msm_audio_g711_mode {
+	MVS_G711_MODE_MULAW,
+	MVS_G711_MODE_ALAW
+};
+
+enum mvs_g722_mode_type {
+	MVS_G722_MODE_01,
+	MVS_G722_MODE_02,
+	MVS_G722_MODE_03,
+	MVS_G722_MODE_MAX,
+	MVS_G722_MODE_UNDEF
+};
+
+enum msm_audio_g711a_frame_type {
+	MVS_G711A_SPEECH_GOOD,
+	MVS_G711A_SID,
+	MVS_G711A_NO_DATA,
+	MVS_G711A_ERASURE
+};
+
+enum msm_audio_g729a_frame_type {
+	MVS_G729A_NO_DATA,
+	MVS_G729A_SPEECH_GOOD,
+	MVS_G729A_SID,
+	MVS_G729A_ERASURE
+};
+
+struct min_max_rate {
+	uint32_t min_rate;
+	uint32_t max_rate;
+};
+
+struct msm_audio_mvs_config {
+	uint32_t mvs_mode;
+	uint32_t rate_type;
+	struct min_max_rate min_max_rate;
+	uint32_t dtx_mode;
+};
+
+#define MVS_MAX_VOC_PKT_SIZE 640
+
+struct gsm_header {
+	uint8_t bfi;
+	uint8_t sid;
+	uint8_t taf;
+	uint8_t ufi;
+};
+
+struct q6_msm_audio_mvs_frame {
+	union {
+	uint32_t frame_type;
+	uint32_t packet_rate;
+	struct gsm_header gsm_frame_type;
+	} header;
+	uint32_t len;
+	uint8_t voc_pkt[MVS_MAX_VOC_PKT_SIZE];
+
+};
+
+struct msm_audio_mvs_frame {
+	uint32_t frame_type;
+	uint32_t len;
+	uint8_t voc_pkt[MVS_MAX_VOC_PKT_SIZE];
+
+};
+
+#define Q5V2_MVS_MAX_VOC_PKT_SIZE 320
+
+struct q5v2_msm_audio_mvs_frame {
+	uint32_t frame_type;
+	uint32_t len;
+	uint8_t voc_pkt[Q5V2_MVS_MAX_VOC_PKT_SIZE];
+
+};
+#endif /* _UAPI_MSM_AUDIO_MVS_H */
diff --git a/include/uapi/linux/msm_audio_qcp.h b/include/uapi/linux/msm_audio_qcp.h
new file mode 100644
index 000000000000..fdb234e91acf
--- /dev/null
+++ b/include/uapi/linux/msm_audio_qcp.h
@@ -0,0 +1,37 @@
+#ifndef _UAPI_MSM_AUDIO_QCP_H
+#define _UAPI_MSM_AUDIO_QCP_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_SET_QCELP_ENC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	0, struct msm_audio_qcelp_enc_config)
+
+#define AUDIO_GET_QCELP_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	1, struct msm_audio_qcelp_enc_config)
+
+#define AUDIO_SET_EVRC_ENC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	2, struct msm_audio_evrc_enc_config)
+
+#define AUDIO_GET_EVRC_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	3, struct msm_audio_evrc_enc_config)
+
+#define CDMA_RATE_BLANK		0x00
+#define CDMA_RATE_EIGHTH	0x01
+#define CDMA_RATE_QUARTER	0x02
+#define CDMA_RATE_HALF		0x03
+#define CDMA_RATE_FULL		0x04
+#define CDMA_RATE_ERASURE	0x05
+
+struct msm_audio_qcelp_enc_config {
+	uint32_t cdma_rate;
+	uint32_t min_bit_rate;
+	uint32_t max_bit_rate;
+};
+
+struct msm_audio_evrc_enc_config {
+	uint32_t cdma_rate;
+	uint32_t min_bit_rate;
+	uint32_t max_bit_rate;
+};
+
+#endif /* _UAPI_MSM_AUDIO_QCP_H */
diff --git a/include/uapi/linux/msm_audio_sbc.h b/include/uapi/linux/msm_audio_sbc.h
new file mode 100644
index 000000000000..4a1a6b76d37a
--- /dev/null
+++ b/include/uapi/linux/msm_audio_sbc.h
@@ -0,0 +1,36 @@
+#ifndef _UAPI_MSM_AUDIO_SBC_H
+#define _UAPI_MSM_AUDIO_SBC_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_SET_SBC_ENC_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+  (AUDIO_MAX_COMMON_IOCTL_NUM+0), struct msm_audio_sbc_enc_config)
+
+#define AUDIO_GET_SBC_ENC_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+  (AUDIO_MAX_COMMON_IOCTL_NUM+1), struct msm_audio_sbc_enc_config)
+
+#define AUDIO_SBC_BA_LOUDNESS		0x0
+#define AUDIO_SBC_BA_SNR		0x1
+
+#define AUDIO_SBC_MODE_MONO		0x0
+#define AUDIO_SBC_MODE_DUAL		0x1
+#define AUDIO_SBC_MODE_STEREO		0x2
+#define AUDIO_SBC_MODE_JSTEREO		0x3
+
+#define AUDIO_SBC_BANDS_8		0x1
+
+#define AUDIO_SBC_BLOCKS_4		0x0
+#define AUDIO_SBC_BLOCKS_8		0x1
+#define AUDIO_SBC_BLOCKS_12		0x2
+#define AUDIO_SBC_BLOCKS_16		0x3
+
+struct msm_audio_sbc_enc_config {
+	uint32_t channels;
+	uint32_t sample_rate;
+	uint32_t bit_allocation;
+	uint32_t number_of_subbands;
+	uint32_t number_of_blocks;
+	uint32_t bit_rate;
+	uint32_t mode;
+};
+#endif /* _UAPI_MSM_AUDIO_SBC_H */
diff --git a/include/uapi/linux/msm_audio_voicememo.h b/include/uapi/linux/msm_audio_voicememo.h
new file mode 100644
index 000000000000..48690d08ea7a
--- /dev/null
+++ b/include/uapi/linux/msm_audio_voicememo.h
@@ -0,0 +1,66 @@
+#ifndef _UAPI_MSM_AUDIO_VOICEMEMO_H
+#define _UAPI_MSM_AUDIO_VOICEMEMO_H
+
+#include <linux/msm_audio.h>
+
+#define AUDIO_GET_VOICEMEMO_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+0), unsigned)
+#define AUDIO_SET_VOICEMEMO_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	(AUDIO_MAX_COMMON_IOCTL_NUM+1), unsigned)
+
+/* rec_type */
+enum rpc_voc_rec_dir_type {
+	RPC_VOC_REC_NONE,
+	RPC_VOC_REC_FORWARD,
+	RPC_VOC_REC_REVERSE,
+	RPC_VOC_REC_BOTH,
+	RPC_VOC_MAX_REC_TYPE
+};
+
+/* capability */
+enum rpc_voc_capability_type {
+	RPC_VOC_CAP_IS733 = 4,
+	RPC_VOC_CAP_IS127 = 8,
+	RPC_VOC_CAP_AMR = 64,
+	RPC_VOC_CAP_32BIT_DUMMY = 2147483647
+};
+
+/* Rate */
+enum rpc_voc_rate_type {
+	RPC_VOC_0_RATE = 0,
+	RPC_VOC_8_RATE,
+	RPC_VOC_4_RATE,
+	RPC_VOC_2_RATE,
+	RPC_VOC_1_RATE,
+	RPC_VOC_ERASURE,
+	RPC_VOC_ERR_RATE,
+	RPC_VOC_AMR_RATE_475 = 0,
+	RPC_VOC_AMR_RATE_515 = 1,
+	RPC_VOC_AMR_RATE_590 = 2,
+	RPC_VOC_AMR_RATE_670 = 3,
+	RPC_VOC_AMR_RATE_740 = 4,
+	RPC_VOC_AMR_RATE_795 = 5,
+	RPC_VOC_AMR_RATE_1020 = 6,
+	RPC_VOC_AMR_RATE_1220 = 7,
+};
+
+/* frame_format */
+enum rpc_voc_pb_len_rate_var_type {
+	RPC_VOC_PB_NATIVE_QCP = 3,
+	RPC_VOC_PB_AMR,
+	RPC_VOC_PB_EVB
+};
+
+struct msm_audio_voicememo_config {
+	uint32_t rec_type;
+	uint32_t rec_interval_ms;
+	uint32_t auto_stop_ms;
+	uint32_t capability;
+	uint32_t max_rate;
+	uint32_t min_rate;
+	uint32_t frame_format;
+	uint32_t dtx_enable;
+	uint32_t data_req_ms;
+};
+
+#endif /* _UAPI_MSM_AUDIO_VOICEMEMO_H */
diff --git a/include/uapi/linux/msm_audio_wma.h b/include/uapi/linux/msm_audio_wma.h
new file mode 100644
index 000000000000..76dac7b61d68
--- /dev/null
+++ b/include/uapi/linux/msm_audio_wma.h
@@ -0,0 +1,33 @@
+#ifndef _UAPI_MSM_AUDIO_WMA_H
+#define _UAPI_MSM_AUDIO_WMA_H
+
+#define AUDIO_GET_WMA_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+0), unsigned)
+#define AUDIO_SET_WMA_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+1), unsigned)
+
+#define AUDIO_GET_WMA_CONFIG_V2  _IOR(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+2), struct msm_audio_wma_config_v2)
+#define AUDIO_SET_WMA_CONFIG_V2  _IOW(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+3), struct msm_audio_wma_config_v2)
+
+struct msm_audio_wma_config {
+	unsigned short 	armdatareqthr;
+	unsigned short 	channelsdecoded;
+	unsigned short 	wmabytespersec;
+	unsigned short	wmasamplingfreq;
+	unsigned short	wmaencoderopts;
+};
+
+struct msm_audio_wma_config_v2 {
+	unsigned short	format_tag;
+	unsigned short	numchannels;
+	uint32_t	samplingrate;
+	uint32_t	avgbytespersecond;
+	unsigned short	block_align;
+	unsigned short  validbitspersample;
+	uint32_t	channelmask;
+	unsigned short	encodeopt;
+};
+
+#endif /* _UAPI_MSM_AUDIO_WMA_H */
diff --git a/include/uapi/linux/msm_audio_wmapro.h b/include/uapi/linux/msm_audio_wmapro.h
new file mode 100644
index 000000000000..64cbf9e079d6
--- /dev/null
+++ b/include/uapi/linux/msm_audio_wmapro.h
@@ -0,0 +1,22 @@
+#ifndef _UAPI_MSM_AUDIO_WMAPRO_H
+#define _UAPI_MSM_AUDIO_WMAPRO_H
+
+#define AUDIO_GET_WMAPRO_CONFIG  _IOR(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+0), struct msm_audio_wmapro_config)
+#define AUDIO_SET_WMAPRO_CONFIG  _IOW(AUDIO_IOCTL_MAGIC, \
+	  (AUDIO_MAX_COMMON_IOCTL_NUM+1), struct msm_audio_wmapro_config)
+
+struct msm_audio_wmapro_config {
+	unsigned short  armdatareqthr;
+	uint8_t         validbitspersample;
+	uint8_t         numchannels;
+	unsigned short  formattag;
+	uint32_t        samplingrate;
+	uint32_t        avgbytespersecond;
+	unsigned short  asfpacketlength;
+	uint32_t        channelmask;
+	unsigned short  encodeopt;
+	unsigned short  advancedencodeopt;
+	uint32_t        advancedencodeopt2;
+};
+#endif /* _UAPI_MSM_AUDIO_WMAPRO_H */
diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild
index a7f27704f980..8fddb47d1fc4 100644
--- a/include/uapi/sound/Kbuild
+++ b/include/uapi/sound/Kbuild
@@ -10,3 +10,11 @@ header-y += hdsp.h
 header-y += hdspm.h
 header-y += sb16_csp.h
 header-y += sfnt_info.h
+header-y += tlv.h
+header-y += lsm_params.h
+header-y += audio_slimslave.h
+header-y += voice_params.h
+header-y += audio_effects.h
+header-y += voice_svc.h
+header-y += devdep_params.h
+header-y += msmcal-hwdep.h
diff --git a/include/uapi/sound/audio_effects.h b/include/uapi/sound/audio_effects.h
new file mode 100644
index 000000000000..6565acff4073
--- /dev/null
+++ b/include/uapi/sound/audio_effects.h
@@ -0,0 +1,375 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef _AUDIO_EFFECTS_H
+#define _AUDIO_EFFECTS_H
+
+/** AUDIO EFFECTS **/
+
+
+/* CONFIG GET/SET */
+#define CONFIG_CACHE			0
+#define CONFIG_SET			1
+#define CONFIG_GET			2
+
+/* CONFIG HEADER */
+/*
+
+	MODULE_ID,
+	DEVICE,
+	NUM_COMMANDS,
+	COMMAND_ID_1,
+	CONFIG_CACHE/SET/GET,
+	OFFSET_1,
+	LENGTH_1,
+	VALUES_1,
+	...,
+	...,
+	COMMAND_ID_2,
+	CONFIG_CACHE/SET/GET,
+	OFFSET_2,
+	LENGTH_2,
+	VALUES_2,
+	...,
+	...,
+	COMMAND_ID_3,
+	...
+*/
+
+
+/* CONFIG PARAM IDs */
+#define VIRTUALIZER_MODULE		0x00001000
+#define VIRTUALIZER_ENABLE		0x00001001
+#define VIRTUALIZER_STRENGTH		0x00001002
+#define VIRTUALIZER_OUT_TYPE		0x00001003
+#define VIRTUALIZER_GAIN_ADJUST		0x00001004
+#define VIRTUALIZER_ENABLE_PARAM_LEN		1
+#define VIRTUALIZER_STRENGTH_PARAM_LEN		1
+#define VIRTUALIZER_OUT_TYPE_PARAM_LEN		1
+#define VIRTUALIZER_GAIN_ADJUST_PARAM_LEN	1
+
+#define REVERB_MODULE			0x00002000
+#define REVERB_ENABLE			0x00002001
+#define REVERB_MODE			0x00002002
+#define REVERB_PRESET			0x00002003
+#define REVERB_WET_MIX			0x00002004
+#define REVERB_GAIN_ADJUST		0x00002005
+#define REVERB_ROOM_LEVEL		0x00002006
+#define REVERB_ROOM_HF_LEVEL		0x00002007
+#define REVERB_DECAY_TIME		0x00002008
+#define REVERB_DECAY_HF_RATIO		0x00002009
+#define REVERB_REFLECTIONS_LEVEL	0x0000200a
+#define REVERB_REFLECTIONS_DELAY	0x0000200b
+#define REVERB_LEVEL			0x0000200c
+#define REVERB_DELAY			0x0000200d
+#define REVERB_DIFFUSION		0x0000200e
+#define REVERB_DENSITY			0x0000200f
+#define REVERB_ENABLE_PARAM_LEN			1
+#define REVERB_MODE_PARAM_LEN			1
+#define REVERB_PRESET_PARAM_LEN			1
+#define REVERB_WET_MIX_PARAM_LEN		1
+#define REVERB_GAIN_ADJUST_PARAM_LEN		1
+#define REVERB_ROOM_LEVEL_PARAM_LEN		1
+#define REVERB_ROOM_HF_LEVEL_PARAM_LEN		1
+#define REVERB_DECAY_TIME_PARAM_LEN		1
+#define REVERB_DECAY_HF_RATIO_PARAM_LEN		1
+#define REVERB_REFLECTIONS_LEVEL_PARAM_LEN	1
+#define REVERB_REFLECTIONS_DELAY_PARAM_LEN	1
+#define REVERB_LEVEL_PARAM_LEN			1
+#define REVERB_DELAY_PARAM_LEN			1
+#define REVERB_DIFFUSION_PARAM_LEN		1
+#define REVERB_DENSITY_PARAM_LEN		1
+
+#define BASS_BOOST_MODULE		0x00003000
+#define BASS_BOOST_ENABLE		0x00003001
+#define BASS_BOOST_MODE			0x00003002
+#define BASS_BOOST_STRENGTH		0x00003003
+#define BASS_BOOST_ENABLE_PARAM_LEN		1
+#define BASS_BOOST_MODE_PARAM_LEN		1
+#define BASS_BOOST_STRENGTH_PARAM_LEN		1
+
+#define EQ_MODULE			0x00004000
+#define EQ_ENABLE			0x00004001
+#define EQ_CONFIG			0x00004002
+#define EQ_NUM_BANDS			0x00004003
+#define EQ_BAND_LEVELS			0x00004004
+#define EQ_BAND_LEVEL_RANGE		0x00004005
+#define EQ_BAND_FREQS			0x00004006
+#define EQ_SINGLE_BAND_FREQ_RANGE	0x00004007
+#define EQ_SINGLE_BAND_FREQ		0x00004008
+#define EQ_BAND_INDEX			0x00004009
+#define EQ_PRESET_ID			0x0000400a
+#define EQ_NUM_PRESETS			0x0000400b
+#define EQ_PRESET_NAME			0x0000400c
+#define EQ_ENABLE_PARAM_LEN			1
+#define EQ_CONFIG_PARAM_LEN			3
+#define EQ_CONFIG_PER_BAND_PARAM_LEN		5
+#define EQ_NUM_BANDS_PARAM_LEN			1
+#define EQ_BAND_LEVELS_PARAM_LEN		13
+#define EQ_BAND_LEVEL_RANGE_PARAM_LEN		2
+#define EQ_BAND_FREQS_PARAM_LEN			13
+#define EQ_SINGLE_BAND_FREQ_RANGE_PARAM_LEN	2
+#define EQ_SINGLE_BAND_FREQ_PARAM_LEN		1
+#define EQ_BAND_INDEX_PARAM_LEN			1
+#define EQ_PRESET_ID_PARAM_LEN			1
+#define EQ_NUM_PRESETS_PARAM_LEN		1
+#define EQ_PRESET_NAME_PARAM_LEN		32
+
+#define EQ_TYPE_NONE	0
+#define EQ_BASS_BOOST	1
+#define EQ_BASS_CUT	2
+#define EQ_TREBLE_BOOST	3
+#define EQ_TREBLE_CUT	4
+#define EQ_BAND_BOOST	5
+#define EQ_BAND_CUT	6
+
+#define SOFT_VOLUME_MODULE		0x00006000
+#define SOFT_VOLUME_ENABLE		0x00006001
+#define SOFT_VOLUME_GAIN_2CH		0x00006002
+#define SOFT_VOLUME_GAIN_MASTER		0x00006003
+#define SOFT_VOLUME_ENABLE_PARAM_LEN		1
+#define SOFT_VOLUME_GAIN_2CH_PARAM_LEN		2
+#define SOFT_VOLUME_GAIN_MASTER_PARAM_LEN	1
+
+#define SOFT_VOLUME2_MODULE		0x00007000
+#define SOFT_VOLUME2_ENABLE		0x00007001
+#define SOFT_VOLUME2_GAIN_2CH		0x00007002
+#define SOFT_VOLUME2_GAIN_MASTER	0x00007003
+#define SOFT_VOLUME2_ENABLE_PARAM_LEN		SOFT_VOLUME_ENABLE_PARAM_LEN
+#define SOFT_VOLUME2_GAIN_2CH_PARAM_LEN		SOFT_VOLUME_GAIN_2CH_PARAM_LEN
+#define SOFT_VOLUME2_GAIN_MASTER_PARAM_LEN	\
+					SOFT_VOLUME_GAIN_MASTER_PARAM_LEN
+
+#define PBE_CONF_MODULE_ID	0x00010C2A
+#define PBE_CONF_PARAM_ID	0x00010C49
+
+#define PBE_MODULE		0x00008000
+#define PBE_ENABLE		0x00008001
+#define PBE_CONFIG		0x00008002
+#define PBE_ENABLE_PARAM_LEN		1
+#define PBE_CONFIG_PARAM_LEN		28
+
+#define COMMAND_PAYLOAD_LEN	3
+#define COMMAND_PAYLOAD_SZ	(COMMAND_PAYLOAD_LEN * sizeof(uint32_t))
+#define MAX_INBAND_PARAM_SZ	4096
+#define Q27_UNITY		(1 << 27)
+#define Q8_UNITY		(1 << 8)
+#define CUSTOM_OPENSL_PRESET	18
+
+#define VIRTUALIZER_ENABLE_PARAM_SZ	\
+			(VIRTUALIZER_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define VIRTUALIZER_STRENGTH_PARAM_SZ	\
+			(VIRTUALIZER_STRENGTH_PARAM_LEN*sizeof(uint32_t))
+#define VIRTUALIZER_OUT_TYPE_PARAM_SZ	\
+			(VIRTUALIZER_OUT_TYPE_PARAM_LEN*sizeof(uint32_t))
+#define VIRTUALIZER_GAIN_ADJUST_PARAM_SZ	\
+			(VIRTUALIZER_GAIN_ADJUST_PARAM_LEN*sizeof(uint32_t))
+struct virtualizer_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	uint32_t strength;
+	uint32_t out_type;
+	int32_t gain_adjust;
+};
+
+#define NUM_OSL_REVERB_PRESETS_SUPPORTED	6
+#define REVERB_ENABLE_PARAM_SZ		\
+			(REVERB_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_MODE_PARAM_SZ		\
+			(REVERB_MODE_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_PRESET_PARAM_SZ		\
+			(REVERB_PRESET_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_WET_MIX_PARAM_SZ		\
+			(REVERB_WET_MIX_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_GAIN_ADJUST_PARAM_SZ	\
+			(REVERB_GAIN_ADJUST_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_ROOM_LEVEL_PARAM_SZ	\
+			(REVERB_ROOM_LEVEL_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_ROOM_HF_LEVEL_PARAM_SZ	\
+			(REVERB_ROOM_HF_LEVEL_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_DECAY_TIME_PARAM_SZ	\
+			(REVERB_DECAY_TIME_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_DECAY_HF_RATIO_PARAM_SZ	\
+			(REVERB_DECAY_HF_RATIO_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_REFLECTIONS_LEVEL_PARAM_SZ	\
+			(REVERB_REFLECTIONS_LEVEL_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_REFLECTIONS_DELAY_PARAM_SZ	\
+			(REVERB_REFLECTIONS_DELAY_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_LEVEL_PARAM_SZ		\
+			(REVERB_LEVEL_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_DELAY_PARAM_SZ		\
+			(REVERB_DELAY_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_DIFFUSION_PARAM_SZ	\
+			(REVERB_DIFFUSION_PARAM_LEN*sizeof(uint32_t))
+#define REVERB_DENSITY_PARAM_SZ		\
+			(REVERB_DENSITY_PARAM_LEN*sizeof(uint32_t))
+struct reverb_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	uint32_t mode;
+	uint32_t preset;
+	uint32_t wet_mix;
+	int32_t  gain_adjust;
+	int32_t  room_level;
+	int32_t  room_hf_level;
+	uint32_t decay_time;
+	uint32_t decay_hf_ratio;
+	int32_t  reflections_level;
+	uint32_t reflections_delay;
+	int32_t  level;
+	uint32_t delay;
+	uint32_t diffusion;
+	uint32_t density;
+};
+
+#define BASS_BOOST_ENABLE_PARAM_SZ	\
+			(BASS_BOOST_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define BASS_BOOST_MODE_PARAM_SZ	\
+			(BASS_BOOST_MODE_PARAM_LEN*sizeof(uint32_t))
+#define BASS_BOOST_STRENGTH_PARAM_SZ	\
+			(BASS_BOOST_STRENGTH_PARAM_LEN*sizeof(uint32_t))
+struct bass_boost_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	uint32_t mode;
+	uint32_t strength;
+};
+
+
+#define MAX_EQ_BANDS 12
+#define MAX_OSL_EQ_BANDS 5
+#define EQ_ENABLE_PARAM_SZ			\
+			(EQ_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define EQ_CONFIG_PARAM_SZ			\
+			(EQ_CONFIG_PARAM_LEN*sizeof(uint32_t))
+#define EQ_CONFIG_PER_BAND_PARAM_SZ		\
+			(EQ_CONFIG_PER_BAND_PARAM_LEN*sizeof(uint32_t))
+#define EQ_CONFIG_PARAM_MAX_LEN			(EQ_CONFIG_PARAM_LEN+\
+			MAX_EQ_BANDS*EQ_CONFIG_PER_BAND_PARAM_LEN)
+#define EQ_CONFIG_PARAM_MAX_SZ			\
+			(EQ_CONFIG_PARAM_MAX_LEN*sizeof(uint32_t))
+#define EQ_NUM_BANDS_PARAM_SZ			\
+			(EQ_NUM_BANDS_PARAM_LEN*sizeof(uint32_t))
+#define EQ_BAND_LEVELS_PARAM_SZ			\
+			(EQ_BAND_LEVELS_PARAM_LEN*sizeof(uint32_t))
+#define EQ_BAND_LEVEL_RANGE_PARAM_SZ		\
+			(EQ_BAND_LEVEL_RANGE_PARAM_LEN*sizeof(uint32_t))
+#define EQ_BAND_FREQS_PARAM_SZ			\
+			(EQ_BAND_FREQS_PARAM_LEN*sizeof(uint32_t))
+#define EQ_SINGLE_BAND_FREQ_RANGE_PARAM_SZ	\
+			(EQ_SINGLE_BAND_FREQ_RANGE_PARAM_LEN*sizeof(uint32_t))
+#define EQ_SINGLE_BAND_FREQ_PARAM_SZ		\
+			(EQ_SINGLE_BAND_FREQ_PARAM_LEN*sizeof(uint32_t))
+#define EQ_BAND_INDEX_PARAM_SZ			\
+			(EQ_BAND_INDEX_PARAM_LEN*sizeof(uint32_t))
+#define EQ_PRESET_ID_PARAM_SZ			\
+			(EQ_PRESET_ID_PARAM_LEN*sizeof(uint32_t))
+#define EQ_NUM_PRESETS_PARAM_SZ			\
+			(EQ_NUM_PRESETS_PARAM_LEN*sizeof(uint8_t))
+struct eq_config_t {
+	int32_t eq_pregain;
+	int32_t preset_id;
+	uint32_t num_bands;
+};
+struct eq_per_band_config_t {
+	int32_t band_idx;
+	uint32_t filter_type;
+	uint32_t freq_millihertz;
+	int32_t  gain_millibels;
+	uint32_t quality_factor;
+};
+struct eq_per_band_freq_range_t {
+	uint32_t band_index;
+	uint32_t min_freq_millihertz;
+	uint32_t max_freq_millihertz;
+};
+
+struct eq_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	struct eq_config_t config;
+	struct eq_per_band_config_t per_band_cfg[MAX_EQ_BANDS];
+	struct eq_per_band_freq_range_t per_band_freq_range[MAX_EQ_BANDS];
+	uint32_t band_index;
+	uint32_t freq_millihertz;
+};
+
+#define PBE_ENABLE_PARAM_SZ	\
+			(PBE_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define PBE_CONFIG_PARAM_SZ	\
+			(PBE_CONFIG_PARAM_LEN*sizeof(uint16_t))
+struct pbe_config_t {
+	int16_t  real_bass_mix;
+	int16_t  bass_color_control;
+	uint16_t main_chain_delay;
+	uint16_t xover_filter_order;
+	uint16_t bandpass_filter_order;
+	int16_t  drc_delay;
+	uint16_t rms_tav;
+	int16_t exp_threshold;
+	uint16_t exp_slope;
+	int16_t comp_threshold;
+	uint16_t comp_slope;
+	uint16_t makeup_gain;
+	uint32_t comp_attack;
+	uint32_t comp_release;
+	uint32_t exp_attack;
+	uint32_t exp_release;
+	int16_t limiter_bass_threshold;
+	int16_t limiter_high_threshold;
+	int16_t limiter_bass_makeup_gain;
+	int16_t limiter_high_makeup_gain;
+	int16_t limiter_bass_gc;
+	int16_t limiter_high_gc;
+	int16_t  limiter_delay;
+	uint16_t reserved;
+	/* place holder for filter coeffs to be followed */
+	int32_t p1LowPassCoeffs[5*2];
+	int32_t p1HighPassCoeffs[5*2];
+	int32_t p1BandPassCoeffs[5*3];
+	int32_t p1BassShelfCoeffs[5];
+	int32_t p1TrebleShelfCoeffs[5];
+} __packed;
+
+struct pbe_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	uint32_t cfg_len;
+	struct pbe_config_t config;
+};
+
+#define SOFT_VOLUME_ENABLE_PARAM_SZ		\
+			(SOFT_VOLUME_ENABLE_PARAM_LEN*sizeof(uint32_t))
+#define SOFT_VOLUME_GAIN_MASTER_PARAM_SZ	\
+			(SOFT_VOLUME_GAIN_MASTER_PARAM_LEN*sizeof(uint32_t))
+#define SOFT_VOLUME_GAIN_2CH_PARAM_SZ		\
+			(SOFT_VOLUME_GAIN_2CH_PARAM_LEN*sizeof(uint16_t))
+struct soft_volume_params {
+	uint32_t device;
+	uint32_t enable_flag;
+	uint32_t master_gain;
+	uint32_t left_gain;
+	uint32_t right_gain;
+};
+
+struct msm_nt_eff_all_config {
+	struct bass_boost_params bass_boost;
+	struct pbe_params pbe;
+	struct virtualizer_params virtualizer;
+	struct reverb_params reverb;
+	struct eq_params equalizer;
+	struct soft_volume_params saplus_vol;
+	struct soft_volume_params topo_switch_vol;
+};
+
+#endif /*_MSM_AUDIO_EFFECTS_H*/
diff --git a/include/uapi/sound/devdep_params.h b/include/uapi/sound/devdep_params.h
new file mode 100644
index 000000000000..5061ec0da356
--- /dev/null
+++ b/include/uapi/sound/devdep_params.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+#ifndef _DEV_DEP_H
+#define _DEV_DEP_H
+
+struct dolby_param_data {
+	int32_t version;
+	int32_t device_id;
+	int32_t be_id;
+	int32_t param_id;
+	int32_t length;
+	int32_t __user *data;
+};
+
+struct dolby_param_license {
+	int32_t dmid;
+	int32_t license_key;
+};
+
+#define SNDRV_DEVDEP_DAP_IOCTL_SET_PARAM\
+		_IOWR('U', 0x10, struct dolby_param_data)
+#define SNDRV_DEVDEP_DAP_IOCTL_GET_PARAM\
+		_IOR('U', 0x11, struct dolby_param_data)
+#define SNDRV_DEVDEP_DAP_IOCTL_DAP_COMMAND\
+		_IOWR('U', 0x13, struct dolby_param_data)
+#define SNDRV_DEVDEP_DAP_IOCTL_DAP_LICENSE\
+		_IOWR('U', 0x14, struct dolby_param_license)
+#define SNDRV_DEVDEP_DAP_IOCTL_GET_VISUALIZER\
+		_IOR('U', 0x15, struct dolby_param_data)
+
+#define DTS_EAGLE_MODULE			0x00005000
+#define DTS_EAGLE_MODULE_ENABLE			0x00005001
+#define EAGLE_DRIVER_ID				0xF2
+#define DTS_EAGLE_IOCTL_GET_CACHE_SIZE		_IOR(EAGLE_DRIVER_ID, 0, int)
+#define DTS_EAGLE_IOCTL_SET_CACHE_SIZE		_IOW(EAGLE_DRIVER_ID, 1, int)
+#define DTS_EAGLE_IOCTL_GET_PARAM		_IOR(EAGLE_DRIVER_ID, 2, void*)
+#define DTS_EAGLE_IOCTL_SET_PARAM		_IOW(EAGLE_DRIVER_ID, 3, void*)
+#define DTS_EAGLE_IOCTL_SET_CACHE_BLOCK		_IOW(EAGLE_DRIVER_ID, 4, void*)
+#define DTS_EAGLE_IOCTL_SET_ACTIVE_DEVICE	_IOW(EAGLE_DRIVER_ID, 5, void*)
+#define DTS_EAGLE_IOCTL_GET_LICENSE		_IOR(EAGLE_DRIVER_ID, 6, void*)
+#define DTS_EAGLE_IOCTL_SET_LICENSE		_IOW(EAGLE_DRIVER_ID, 7, void*)
+#define DTS_EAGLE_IOCTL_SEND_LICENSE		_IOW(EAGLE_DRIVER_ID, 8, int)
+#define DTS_EAGLE_IOCTL_SET_VOLUME_COMMANDS	_IOW(EAGLE_DRIVER_ID, 9, void*)
+#define DTS_EAGLE_FLAG_IOCTL_PRE		(1<<30)
+#define DTS_EAGLE_FLAG_IOCTL_JUSTSETCACHE	(1<<31)
+#define DTS_EAGLE_FLAG_IOCTL_GETFROMCORE       DTS_EAGLE_FLAG_IOCTL_JUSTSETCACHE
+#define DTS_EAGLE_FLAG_IOCTL_MASK		(~(DTS_EAGLE_FLAG_IOCTL_PRE | \
+					     DTS_EAGLE_FLAG_IOCTL_JUSTSETCACHE))
+#define DTS_EAGLE_FLAG_ALSA_GET			(1<<31)
+
+struct dts_eagle_param_desc {
+	uint32_t id;
+	uint32_t size;
+	int32_t offset;
+	uint32_t device;
+} __packed;
+
+#endif
diff --git a/include/uapi/sound/lsm_params.h b/include/uapi/sound/lsm_params.h
new file mode 100644
index 000000000000..eafdc117413a
--- /dev/null
+++ b/include/uapi/sound/lsm_params.h
@@ -0,0 +1,175 @@
+#ifndef _UAPI_LSM_PARAMS_H__
+#define _UAPI_LSM_PARAMS_H__
+
+#include <linux/types.h>
+#include <sound/asound.h>
+
+#define SNDRV_LSM_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
+
+#define LSM_OUT_FORMAT_PCM (0)
+#define LSM_OUT_FORMAT_ADPCM (1 << 0)
+
+#define LSM_OUT_DATA_RAW (0)
+#define LSM_OUT_DATA_PACKED (1)
+
+#define LSM_OUT_DATA_EVENTS_DISABLED (0)
+#define LSM_OUT_DATA_EVENTS_ENABLED (1)
+
+#define LSM_OUT_TRANSFER_MODE_RT (0)
+#define LSM_OUT_TRANSFER_MODE_FTRT (1)
+
+enum lsm_app_id {
+	LSM_VOICE_WAKEUP_APP_ID = 1,
+	LSM_VOICE_WAKEUP_APP_ID_V2 = 2,
+};
+
+enum lsm_detection_mode {
+	LSM_MODE_KEYWORD_ONLY_DETECTION = 1,
+	LSM_MODE_USER_KEYWORD_DETECTION
+};
+
+enum lsm_vw_status {
+	LSM_VOICE_WAKEUP_STATUS_RUNNING = 1,
+	LSM_VOICE_WAKEUP_STATUS_DETECTED,
+	LSM_VOICE_WAKEUP_STATUS_END_SPEECH,
+	LSM_VOICE_WAKEUP_STATUS_REJECTED
+};
+
+enum LSM_PARAM_TYPE {
+	LSM_ENDPOINT_DETECT_THRESHOLD = 0,
+	LSM_OPERATION_MODE,
+	LSM_GAIN,
+	LSM_MIN_CONFIDENCE_LEVELS,
+	LSM_REG_SND_MODEL,
+	LSM_DEREG_SND_MODEL,
+	LSM_CUSTOM_PARAMS,
+	/* driver ioctl will parse only so many params */
+	LSM_PARAMS_MAX,
+};
+
+/*
+ * Data for LSM_ENDPOINT_DETECT_THRESHOLD param_type
+ * @epd_begin: Begin threshold
+ * @epd_end: End threshold
+ */
+struct snd_lsm_ep_det_thres {
+	__u32 epd_begin;
+	__u32 epd_end;
+};
+
+/*
+ * Data for LSM_OPERATION_MODE param_type
+ * @mode: The detection mode to be used
+ * @detect_failure: Setting to enable failure detections.
+ */
+struct snd_lsm_detect_mode {
+	enum lsm_detection_mode mode;
+	bool detect_failure;
+};
+
+/*
+ * Data for LSM_GAIN param_type
+ * @gain: The gain to be applied on LSM
+ */
+struct snd_lsm_gain {
+	__u16 gain;
+};
+
+
+struct snd_lsm_sound_model_v2 {
+	__u8 __user *data;
+	__u8 *confidence_level;
+	__u32 data_size;
+	enum lsm_detection_mode detection_mode;
+	__u8 num_confidence_levels;
+	bool detect_failure;
+};
+
+struct snd_lsm_session_data {
+	enum lsm_app_id app_id;
+};
+
+struct snd_lsm_event_status {
+	__u16 status;
+	__u16 payload_size;
+	__u8 payload[0];
+};
+
+struct snd_lsm_detection_params {
+	__u8 *conf_level;
+	enum lsm_detection_mode detect_mode;
+	__u8 num_confidence_levels;
+	bool detect_failure;
+};
+
+/*
+ * Param info for each parameter type
+ * @module_id: Module to which parameter is to be set
+ * @param_id: Parameter that is to be set
+ * @param_size: size (in number of bytes) for the data
+ *		in param_data.
+ *		For confidence levels, this is num_conf_levels
+ *		For REG_SND_MODEL, this is size of sound model
+ *		For CUSTOM_PARAMS, this is size of the entire blob of data
+ * @param_data: Data for the parameter.
+ *		For some param_types this is a structure defined, ex: LSM_GAIN
+ *		For CONFIDENCE_LEVELS, this is array of confidence levels
+ *		For REG_SND_MODEL, this is the sound model data
+ *		For CUSTOM_PARAMS, this is the blob of custom data.
+ */
+struct lsm_params_info {
+	__u32 module_id;
+	__u32 param_id;
+	__u32 param_size;
+	__u8 __user *param_data;
+	enum LSM_PARAM_TYPE param_type;
+};
+
+/*
+ * Data passed to the SET_PARAM_V2 IOCTL
+ * @num_params: Number of params that are to be set
+ *		should not be greater than LSM_PARAMS_MAX
+ * @params: Points to an array of lsm_params_info
+ *	    Each entry points to one parameter to set
+ * @data_size: size (in bytes) for params
+ *	       should be equal to
+ *	       num_params * sizeof(struct lsm_parms_info)
+ */
+struct snd_lsm_module_params {
+	__u8 __user *params;
+	__u32 num_params;
+	__u32 data_size;
+};
+
+/*
+ * Data passed to LSM_OUT_FORMAT_CFG IOCTL
+ * @format: The media format enum
+ * @packing: indicates the packing method used for data path
+ * @events: indicates whether data path events need to be enabled
+ * @transfer_mode: indicates whether FTRT mode or RT mode.
+ */
+struct snd_lsm_output_format_cfg {
+	__u8 format;
+	__u8 packing;
+	__u8 events;
+	__u8 mode;
+};
+
+#define SNDRV_LSM_DEREG_SND_MODEL _IOW('U', 0x01, int)
+#define SNDRV_LSM_EVENT_STATUS	_IOW('U', 0x02, struct snd_lsm_event_status)
+#define SNDRV_LSM_ABORT_EVENT	_IOW('U', 0x03, int)
+#define SNDRV_LSM_START		_IOW('U', 0x04, int)
+#define SNDRV_LSM_STOP		_IOW('U', 0x05, int)
+#define SNDRV_LSM_SET_SESSION_DATA _IOW('U', 0x06, struct snd_lsm_session_data)
+#define SNDRV_LSM_REG_SND_MODEL_V2 _IOW('U', 0x07,\
+					struct snd_lsm_sound_model_v2)
+#define SNDRV_LSM_LAB_CONTROL	_IOW('U', 0x08, uint32_t)
+#define SNDRV_LSM_STOP_LAB	_IO('U', 0x09)
+#define SNDRV_LSM_SET_PARAMS	_IOW('U', 0x0A, \
+					struct snd_lsm_detection_params)
+#define SNDRV_LSM_SET_MODULE_PARAMS	_IOW('U', 0x0B, \
+					struct snd_lsm_module_params)
+#define SNDRV_LSM_OUT_FORMAT_CFG _IOW('U', 0x0C, \
+				      struct snd_lsm_output_format_cfg)
+
+#endif
diff --git a/include/uapi/sound/msmcal-hwdep.h b/include/uapi/sound/msmcal-hwdep.h
new file mode 100644
index 000000000000..2a294824fb00
--- /dev/null
+++ b/include/uapi/sound/msmcal-hwdep.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2014-2015, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+#ifndef _CALIB_HWDEP_H
+#define _CALIB_HWDEP_H
+
+#define WCD9XXX_CODEC_HWDEP_NODE    1000
+enum wcd_cal_type {
+	WCD9XXX_MIN_CAL,
+	WCD9XXX_ANC_CAL = WCD9XXX_MIN_CAL,
+	WCD9XXX_MAD_CAL,
+	WCD9XXX_MBHC_CAL,
+	WCD9XXX_VBAT_CAL,
+	WCD9XXX_MAX_CAL,
+};
+
+struct wcdcal_ioctl_buffer {
+	__u32 size;
+	__u8 __user *buffer;
+	enum wcd_cal_type cal_type;
+};
+
+#define SNDRV_CTL_IOCTL_HWDEP_CAL_TYPE \
+	_IOW('U', 0x1, struct wcdcal_ioctl_buffer)
+
+#endif /*_CALIB_HWDEP_H*/