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android_kernel_xiaomi_sm8350/include/dsp/apr_audio-v2.h
Deru Wang 7903818708 asoc: add AFE logging PID and APIs 
The current AFE logging point is global, once enabled all active
AFE ports will log its data through DIAG to PC, However, diag
logging throughput is limited.
Need add kcontrol for enable/disable AFE logging by AFE port
per direction.

Change-Id: I05a12f2eb4bc5b5a3ad39b8bbf2f4148bec05002
Signed-off-by: Deru Wang <deruwang@codeaurora.org>
2021-03-14 19:29:33 -07:00

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/* SPDX-License-Identifier: GPL-2.0-only */
/*
* Copyright (c) 2012-2021, The Linux Foundation. All rights reserved.
*/
#ifndef _APR_AUDIO_V2_H_
#define _APR_AUDIO_V2_H_
#include <ipc/apr.h>
#include <audio/linux/msm_audio.h>
/* number of threshold levels in speaker protection module */
#define MAX_CPS_LEVELS 3
/* 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;
};
/* Common structures and definitions used for instance ID support */
/* Instance ID definitions */
#define INSTANCE_ID_0 0x0000
struct mem_mapping_hdr {
/*
* LSW of parameter data payload address. Supported values: any.
* - Must be set to zero for in-band data.
*/
u32 data_payload_addr_lsw;
/*
* 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 data_payload_addr_msw;
/*
* Memory map handle returned by DSP through
* ASM_CMD_SHARED_MEM_MAP_REGIONS command.
* Supported Values: Any.
* If memory map handle is NULL, the parameter data payloads are
* within the message payload (in-band).
* If memory map handle is non-NULL, the parameter data payloads begin
* at the address specified in the address MSW and LSW (out-of-band).
*/
u32 mem_map_handle;
} __packed;
/*
* Payload format for parameter data.
* Immediately following these structures are param_size bytes of parameter
* data.
*/
struct param_hdr_v1 {
/* Valid ID of the module. */
uint32_t module_id;
/* Valid ID of the parameter. */
uint32_t param_id;
/* The size of the parameter specified by the module/param ID combo */
uint16_t param_size;
/* This field must be set to zero. */
uint16_t reserved;
} __packed;
struct param_hdr_v2 {
/* Valid ID of the module. */
uint32_t module_id;
/* Valid ID of the parameter. */
uint32_t param_id;
/* The size of the parameter specified by the module/param ID combo */
uint32_t param_size;
} __packed;
struct param_hdr_v3 {
/* Valid ID of the module. */
uint32_t module_id;
/* Instance of the module. */
uint16_t instance_id;
/* This field must be set to zero. */
uint16_t reserved;
/* Valid ID of the parameter. */
uint32_t param_id;
/* The size of the parameter specified by the module/param ID combo */
uint32_t param_size;
} __packed;
/* A union of all param_hdr versions for versitility and max size */
union param_hdrs {
struct param_hdr_v1 v1;
struct param_hdr_v2 v2;
struct param_hdr_v3 v3;
};
struct module_instance_info {
/* Module ID. */
u32 module_id;
/* Instance of the module */
u16 instance_id;
/* Reserved. This field must be set to zero. */
u16 reserved;
} __packed;
/* Begin service specific definitions and structures */
#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
#define ADM_MATRIX_ID_COMPRESSED_AUDIO_TX 3
#define ADM_MATRIX_ID_LISTEN_TX 4
/* 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
/* This command allows a client to open a COPP/Voice Proc the
* way as ADM_CMD_DEVICE_OPEN_V5 but supports multiple endpoint2
* channels.
*
* @return
* #ADM_CMDRSP_DEVICE_OPEN_V6 with the resulting status and
* COPP ID.
*/
#define ADM_CMD_DEVICE_OPEN_V6 0x00010356
/* This command allows a client to open a COPP/Voice Proc the
* way as ADM_CMD_DEVICE_OPEN_V8 but supports any number channel
* of configuration.
*
* @return
* #ADM_CMDRSP_DEVICE_OPEN_V8 with the resulting status and
* COPP ID.
*/
#define ADM_CMD_DEVICE_OPEN_V8 0x0001036A
/* 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 32000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_32K 32000
/* 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 relevant 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;
/* ADM device open command payload of the
* #ADM_CMD_DEVICE_OPEN_V6 command.
*/
struct adm_cmd_device_open_v6 {
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 relevant only for an audio Rx COPP.
* For the voice processor block and Tx audio block, this field
* is set to zero and is ignored.
*/
u16 dev_num_channel_eid2;
/* Number of channels the voice processor block sends
* to/receives from the endpoint2.
* Supported values: 1 to 8.
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u16 bit_width_eid2;
/* Bit width (in bits) that the voice processor sends
* to/receives from the endpoint2.
* Supported values: 16 and 24.
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u32 sample_rate_eid2;
/* Sampling rate at which the voice processor Tx block
* interfaces with the endpoint2.
* Supported values for Tx voice processor: >0 and <=384 kHz
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u8 dev_channel_mapping_eid2[8];
/* Array of channel mapping of buffers that the voice processor
* sends to the endpoint. Channel[i] mapping describes channel
* I inside the buffer, where 0 < i < dev_num_channel.
* This value is relevant only for the Tx voice processor.
* The values are ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
} __packed;
/* ADM device open endpoint payload the
* #ADM_CMD_DEVICE_OPEN_V8 command.
*/
struct adm_device_endpoint_payload {
u16 dev_num_channel;
/* Number of channels the audio COPP sends to/receives from
* the endpoint.
* Supported values: 1 to 32.
* 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[32];
} __packed;
/* ADM device open command payload of the
* #ADM_CMD_DEVICE_OPEN_V8 command.
*/
struct adm_cmd_device_open_v8 {
struct apr_hdr hdr;
u16 flags;
/* Bit width Native mode enabled : 11th bit of flag parameter
* If 11th bit of flag is set then that means matrix mixer will be
* running in native mode for bit width for this device session.
*
* Channel Native mode enabled : 12th bit of flag parameter
* If 12th bit of flag is set then that means matrix mixer will be
* running in native mode for channel configuration for this device session.
* All other bits are reserved; clients must set them to 0.
*/
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.
*/
u32 topology_id;
/* Audio COPP topology ID; 32-bit GUID. */
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]}.
*/
u16 endpoint_id_3;
/*
* Logical and physical endpoint ID of the audio path.
* This indicated afe rx port in ADM loopback use cases.
* In all other use cases this should be set to 0xffff
*/
u16 compressed_data_type;
} __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
#define ADM_CMD_SET_PP_PARAMS_V6 0x0001035D
/*
* Structure of the ADM Set PP Params command. Parameter data must be
* pre-packed with correct header for either V2 or V3 when sent in-band.
* Use q6core_pack_pp_params to pack the header and data correctly depending on
* Instance ID support.
*/
struct adm_cmd_set_pp_params {
/* APR Header */
struct apr_hdr apr_hdr;
/* The memory mapping header to be used when sending out of band */
struct mem_mapping_hdr mem_hdr;
/*
* 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;
/*
* Parameter data for in band payload. This should be structured as the
* parameter header immediately followed by the parameter data. Multiple
* parameters can be set in one command by repeating the header followed
* by the data for as many parameters as need to be set.
* Use q6core_pack_pp_params to pack the header and data correctly
* depending on Instance ID support.
*/
u8 param_data[0];
} __packed;
#define ASM_STREAM_CMD_REGISTER_PP_EVENTS 0x00013213
#define ASM_STREAM_PP_EVENT 0x00013214
#define ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE 0x1321C
#define ASM_IEC_61937_MEDIA_FMT_EVENT 0x1321D
#define DSP_STREAM_CMD "ADSP Stream Cmd"
#define DSP_STREAM_CALLBACK "ADSP Stream Callback Event"
#define DSP_STREAM_CALLBACK_QUEUE_SIZE 1024
struct dsp_stream_callback_list {
struct list_head list;
struct msm_adsp_event_data event;
};
struct dsp_stream_callback_prtd {
uint16_t event_count;
struct list_head event_queue;
spinlock_t prtd_spin_lock;
};
/* 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;
/* set customized mixing on matrix mixer.
* Updated to account for both LSM as well as ASM path.
*/
#define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V6 0x00010364
struct adm_cmd_set_pspd_mtmx_strtr_params_v6 {
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 stream_type;
} __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;
/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command. */
#define ADM_CMDRSP_DEVICE_OPEN_V6 0x00010357
/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V8 command. */
#define ADM_CMDRSP_DEVICE_OPEN_V8 0x0001036B
/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message,
* which returns the
* status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command
* is the exact same as ADM_CMDRSP_DEVICE_OPEN_V5.
*/
/* This command allows a query of one COPP parameter. */
#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A
#define ADM_CMD_GET_PP_PARAMS_V6 0x0001035E
/*
* Structure of the ADM Get PP Params command. Parameter header must be
* packed correctly for either V2 or V3. Use q6core_pack_pp_params to pack the
* header correctly depending on Instance ID support.
*/
struct adm_cmd_get_pp_params {
struct apr_hdr apr_hdr;
/* The memory mapping header to be used when requesting outband */
struct mem_mapping_hdr mem_hdr;
/* Parameter header for in band payload. */
union param_hdrs param_hdr;
} __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 param_hdr_v1
* 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 {
/* Status message (error code).*/
u32 status;
/* The header that identifies the subsequent parameter data */
struct param_hdr_v1 param_hdr;
/* The parameter data returned */
u32 param_data[0];
} __packed;
/*
* Returns parameter values in response to an #ADM_CMD_GET_PP_PARAMS_V5/6
* command.
*/
#define ADM_CMDRSP_GET_PP_PARAMS_V6 0x0001035F
/*
* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V6 message,
* which returns parameter values in response
* to an #ADM_CMD_GET_PP_PARAMS_V6 command.
* Immediately following this
* structure is the param_hdr_v3
* 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_v6 {
/* Status message (error code).*/
u32 status;
/* The header that identifies the subsequent parameter data */
struct param_hdr_v3 param_hdr;
/* The parameter data returned */
u32 param_data[0];
} __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;
/*
* ID of the Media Format Converter (MFC) module.
* This module supports the following parameter IDs:
* #AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT
* #AUDPROC_CHMIXER_PARAM_ID_COEFF
*/
#define AUDPROC_MODULE_ID_MFC 0x00010912
/* ID of the Output Media Format parameters used by AUDPROC_MODULE_ID_MFC.
*
*/
#define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913
/* Param ID of Channel Mixer used by AUDPROC_MODULE_ID_MFC */
#define AUDPROC_CHMIXER_PARAM_ID_COEFF 0x00010342
/*
* ID of the Media Format Converter (MFC) module present in EC REF COPP.
* This module supports the all param IDs supported by AUDPROC_MODULE_ID_MFC.
*/
#define AUDPROC_MODULE_ID_MFC_EC_REF 0x0001092C
#define PARAM_ID_FFV_SPF_FREEZE 0x00010960
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;
/* Maximum number of channels supported by MFC media fmt params */
#define AUDPROC_MFC_OUT_CHANNELS_MAX 8
struct audproc_mfc_param_media_fmt {
uint32_t sampling_rate;
uint16_t bits_per_sample;
uint16_t num_channels;
uint16_t channel_type[AUDPROC_MFC_OUT_CHANNELS_MAX];
} __packed;
struct audproc_volume_ctrl_master_gain {
/* 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 {
/*
* 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 {
/*
* 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
/*
* Allows a client to control the gains on various session-to-COPP paths.
* Maximum support 32 channels
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_V7 0x0001036C
/* 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;
/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V7 command.
* Immediately following this structure are num_gains of the
* adm_session_copp_gain_v5structure.
*/
struct adm_cmd_matrix_ramp_gains_v7 {
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.
*/
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_V7 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_v7 {
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 stream_type;
/* Type of stream_type.
* Supported value: #STREAM_TYPE_ASM STREAM_TYPE_LSM
*/
u16 num_channels;
/* Number of channels on which gain needs to be applied.
* Supported value: 1 to 32.
*/
u16 reserved_for_align;
/* Reserved. This field must be set to zero. */
} __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
/* 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_V7 0x0001036D
/* 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;
/* Payload of the #ADM_CMD_MATRIX_MUTE_V7 command*/
struct adm_cmd_matrix_mute_v7 {
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.
*/
u16 session_id;
/* Handle of the ASM session.
* Supported values: 1 to .
*/
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
*/
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 stream_type;
/* Specify whether the stream type is connectedon the ASM or LSM
* Supported value: 1
*/
u16 num_channels;
/* Number of channels on which gain needs to be applied
* Supported value: 1 to 32
*/
} __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_7_RX 0x400e
#define SLIMBUS_7_TX 0x400f
#define SLIMBUS_8_RX 0x4010
#define SLIMBUS_8_TX 0x4011
#define SLIMBUS_9_RX 0x4012
#define SLIMBUS_9_TX 0x4013
#define SLIMBUS_PORT_LAST SLIMBUS_9_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 RT_PROXY_PORT_002_RX 0x2002
#define AFE_LOOPBACK_TX 0x6001
#define HDMI_RX_MS 0x6002
#define DISPLAY_PORT_RX 0x6020
#define AFE_LANE_MASK_INVALID 0
#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 0x4
/* 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+0x60-1)
/* 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_TERTIARY_PCM_RX 0x1012
#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013
#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014
#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015
#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
/* ID of the Internal 0 MI2S Rx port */
#define AFE_PORT_ID_INT0_MI2S_RX 0x102E
/* ID of the Internal 0 MI2S Tx port */
#define AFE_PORT_ID_INT0_MI2S_TX 0x102F
/* ID of the Internal 1 MI2S Rx port */
#define AFE_PORT_ID_INT1_MI2S_RX 0x1030
/* ID of the Internal 1 MI2S Tx port */
#define AFE_PORT_ID_INT1_MI2S_TX 0x1031
/* ID of the Internal 2 MI2S Rx port */
#define AFE_PORT_ID_INT2_MI2S_RX 0x1032
/* ID of the Internal 2 MI2S Tx port */
#define AFE_PORT_ID_INT2_MI2S_TX 0x1033
/* ID of the Internal 3 MI2S Rx port */
#define AFE_PORT_ID_INT3_MI2S_RX 0x1034
/* ID of the Internal 3 MI2S Tx port */
#define AFE_PORT_ID_INT3_MI2S_TX 0x1035
/* ID of the Internal 4 MI2S Rx port */
#define AFE_PORT_ID_INT4_MI2S_RX 0x1036
/* ID of the Internal 4 MI2S Tx port */
#define AFE_PORT_ID_INT4_MI2S_TX 0x1037
/* ID of the Internal 5 MI2S Rx port */
#define AFE_PORT_ID_INT5_MI2S_RX 0x1038
/* ID of the Internal 5 MI2S Tx port */
#define AFE_PORT_ID_INT5_MI2S_TX 0x1039
/* ID of the Internal 6 MI2S Rx port */
#define AFE_PORT_ID_INT6_MI2S_RX 0x103A
/* ID of the Internal 6 MI2S Tx port */
#define AFE_PORT_ID_INT6_MI2S_TX 0x103B
#define AFE_PORT_ID_QUINARY_PCM_RX 0x103C
#define AFE_PORT_ID_QUINARY_PCM_TX 0x103D
/* ID of the senary auxiliary PCM Rx port. */
#define AFE_PORT_ID_SENARY_PCM_RX 0x103E
/* ID of the senary auxiliary PCM Tx port. */
#define AFE_PORT_ID_SENARY_PCM_TX 0x103F
#define AFE_PORT_ID_PRIMARY_META_MI2S_RX 0x1300
#define AFE_PORT_ID_SECONDARY_META_MI2S_RX 0x1302
#define AFE_PORT_ID_PRIMARY_SPDIF_RX 0x5000
#define AFE_PORT_ID_PRIMARY_SPDIF_TX 0x5001
#define AFE_PORT_ID_SECONDARY_SPDIF_RX 0x5002
#define AFE_PORT_ID_SECONDARY_SPDIF_TX 0x5003
#define AFE_PORT_ID_SPDIF_RX AFE_PORT_ID_PRIMARY_SPDIF_RX
#define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000
#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001
#define AFE_PORT_ID_RT_PROXY_PORT_002_RX 0x2002
#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
/* SLIMbus Rx port on channel 7. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_RX 0x400e
/* SLIMbus Tx port on channel 7. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_TX 0x400f
/* SLIMbus Rx port on channel 8. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_RX 0x4010
/* SLIMbus Tx port on channel 8. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_TX 0x4011
/* SLIMbus Rx port on channel 9. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_9_RX 0x4012
/* SLIMbus Tx port on channel 9. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_9_TX 0x4013
/*AFE Rx port for audio over hdmi*/
#define AFE_PORT_ID_HDMI_MS 0x6002
/* AFE Rx port for audio over Display port */
#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020
/*USB AFE port */
#define AFE_PORT_ID_USB_RX 0x7000
#define AFE_PORT_ID_USB_TX 0x7001
/* AFE WSA Codec DMA Rx port 0 */
#define AFE_PORT_ID_WSA_CODEC_DMA_RX_0 0xB000
/* AFE WSA Codec DMA Tx port 0 */
#define AFE_PORT_ID_WSA_CODEC_DMA_TX_0 0xB001
/* AFE WSA Codec DMA Rx port 1 */
#define AFE_PORT_ID_WSA_CODEC_DMA_RX_1 0xB002
/* AFE WSA Codec DMA Tx port 1 */
#define AFE_PORT_ID_WSA_CODEC_DMA_TX_1 0xB003
/* AFE WSA Codec DMA Tx port 2 */
#define AFE_PORT_ID_WSA_CODEC_DMA_TX_2 0xB005
/* AFE VA Codec DMA Tx port 0 */
#define AFE_PORT_ID_VA_CODEC_DMA_TX_0 0xB021
/* AFE VA Codec DMA Tx port 1 */
#define AFE_PORT_ID_VA_CODEC_DMA_TX_1 0xB023
/* AFE VA Codec DMA Tx port 2 */
#define AFE_PORT_ID_VA_CODEC_DMA_TX_2 0xB025
/* AFE Rx Codec DMA Rx port 0 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_0 0xB030
/* AFE Tx Codec DMA Tx port 0 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_0 0xB031
/* AFE Rx Codec DMA Rx port 1 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_1 0xB032
/* AFE Tx Codec DMA Tx port 1 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_1 0xB033
/* AFE Rx Codec DMA Rx port 2 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_2 0xB034
/* AFE Tx Codec DMA Tx port 2 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_2 0xB035
/* AFE Rx Codec DMA Rx port 3 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_3 0xB036
/* AFE Tx Codec DMA Tx port 3 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_3 0xB037
/* AFE Rx Codec DMA Rx port 4 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_4 0xB038
/* AFE Tx Codec DMA Tx port 4 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_4 0xB039
/* AFE Rx Codec DMA Rx port 5 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_5 0xB03A
/* AFE Tx Codec DMA Tx port 5 */
#define AFE_PORT_ID_TX_CODEC_DMA_TX_5 0xB03B
/* AFE Rx Codec DMA Rx port 6 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_6 0xB03C
/* AFE Rx Codec DMA Rx port 7 */
#define AFE_PORT_ID_RX_CODEC_DMA_RX_7 0xB03E
/* 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
/*
* Proxyport used for voice call data processing.
* In cases like call-screening feature, where user can communicate
* with caller with the help of "call screen" mode, and without
* connecting the call with any HW input/output devices in the phone,
* voice call can use Proxy port to start voice data processing.
*/
#define RT_PROXY_PORT_002_TX 0x2003
#define RT_PROXY_PORT_002_RX 0x2002
#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_QUINARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x40)
#define AFE_PORT_ID_QUINARY_TDM_RX_1 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x02)
#define AFE_PORT_ID_QUINARY_TDM_RX_2 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x04)
#define AFE_PORT_ID_QUINARY_TDM_RX_3 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x06)
#define AFE_PORT_ID_QUINARY_TDM_RX_4 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x08)
#define AFE_PORT_ID_QUINARY_TDM_RX_5 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_QUINARY_TDM_RX_6 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_QUINARY_TDM_RX_7 \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_QUINARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x41)
#define AFE_PORT_ID_QUINARY_TDM_TX_1 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x02)
#define AFE_PORT_ID_QUINARY_TDM_TX_2 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x04)
#define AFE_PORT_ID_QUINARY_TDM_TX_3 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x06)
#define AFE_PORT_ID_QUINARY_TDM_TX_4 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x08)
#define AFE_PORT_ID_QUINARY_TDM_TX_5 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_QUINARY_TDM_TX_6 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_QUINARY_TDM_TX_7 \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x0E)
#define AFE_PORT_ID_SENARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x50)
#define AFE_PORT_ID_SENARY_TDM_RX_1 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x02)
#define AFE_PORT_ID_SENARY_TDM_RX_2 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x04)
#define AFE_PORT_ID_SENARY_TDM_RX_3 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x06)
#define AFE_PORT_ID_SENARY_TDM_RX_4 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x08)
#define AFE_PORT_ID_SENARY_TDM_RX_5 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_SENARY_TDM_RX_6 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_SENARY_TDM_RX_7 \
(AFE_PORT_ID_SENARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_SENARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x51)
#define AFE_PORT_ID_SENARY_TDM_TX_1 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x02)
#define AFE_PORT_ID_SENARY_TDM_TX_2 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x04)
#define AFE_PORT_ID_SENARY_TDM_TX_3 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x06)
#define AFE_PORT_ID_SENARY_TDM_TX_4 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x08)
#define AFE_PORT_ID_SENARY_TDM_TX_5 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_SENARY_TDM_TX_6 \
(AFE_PORT_ID_SENARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_SENARY_TDM_TX_7 \
(AFE_PORT_ID_SENARY_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
#define MAX_SIDETONE_IIR_DATA_SIZE 224
#define MAX_NO_IIR_FILTER_STAGE 10
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
*/
uint8_t iir_config[MAX_SIDETONE_IIR_DATA_SIZE];
} __packed;
#define AFE_MODULE_LOOPBACK 0x00010205
#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206
/* Used by RTAC */
struct afe_rtac_user_data_set_v2 {
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/* Actual size of the payload in bytes.
* This is used for parsing the parameter payload.
* Supported values: > 0
*/
u16 payload_size;
/* The header detailing the memory mapping for out of band. */
struct mem_mapping_hdr mem_hdr;
/* The parameter header for the parameter data to set */
struct param_hdr_v1 param_hdr;
/* The parameter data to be filled when sent inband */
u32 *param_data;
} __packed;
struct afe_rtac_user_data_set_v3 {
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/* Reserved for future enhancements. Must be 0. */
u16 reserved;
/* The header detailing the memory mapping for out of band. */
struct mem_mapping_hdr mem_hdr;
/* The size of the parameter header and parameter data */
u32 payload_size;
/* The parameter header for the parameter data to set */
struct param_hdr_v3 param_hdr;
/* The parameter data to be filled when sent inband */
u32 *param_data;
} __packed;
struct afe_rtac_user_data_get_v2 {
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/* Actual size of the payload in bytes.
* This is used for parsing the parameter payload.
* Supported values: > 0
*/
u16 payload_size;
/* The header detailing the memory mapping for out of band. */
struct mem_mapping_hdr mem_hdr;
/* The module ID of the parameter to get */
u32 module_id;
/* The parameter ID of the parameter to get */
u32 param_id;
/* The parameter data to be filled when sent inband */
struct param_hdr_v1 param_hdr;
} __packed;
struct afe_rtac_user_data_get_v3 {
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/* Reserved for future enhancements. Must be 0. */
u16 reserved;
/* The header detailing the memory mapping for out of band. */
struct mem_mapping_hdr mem_hdr;
/* The parameter data to be filled when sent inband */
struct param_hdr_v3 param_hdr;
} __packed;
#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
struct afe_port_cmd_set_param_v2 {
/* APR Header */
struct apr_hdr apr_hdr;
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/*
* Actual size of the payload in bytes.
* This is used for parsing the parameter payload.
* Supported values: > 0
*/
u16 payload_size;
/* The header detailing the memory mapping for out of band. */
struct mem_mapping_hdr mem_hdr;
/* The parameter data to be filled when sent inband */
u8 param_data[0];
} __packed;
#define AFE_PORT_CMD_SET_PARAM_V3 0x000100FA
struct afe_port_cmd_set_param_v3 {
/* APR Header */
struct apr_hdr apr_hdr;
/* Port ID of the AFE port to configure. Port interface and direction
* (Rx or Tx) to configure. An even number represents the Rx direction,
* and an odd number represents the Tx direction.
*/
u16 port_id;
/* Reserved. This field must be set to zero. */
u16 reserved;
/* The memory mapping header to be used when sending outband */
struct mem_mapping_hdr mem_hdr;
/* The total size of the payload, including param_hdr_v3 */
u32 payload_size;
/*
* The parameter data to be filled when sent inband.
* Must include param_hdr packed correctly.
*/
u8 param_data[0];
} __packed;
/* 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.
*/
struct afe_loopback_gain_per_path_param {
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 which 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 {
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;
struct afe_loopback_sidetone_gain {
u16 rx_port_id;
u16 gain;
} __packed;
struct afe_display_stream_idx {
u32 minor_version;
u32 stream_idx;
} __packed;
struct afe_display_ctl_idx {
u32 minor_version;
u32 ctl_idx;
} __packed;
struct loopback_cfg_data {
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;
struct afe_st_loopback_cfg_v1 {
struct apr_hdr hdr;
struct mem_mapping_hdr mem_hdr;
struct param_hdr_v1 gain_pdata;
struct afe_loopback_sidetone_gain gain_data;
struct param_hdr_v1 cfg_pdata;
struct loopback_cfg_data cfg_data;
} __packed;
struct afe_loopback_iir_cfg_v2 {
struct apr_hdr hdr;
struct mem_mapping_hdr param;
struct param_hdr_v1 st_iir_enable_pdata;
struct afe_mod_enable_param st_iir_mode_enable_data;
struct param_hdr_v1 st_iir_filter_config_pdata;
struct afe_sidetone_iir_filter_config_params st_iir_filter_config_data;
} __packed;
/*
* Param ID and related structures for AFE event
* registration.
*/
#define AFE_PORT_CMD_MOD_EVENT_CFG 0x000100FD
struct afe_port_cmd_event_cfg {
struct apr_hdr hdr;
uint32_t version;
/* Version number. The current version is 0 */
uint32_t port_id;
/*
* Port ID for the AFE port hosting the modules
* being registered for the events
*/
uint32_t num_events;
/*
* Number of events to be registered with the service
* Each event has the structure of
* afe_port_cmd_mod_evt_cfg_payload.
*/
uint8_t payload[0];
};
/** Event registration for a module. */
#define AFE_MODULE_REGISTER_EVENT_FLAG 1
/** Event de-registration for a module. */
#define AFE_MODULE_DEREGISTER_EVENT_FLAG 0
struct afe_port_cmd_mod_evt_cfg_payload {
uint32_t module_id;
/* Valid ID of the module. */
uint16_t instance_id;
/*
* Valid ID of the module instance in the current topology.
* If both module_id and instance_id ID are set to 0, the event is
* registered with all modules and instances in the topology.
* If module_id is set to 0 and instance_id is set to a non-zero value,
* the payload is considered to be invalid.
*/
uint16_t reserved;
/* Used for alignment; must be set to 0.*/
uint32_t event_id;
/* Unique ID of the event. */
uint32_t reg_flag;
/*
* Bit field for enabling or disabling event registration.
* values
* - #AFE_MODULE_REGISTER_EVENT_FLAG
* - #AFE_MODULE_DEREGISTER_EVENT_FLAG
*/
} __packed;
#define AFE_PORT_MOD_EVENT 0x0001010C
struct afe_port_mod_evt_rsp_hdr {
uint32_t minor_version;
/* This indicates the minor version of the payload */
uint32_t port_id;
/* AFE port hosting this module */
uint32_t module_id;
/* Module ID which is raising the event */
uint16_t instance_id;
/* Instance ID of the module which is raising the event */
uint16_t reserved;
/* For alignment purpose, should be set to 0 */
uint32_t event_id;
/* Valid event ID registered by client */
uint32_t payload_size;
/*
* Size of the event payload
* This is followed by actual payload corresponding to the event
*/
} __packed;
#define AFE_PORT_SP_DC_DETECTION_EVENT 0x0001010D
#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;
struct lpass_swr_spkr_dep_cfg_t {
uint32_t vbatt_pkd_reg_addr;
uint32_t temp_pkd_reg_addr;
uint32_t value_normal_thrsd[MAX_CPS_LEVELS];
uint32_t value_low1_thrsd[MAX_CPS_LEVELS];
uint32_t value_low2_thrsd[MAX_CPS_LEVELS];
} __packed;
struct lpass_swr_hw_reg_cfg_t {
uint32_t lpass_wr_cmd_reg_phy_addr;
uint32_t lpass_rd_cmd_reg_phy_addr;
uint32_t lpass_rd_fifo_reg_phy_addr;
uint32_t vbatt_lower1_threshold;
uint32_t vbatt_lower2_threshold;
uint32_t num_spkr;
} __packed;
struct afe_cps_hw_intf_cfg {
uint32_t lpass_hw_intf_cfg_mode;
struct lpass_swr_hw_reg_cfg_t hw_reg_cfg;
struct lpass_swr_spkr_dep_cfg_t *spkr_dep_cfg;
} __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 following 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_32K 32000
#define AFE_PORT_SAMPLE_RATE_44_1K 44100
#define AFE_PORT_SAMPLE_RATE_48K 48000
#define AFE_PORT_SAMPLE_RATE_96K 96000
#define AFE_PORT_SAMPLE_RATE_176P4K 176400
#define AFE_PORT_SAMPLE_RATE_192K 192000
#define AFE_PORT_SAMPLE_RATE_352P8K 352800
#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
#define AFE_GENERIC_COMPRESSED 0x8
#define AFE_LINEAR_PCM_DATA_PACKED_16BIT 0X6
#define AFE_DSD_DOP_W_MARKER_DATA 0x9
#define AFE_DSD_DATA 0xA
/* 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_10CHS 0x9
#define AFE_PORT_I2S_12CHS 0xA
#define AFE_PORT_I2S_14CHS 0xB
#define AFE_PORT_I2S_16CHS 0xC
#define AFE_PORT_I2S_SD4 0xD
#define AFE_PORT_I2S_SD5 0xE
#define AFE_PORT_I2S_SD6 0xF
#define AFE_PORT_I2S_SD7 0x10
#define AFE_PORT_I2S_QUAD45 0x11
#define AFE_PORT_I2S_QUAD67 0x12
#define AFE_PORT_I2S_8CHS_2 0x13
#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
* - #AFE_PORT_I2S_10CHS
* - #AFE_PORT_I2S_12CHS
* - #AFE_PORT_I2S_14CHS
* - #AFE_PORT_I2S_16CHS
* - #AFE_PORT_I2S_SD4
* - #AFE_PORT_I2S_SD5
* - #AFE_PORT_I2S_SD6
* - #AFE_PORT_I2S_SD7
* - #AFE_PORT_I2S_QUAD45
* - #AFE_PORT_I2S_QUAD67
* - #AFE_PORT_I2S_8CHS_2
*/
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
* - #AFE_DSD_DOP_W_MARKER_DATA
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
/* This param id is used to configure META I2S interface */
#define AFE_PARAM_ID_META_I2S_CONFIG 0x000102C5
#define AFE_API_VERSION_META_I2S_CONFIG 0x1
#define MAX_NUM_I2S_META_PORT_MEMBER_PORTS 4
/* Payload of the #AFE_PARAM_ID_META_I2S_CONFIG
* command's (I2S configuration
* parameter).
*/
struct afe_param_id_meta_i2s_cfg {
u32 minor_version;
/* Minor version used for tracking the version of the I2S
* configuration interface.
* Supported values: #AFE_API_VERSION_META_I2S_CONFIG
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
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 member_port_id[MAX_NUM_I2S_META_PORT_MEMBER_PORTS];
/* Array of member port IDs in this meta device.
* Supported values:
* - #AFE_PORT_ID_PRIMARY_MI2S_RX
* - #AFE_PORT_ID_SECONDARY_MI2S_RX
* - #AFE_PORT_ID_TERTIARY_MI2S_RX
* - #AFE_PORT_ID_QUATERNY_MI2S_RX
* - #AFE_PORT_ID_INVALID
*
* Fill these values from index 0. Set unused index to AFE_PORT_ID_INVALID.
*
* Note:
* the first member port will act as WS master in case
* meta port ws_src is configured as AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL.
* In all other cases member ports will act as slave.
* This must be considered when HLOS enables the interface clocks
*/
u16 member_port_channel_mode[MAX_NUM_I2S_META_PORT_MEMBER_PORTS];
/* I2S lines and multichannel operation per member port.
* The sequence matches the sequence in member_port_id,
* value will be ignored if member port is set to AFE_PORT_ID_INVALID
*
* 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
* - #AFE_PORT_I2S_10CHS
* - #AFE_PORT_I2S_12CHS
* - #AFE_PORT_I2S_14CHS
* - #AFE_PORT_I2S_16CHS
* - #AFE_PORT_I2S_SD4
* - #AFE_PORT_I2S_SD5
* - #AFE_PORT_I2S_SD6
* - #AFE_PORT_I2S_SD7
* - #AFE_PORT_I2S_QUAD45
* - #AFE_PORT_I2S_QUAD67
* - #AFE_PORT_I2S_8CHS_2
*/
u16 data_format;
/* data format
* Supported values:
* - #LINEAR_PCM_DATA
*/
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_V2 0x2
#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
#define AFE_MODULE_CUSTOM_EVENTS 0x00010251
#define AFE_PORT_FMT_UPDATE_EVENT 0x0001010E
#define AFE_API_VERSION_EVENT_FMT_UPDATE 0x1
#define AFE_PORT_STATUS_NO_SIGNAL 0
#define AFE_PORT_STATUS_AUDIO_ACTIVE 1
#define AFE_PORT_STATUS_AUDIO_EOS 2
struct afe_param_id_spdif_cfg_v2 {
/* Minor version used for tracking the version of the SPDIF
* configuration interface.
* Supported values: #AFE_API_VERSION_SPDIF_CONFIG,
* #AFE_API_VERSION_SPDIF_CONFIG_V2
*/
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;
/* Input select for spdif input, must be set to 0 for spdif output. */
u32 src_sel;
} __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;
/* deprecated */
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_event_fmt_update {
/* Tracks the configuration of this event. */
u32 minor_version;
/* Detected port status.
* Supported values:
* - #AFE_PORT_STATUS_NO_SIGNAL
* - #AFE_PORT_STATUS_AUDIO_ACTIVE
* - #AFE_PORT_STATUS_AUDIO_EOS
*/
u32 status;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_44_1K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_88_2K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_176_4K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
u32 sample_rate;
/* Data format of the port.
* Supported values:
* - #AFE_LINEAR_PCM_DATA
* - #AFE_NON_LINEAR_DATA
*/
u16 data_format;
/* First 6 bytes of channel status bits */
u8 channel_status[6];
} __packed;
struct afe_spdif_port_config {
struct afe_param_id_spdif_cfg_v2 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
#define AFE_PARAM_ID_HDMI_DP_MST_VID_IDX_CFG 0x000102b5
#define AFE_PARAM_ID_HDMI_DPTX_IDX_CFG 0x000102b6
/* 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.
*/
/* ID of the parameter used to set the latency mode of the
* USB audio device.
*/
#define AFE_PARAM_ID_PORT_LATENCY_MODE_CONFIG 0x000102B3
/* Minor version used for tracking USB audio latency mode */
#define AFE_API_MINOR_VERSION_USB_AUDIO_LATENCY_MODE 0x1
/* Supported AFE port latency modes */
#define AFE_PORT_DEFAULT_LATENCY_MODE 0x0
#define AFE_PORT_LOW_LATENCY_MODE 0x1
#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
#define AFE_PORT_MAX_AUDIO_CHAN_CNT_V2 0x20
/* 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;
/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS to configure
* USB audio device parameter. It should be used with
* AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5
/* ID of the parameter used to set the endianness value for the
* USB audio device. It should be used with
* AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA
/* Minor version used for tracking USB audio configuration */
#define AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG 0x1
/* Payload of the AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_usb_audio_dev_params {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Token of actual end USB aduio device */
u32 dev_token;
} __packed;
struct afe_param_id_usb_audio_dev_lpcm_fmt {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Endianness of actual end USB audio device */
u32 endian;
} __packed;
struct afe_param_id_usb_audio_dev_latency_mode {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_LATENCY_MODE
*/
u32 minor_version;
/* latency mode for the USB audio device */
u32 mode;
} __packed;
#define AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL 0x000102B7
struct afe_param_id_usb_audio_svc_interval {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Endianness of actual end USB audio device */
u32 svc_interval;
} __packed;
/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_CONFIG to configure
* USB audio interface. It should be used with AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4
/* Payload of the AFE_PARAM_ID_USB_AUDIO_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_usb_audio_cfg {
/* Minor version used for tracking USB audio device configuration.
* Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Sampling rate of the port.
* Supported values:
* - AFE_PORT_SAMPLE_RATE_8K
* - AFE_PORT_SAMPLE_RATE_11025
* - AFE_PORT_SAMPLE_RATE_12K
* - AFE_PORT_SAMPLE_RATE_16K
* - AFE_PORT_SAMPLE_RATE_22050
* - AFE_PORT_SAMPLE_RATE_24K
* - AFE_PORT_SAMPLE_RATE_32K
* - AFE_PORT_SAMPLE_RATE_44P1K
* - AFE_PORT_SAMPLE_RATE_48K
* - AFE_PORT_SAMPLE_RATE_96K
* - AFE_PORT_SAMPLE_RATE_192K
*/
u32 sample_rate;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 bit_width;
/* Number of channels.
* Supported values: 1 and 2
*/
u16 num_channels;
/* Data format supported by the USB. The supported value is
* 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM).
*/
u16 data_format;
/* this field must be 0 */
u16 reserved;
/* device token of actual end USB aduio device */
u32 dev_token;
/* endianness of this interface */
u32 endian;
/* service interval */
u32 service_interval;
} __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
* - #AFE_PORT_SAMPLE_RATE_176P4K
* - #AFE_PORT_SAMPLE_RATE_352P8K @tablebulletend
*/
u32 bit_width;
/* < Bit width of the sample.
* @values 16, 24
*/
u16 data_format;
/* < Data format: linear ,compressed, generic compresssed
* @values
* - #AFE_LINEAR_PCM_DATA
* - #AFE_NON_LINEAR_DATA
* - #AFE_GENERIC_COMPRESSED
*/
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
/** Version information used to handle future additions to slot mapping
* configuration support 32 channels.
*/
#define AFE_API_VERSION_SLOT_MAPPING_CONFIG_V2 0x2
/* 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;
/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG_V2
* command's TDM configuration parameter.
*/
struct afe_param_id_slot_mapping_cfg_v2 {
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, 16, 32 @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_V2];
/**< 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_tdm_port_config {
struct afe_param_id_tdm_cfg tdm;
struct afe_param_id_slot_mapping_cfg slot_mapping;
struct afe_param_id_slot_mapping_cfg_v2 slot_mapping_v2;
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_PARAM_ID_DEREGISTER_TOPOLOGY 0x000102E8
#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;
#define AFE_PARAM_ID_CODEC_DMA_DATA_ALIGN 0x000102EA
struct afe_param_id_cdc_dma_data_align {
uint32_t cdc_dma_data_align;
} __packed;
#define MAX_ABR_LEVELS 5
struct afe_bit_rate_level_map_t {
/*
* Key value pair for link quality level to bitrate
* mapping in AFE
*/
uint32_t link_quality_level;
uint32_t bitrate;
} __packed;
struct afe_quality_level_to_bitrate_info {
/*
* Number of quality levels being mapped.
* This will be equal to the size of mapping table.
*/
uint32_t num_levels;
/*
* Quality level to bitrate mapping table
*/
struct afe_bit_rate_level_map_t bit_rate_level_map[MAX_ABR_LEVELS];
} __packed;
struct afe_imc_dec_enc_info {
/*
* Decoder to encoder communication direction.
* Transmit = 0 / Receive = 1
*/
uint32_t direction;
/*
* Enable / disable IMC between decoder and encoder
*/
uint32_t enable;
/*
* Purpose of IMC being set up between decoder and encoder.
* Param ID defined for link quality feedback in LPASS will
* be the default value sent as purpose.
* Supported values:
* AFE_ENCDEC_PURPOSE_ID_BT_INFO
*/
uint32_t purpose;
/*
* Unique communication instance ID.
* Data type a2dp_abr_instance used to set instance ID.
* purpose and comm_instance together form the actual key
* used in IMC registration, which must be the same for
* encoder and decoder for which IMC is being set up.
*/
uint32_t comm_instance;
} __packed;
struct afe_abr_dec_cfg_t {
struct afe_imc_dec_enc_info imc_info;
bool is_abr_enabled;
} __packed;
struct afe_abr_enc_cfg_t {
/*
* Link quality level to bitrate mapping info sent to DSP.
*/
struct afe_quality_level_to_bitrate_info mapping_info;
/*
* Information to set up IMC between decoder and encoder.
*/
struct afe_imc_dec_enc_info imc_info;
/*
* Flag to indicate whether ABR is enabled.
*/
bool is_abr_enabled;
} __packed;
#define AFE_PARAM_ID_APTX_SYNC_MODE 0x00013205
struct afe_param_id_aptx_sync_mode {
/*
* sync mode: 0x0 = stereo sync mode (default)
* 0x01 = dual mono sync mode
* 0x02 = dual mono with no sync on either L or R
*/
uint32_t sync_mode;
} __packed;
#define AFE_ID_APTX_ADAPTIVE_ENC_INIT 0x00013324
struct afe_id_aptx_adaptive_enc_init
{
uint32_t sampling_freq;
uint32_t mtu;
uint32_t channel_mode;
uint32_t min_sink_modeA;
uint32_t max_sink_modeA;
uint32_t min_sink_modeB;
uint32_t max_sink_modeB;
uint32_t min_sink_modeC;
uint32_t max_sink_modeC;
uint32_t mode;
uint32_t input_mode;
uint32_t fade_duration;
uint8_t sink_cap[11];
} __attribute__ ((packed));
/*
* Generic encoder module ID.
* This module supports the following parameter IDs:
* #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time)
* #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time)
* #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time)
* #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time)
* Opcode - AVS_MODULE_ID_ENCODER
* AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID.
*/
#define AFE_MODULE_ID_ENCODER 0x00013229
/* Macro for defining the packetizer ID: COP. */
#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A
/* Macro for defining the packetizer ID: COP V2 */
#define AFE_MODULE_ID_PACKETIZER_COP_V2 0x000132F9
/*
* Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter cannot be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E
/*
* Encoder config block parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter may be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C
/*
* Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter cannot be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID 0x0001322B
/*
* Decoder format ID parameter for the #AVS_MODULE_ID_DECODER module.
* This parameter cannot be set runtime.
*/
#define AFE_DECODER_PARAM_ID_DEC_FMT_ID 0x00013234
/*
* Encoder scrambler parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter cannot be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_ENABLE_SCRAMBLING 0x0001323C
/*
* Link quality level to bitrate mapping info sent to AFE Encoder.
* This parameter may be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_BIT_RATE_LEVEL_MAP 0x000132E1
/*
* Parameter to set up Inter Module Communication (IMC) between
* AFE Decoder and Encoder.
* This parameter may be set runtime.
*/
#define AFE_ENCDEC_PARAM_ID_DEC_TO_ENC_COMMUNICATION 0x0001323D
/*
* This is needed to be used only for SWB voice call use case.
* This is needed to be issued for each direction (RX AFE and TX AFE)
* along with AFE_PARAM_ID_PORT_MEDIA_TYPE
* (Issuing AF_PARAM_ID_RATE_MATCHED_PORT param alone is not useful).
*/
#define AFE_PARAM_ID_RATE_MATCHED_PORT 0x000102BE
/*
* Purpose of IMC set up between encoder and decoder.
* Communication instance and purpose together form the
* actual key used for IMC registration.
*/
#define AFE_ENCDEC_PURPOSE_ID_BT_INFO 0x000132E2
#define AFE_MODULE_ID_DECODER 0x00013231
/*
* Macro for defining the depacketizer ID: COP.
*/
#define AFE_MODULE_ID_DEPACKETIZER_COP 0x00013233
#define AFE_MODULE_ID_DEPACKETIZER_COP_V1 0x000132E9
#define AFE_MODULE_ID_DEPACKETIZER_COP_V2 0x000132FC
/* Macros for dynamic loading of modules by AVCS */
#define AVS_MODULE_ID_PACKETIZER_COP 0x0001322A
#define AVS_MODULE_ID_PACKETIZER_COP_V1 0x000132E8
#define AVS_MODULE_ID_PACKETIZER_COP_V2 0x000132F9
#define AVS_MODULE_ID_DEPACKETIZER_COP 0x00013233
#define AVS_MODULE_ID_DEPACKETIZER_COP_V1 0x000132E9
#define AVS_MODULE_ID_DEPACKETIZER_COP_V2 0x000132FC
/*
* Depacketizer and packetizer type parameter for the
* #AVS_MODULE_ID_DEPACKETIZER_COP_V2 module and
* #AVS_MODULE_ID_PACKETIZER_COP_V2 module.
*/
#define AVS_COP_V2_PARAM_ID_STREAM_INFO 0x000132FD
/*
* Depacketizer type parameter for the #AVS_MODULE_ID_DECODER module.
* This parameter cannot be set runtime.
*/
#define AFE_DECODER_PARAM_ID_DEPACKETIZER_ID 0x00013235
#define CAPI_V2_PARAM_ID_APTX_ENC_SWITCH_TO_MONO 0x0001332A
#define CAPI_V2_PARAM_ID_APTX_AD_ENC_SWITCH_TO_MONO 0x00013354
struct aptx_channel_mode_param_t {
u32 channel_mode;
} __packed;
#define CAPI_V2_PARAM_SET_LC3_ENC_DOWNMIX_2_MONO 0x00013384
struct lc3_channel_mode_param_t {
u32 channel_mode;
} __packed;
/*
* Decoder buffer ID parameter for the #AVS_MODULE_ID_DECODER module.
* This parameter cannot be set runtime.
*/
#define AFE_DECODER_PARAM_ID_CONGESTION_BUFFER_SIZE 0x000132ec
/*
* Data format to send compressed data
* is transmitted/received over Slimbus lines.
*/
#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED 0x3
/*
* Parameter to send frame control size
* to DSP for AAC encoder in AFE.
*/
#define AFE_PARAM_ID_AAC_FRM_SIZE_CONTROL 0x000132EA
/*
* ID for AFE port module. This will be used to define port properties.
* This module supports following parameter IDs:
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
* To configure the port property, the client must use the
* #AFE_PORT_CMD_SET_PARAM_V2 command,
* and fill the module ID with the respective parameter IDs as listed above.
* @apr_hdr_fields
* Opcode -- AFE_MODULE_PORT
*/
#define AFE_MODULE_PORT 0x000102a6
/*
* ID of the parameter used by #AFE_MODULE_PORT to set the port media type.
* parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command.
*/
#define AFE_PARAM_ID_PORT_MEDIA_TYPE 0x000102a7
/*
* Macros for defining the "data_format" field in the
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
*/
#define AFE_PORT_DATA_FORMAT_PCM 0x0
#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED 0x1
/*
* Macro for defining the "minor_version" field in the
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
*/
#define AFE_API_VERSION_PORT_MEDIA_TYPE 0x1
#define ASM_MEDIA_FMT_NONE 0x0
/*
* Media format ID for SBC encode configuration.
* @par SBC encode configuration (asm_sbc_enc_cfg_t)
* @table{weak__asm__sbc__enc__cfg__t}
*/
#define ASM_MEDIA_FMT_SBC 0x00010BF2
/* SBC channel Mono mode.*/
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO 1
/* SBC channel Stereo mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO 2
/* SBC channel Dual Mono mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO 8
/* SBC channel Joint Stereo mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9
/* SBC bit allocation method = loudness. */
#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0
/* SBC bit allocation method = SNR. */
#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1
/*
* Payload of the SBC encoder configuration parameters in the
* #ASM_MEDIA_FMT_SBC media format.
*/
struct asm_sbc_enc_cfg_t {
/*
* Number of subbands.
* @values 4, 8
*/
uint32_t num_subbands;
/*
* Size of the encoded block in samples.
* @values 4, 8, 12, 16
*/
uint32_t blk_len;
/*
* Mode used to allocate bits between channels.
* @values
* 0 (Native mode)
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO
* Native mode indicates that encoding must be performed with the number
* of channels at the input.
* If postprocessing outputs one-channel data, Mono mode is used. If
* postprocessing outputs two-channel data, Stereo mode is used.
* The number of channels must not change during encoding.
*/
uint32_t channel_mode;
/*
* Encoder bit allocation method.
* @values
* #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS
* #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend
*/
uint32_t alloc_method;
/*
* Number of encoded bits per second.
* @values
* Mono channel -- Maximum of 320 kbps
* Stereo channel -- Maximum of 512 kbps @tablebulletend
*/
uint32_t bit_rate;
/*
* Number of samples per second.
* @values 0 (Native mode), 16000, 32000, 44100, 48000&nbsp;Hz
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
uint32_t sample_rate;
};
#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_FORMAT_FLAG_ADTS 0
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
struct asm_aac_enc_cfg_v2_t {
/* Encoding rate in bits per second.*/
uint32_t bit_rate;
/*
* Encoding mode.
* Supported values:
* #ASM_MEDIA_FMT_AAC_AOT_LC
* #ASM_MEDIA_FMT_AAC_AOT_SBR
* #ASM_MEDIA_FMT_AAC_AOT_PS
*/
uint32_t enc_mode;
/*
* AAC format flag.
* Supported values:
* #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
uint16_t aac_fmt_flag;
/*
* 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.
*/
uint16_t channel_cfg;
/*
* 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.
*/
uint32_t sample_rate;
} __packed;
/* Structure to control frame size of AAC encoded frames. */
struct asm_aac_frame_size_control_t {
/* Type of frame size control: MTU_SIZE / PEAK_BIT_RATE*/
uint32_t ctl_type;
/*
* Control value
* MTU_SIZE: MTU size in bytes
* PEAK_BIT_RATE: Peak bitrate in bits per second.
*/
uint32_t ctl_value;
} __packed;
struct asm_aac_enc_cfg_t {
struct asm_aac_enc_cfg_v2_t aac_cfg;
struct asm_aac_frame_size_control_t frame_ctl;
struct asm_aac_frame_size_control_t frame_ctl_v2;
} __packed;
/* FMT ID for apt-X Classic */
#define ASM_MEDIA_FMT_APTX 0x000131ff
/* FMT ID for apt-X HD */
#define ASM_MEDIA_FMT_APTX_HD 0x00013200
/* FMT ID for apt-X Adaptive */
#define ASM_MEDIA_FMT_APTX_ADAPTIVE 0x00013204
/* FMT ID for apt-X Adaptive speech */
#define ASM_MEDIA_FMT_APTX_AD_SPEECH 0x00013208
/* FMT ID for lc3 codec */
#define ASM_MEDIA_FMT_LC3 0x0001337E
#define ENC_CODEC_TYPE_LC3 0x2B000000
#define PCM_CHANNEL_L 1
#define PCM_CHANNEL_R 2
#define PCM_CHANNEL_C 3
struct asm_custom_enc_cfg_t {
uint32_t sample_rate;
/* Mono or stereo */
uint16_t num_channels;
uint16_t reserved;
/* num_ch == 1, then PCM_CHANNEL_C,
* num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
*/
uint8_t channel_mapping[8];
uint32_t custom_size;
} __packed;
struct asm_aptx_ad_speech_mode_cfg_t
{
uint32_t speech_mode;
/*
* speech mode of codec.
*
* @values 0x0(swb), 0x4(sswb)
*/
uint32_t swapping;
/*
* byte swapping of codec.
*
* @values 0x1, enable swapping
*/
} __packed;
struct asm_aptx_v2_enc_cfg_ext_t {
/*
* sync mode: 0x0 = stereo sync mode (default)
* 0x01 = dual mono sync mode
* 0x02 = dual mono with no sync on either L or R
*/
uint32_t sync_mode;
} __packed;
struct asm_aptx_enc_cfg_t {
struct asm_custom_enc_cfg_t custom_cfg;
struct asm_aptx_v2_enc_cfg_ext_t aptx_v2_cfg;
} __packed;
struct asm_aptx_ad_enc_cfg_t
{
struct asm_custom_enc_cfg_t custom_cfg;
struct afe_id_aptx_adaptive_enc_init aptx_ad_cfg;
struct afe_abr_enc_cfg_t abr_cfg;
} __attribute__ ((packed));
struct asm_aptx_ad_speech_enc_cfg_t
{
struct asm_custom_enc_cfg_t custom_cfg;
struct afe_imc_dec_enc_info imc_info;
struct asm_aptx_ad_speech_mode_cfg_t speech_mode;
} __attribute__ ((packed));
#define CAPI_V2_PARAM_LC3_ENC_INIT 0x00013381
#define CAPI_V2_PARAM_LC3_DEC_MODULE_INIT 0x00013391
struct afe_lc3_stream_map_t {
uint32_t stream_id;
uint32_t direction;
uint32_t channel_mask_lsw;
uint32_t channel_mask_msw;
} __packed;
struct afe_stream_map_t {
uint32_t audio_location;
uint8_t stream_id;
uint8_t direction;
} __packed;
struct afe_lc3_cfg_t {
uint32_t api_version;
uint32_t sampling_freq;
uint32_t max_octets_per_frame;
uint32_t frame_duration;
uint32_t bit_depth;
uint32_t num_blocks;
uint8_t default_q_level;
uint8_t vendor_specific[16];
uint32_t mode;
} __packed;
struct afe_lc3_enc_cfg_t {
struct afe_lc3_cfg_t toAirConfig;
uint32_t stream_map_size;
struct afe_stream_map_t streamMapOut[16];
} __packed;
struct afe_lc3_dec_cfg_t {
struct afe_lc3_cfg_t FromAir;
uint32_t decoder_output_channel;
uint32_t stream_map_size;
struct afe_stream_map_t streamMapIn[16];
} __packed;
struct avs_cop_v2_param_id_stream_info_t {
uint32_t stream_map_size;
struct afe_lc3_stream_map_t streamMap[16];
} __packed;
struct afe_lc3_dec_config_t {
struct avs_cop_v2_param_id_stream_info_t streamMapToAir;
} __packed;
struct afe_lc3_enc_config_t {
struct afe_lc3_enc_cfg_t to_Air_cfg;
struct avs_cop_v2_param_id_stream_info_t streamMapToAir;
} __packed;
struct asm_enc_lc3_cfg_t {
struct afe_imc_dec_enc_info imc_info;
struct afe_lc3_enc_config_t enc_codec;
} __packed;
struct asm_lc3_dec_cfg_t {
struct afe_lc3_dec_config_t dec_codec;
} __packed;
struct afe_matched_port_t
{
uint32_t minor_version;
uint32_t enable;
} __attribute__ ((packed));
#define ASM_MEDIA_FMT_CELT 0x00013221
struct asm_celt_specific_enc_cfg_t {
/*
* Bit rate used for encoding.
* This is used to calculate the upper threshold
* for bytes per frame if vbr_flag is 1.
* Or else, this will be used as a regular constant
* bit rate for encoder output.
* @Range : 32000 to 1536000
* @Default: 128
*/
uint32_t bit_rate;
/*
* Frame size used for encoding.
* @Range : 64, 128, 256, 512
* @Default: 256
*/
uint16_t frame_size;
/*
* complexity of algorithm.
* @Range : 0-10
* @Default: 3
*/
uint16_t complexity;
/*
* Switch variable for prediction feature.
* Used to choose between the level of interframe
* predictions allowed while encoding.
* @Range:
* 0: Independent Frames.
* 1: Short Term interframe prediction allowed.
* 2: Long term prediction allowed.
* @Default: 2
*/
uint16_t prediction_mode;
/*
* Variable Bit Rate flag.
* @Default: 0
*/
uint16_t vbr_flag;
} __packed;
struct asm_celt_enc_cfg_t {
struct asm_custom_enc_cfg_t custom_config;
struct asm_celt_specific_enc_cfg_t celt_specific_config;
} __packed;
#define ASM_MEDIA_FMT_LDAC 0x00013224
#define ENC_CODEC_TYPE_LDAC 0x23000000
struct asm_ldac_specific_enc_cfg_t {
/*
* This is used to calculate the encoder output
* bytes per frame (i.e. bytes per packet).
* Bit rate also configures the EQMID.
* The min bit rate 303000 bps is calculated for
* 44.1 kHz and 88.2 KHz sampling frequencies with
* Mobile use Quality.
* The max bit rate of 990000 bps is calculated for
* 96kHz and 48 KHz with High Quality
* @Range(in bits per second)
* 303000 for Mobile use Quality
* 606000 for standard Quality
* 909000 for High Quality
*/
uint32_t bit_rate;
/*
* The channel setting information for LDAC specification
* of Bluetooth A2DP which is determined by SRC and SNK
* devices in Bluetooth transmission.
* @Range:
* 0 for native mode
* 4 for mono
* 2 for dual channel
* 1 for stereo
*/
uint16_t channel_mode;
/*
* Maximum Transmission Unit (MTU).
* The minimum MTU that a L2CAP implementation for LDAC shall
* support is 679 bytes, because LDAC is optimized with 2-DH5
* packet as its target.
* @Range : 679
* @Default: 679 for LDACBT_MTU_2DH5
*/
uint16_t mtu;
} __packed;
struct asm_ldac_enc_cfg_t {
struct asm_custom_enc_cfg_t custom_config;
struct asm_ldac_specific_enc_cfg_t ldac_specific_config;
struct afe_abr_enc_cfg_t abr_config;
} __packed;
struct afe_enc_fmt_id_param_t {
/*
* Supported values:
* #ASM_MEDIA_FMT_SBC
* #ASM_MEDIA_FMT_AAC_V2
* Any OpenDSP supported values
*/
uint32_t fmt_id;
} __packed;
struct afe_port_media_type_t {
/*
* Minor version
* @values #AFE_API_VERSION_PORT_MEDIA_TYPE.
*/
uint32_t minor_version;
/*
* Sampling rate of the port.
* @values
* #AFE_PORT_SAMPLE_RATE_8K
* #AFE_PORT_SAMPLE_RATE_11_025K
* #AFE_PORT_SAMPLE_RATE_12K
* #AFE_PORT_SAMPLE_RATE_16K
* #AFE_PORT_SAMPLE_RATE_22_05K
* #AFE_PORT_SAMPLE_RATE_24K
* #AFE_PORT_SAMPLE_RATE_32K
* #AFE_PORT_SAMPLE_RATE_44_1K
* #AFE_PORT_SAMPLE_RATE_48K
* #AFE_PORT_SAMPLE_RATE_88_2K
* #AFE_PORT_SAMPLE_RATE_96K
* #AFE_PORT_SAMPLE_RATE_176_4K
* #AFE_PORT_SAMPLE_RATE_192K
* #AFE_PORT_SAMPLE_RATE_352_8K
* #AFE_PORT_SAMPLE_RATE_384K
*/
uint32_t sample_rate;
/*
* Bit width of the sample.
* @values 16, 24
*/
uint16_t bit_width;
/*
* Number of channels.
* @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
uint16_t num_channels;
/*
* Data format supported by this port.
* If the port media type and device media type are different,
* it signifies a encoding/decoding use case
* @values
* #AFE_PORT_DATA_FORMAT_PCM
* #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED
*/
uint16_t data_format;
/*This field must be set to zero.*/
uint16_t reserved;
} __packed;
/*
* Payload of the SBC decoder configuration parameters in the
* #ASM_MEDIA_FMT_SBC media format.
*/
struct asm_sbc_dec_cfg_t {
uint16_t channels;
/*
* Number of channels present in the SBC stream.
*
* @values
* - 1 -- Mono
* - 2 -- Stereo
*/
uint32_t sample_rate;
/*
* Number of samples per second.
*
* @values 8000, 11025, 12000, 16000, 22050, 24000, 32000,
* 44100, 48000, 64000, 88200, 96000 Hz
*/
} __packed;
/*
* Payload of the MP3 decoder configuration parameters in the
* #ASM_MEDIA_FMT_MP3 media format.
*/
struct asm_mp3_dec_cfg_t {
/* All configuration is extracted from the stream */
} __packed;
struct asm_aac_dec_cfg_v2_t {
uint16_t aac_fmt_flag;
/*
* Bit stream format option.
*
* @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
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM
*/
uint16_t audio_obj_type;
/*
* Audio Object Type (AOT) present in the AAC stream.
*
* @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
*
* Other values are not supported.
*/
uint16_t channel_config;
/*
* Number of channels present in the AAC stream.
*
* @values
* - 0 -- PCE
* - 1 -- Mono
* - 2 -- Stereo
* - 6 -- 5.1 content
*/
uint16_t total_size_of_PCE_bits;
/*
* For RAW formats and if channel_config=0 (PCE),
* the client can send the bit stream containing PCE
* immediately following this structure (in band).
*
* @values @ge 0 (does not include the bits required
* for 32-bit alignment)
*
* If this field is set to 0, the PCE information is
* assumed to be available in the audio bit stream
* and not in band.
*
* If this field is greater than 0, the PCE information
* follows this structure. Additional bits might
* be required for 32-bit alignment.
*/
uint32_t sample_rate;
/*
* Number of samples per second.
*
* @values 8000, 11025, 12000, 16000, 22050, 24000, 32000,
* 44100, 48000, 64000, 88200, 96000 Hz
*
* This field must be equal to the sample rate of the
* AAC-LC decoder output.
* - For MP4 or 3GP containers, this sample rate
* is indicated by the
* samplingFrequencyIndex field in the
* AudioSpecificConfig element.
* - For ADTS format, this sample rate is indicated by the
* samplingFrequencyIndex in the ADTS fixed header.
* - For ADIF format, this sample rate is indicated by the
* samplingFrequencyIndex in the program_config_element
* present in the ADIF header.
*/
} __packed;
/*
* Payload of the APTX AD decoder configuration parameters in the
* #ASM_MEDIA_FMT_APTX_ADAPTIVE media format.
*/
struct asm_aptx_ad_dec_cfg_t {
uint32_t sample_rate;
/*
* Number of samples per second.
*
* @values 0x0(48000Hz), 0x1(44100Hz)
*/
} __packed;
struct asm_aptx_ad_speech_dec_cfg_t {
struct asm_aptx_ad_speech_mode_cfg_t speech_mode;
};
union afe_enc_config_data {
struct asm_sbc_enc_cfg_t sbc_config;
struct asm_aac_enc_cfg_t aac_config;
struct asm_custom_enc_cfg_t custom_config;
struct asm_celt_enc_cfg_t celt_config;
struct asm_aptx_enc_cfg_t aptx_config;
struct asm_ldac_enc_cfg_t ldac_config;
struct asm_aptx_ad_enc_cfg_t aptx_ad_config;
struct asm_aptx_ad_speech_enc_cfg_t aptx_ad_speech_config;
struct asm_enc_lc3_cfg_t lc3_enc_config;
};
struct afe_enc_config {
u32 format;
u32 scrambler_mode;
u32 mono_mode;
u32 lc3_mono_mode;
union afe_enc_config_data data;
};
/*
* Enable TTP generator in AFE.
*/
#define AVS_DEPACKETIZER_PARAM_ID_TTP_GEN_STATE 0x000132EF
/*
* Configure TTP generator params in AFE.
*/
#define AVS_DEPACKETIZER_PARAM_ID_TTP_GEN_CFG 0x000132F0
#define MAX_TTP_OFFSET_PAIRS 4
struct afe_ttp_gen_enable_t {
uint16_t enable;
uint16_t reserved;
} __packed;
struct afe_ttp_ssrc_offset_pair_t {
uint32_t ssrc;
uint32_t offset;
} __packed;
struct afe_ttp_gen_cfg_t {
uint32_t ttp_offset_default;
/*
* TTP offset uses for all other cases
* where no valid SSRC is received.
*/
uint32_t settling_time;
/*
* If settling_mode==0x00: time in [us]
* after first received packet until
* packets are no longer dropped.
*/
uint16_t settling_mode;
/*
* 0x00(Drop), 0x01(Settle)
*/
uint16_t num_ssrc_offsets;
/*
* Number of SSRC/TTPOFFSET pairs to follow
*/
struct afe_ttp_ssrc_offset_pair_t ssrc_ttp_offset[MAX_TTP_OFFSET_PAIRS];
/*
* Array of ssrc/offset pairs
*/
} __packed;
struct afe_ttp_config {
struct afe_ttp_gen_enable_t ttp_gen_enable;
struct afe_ttp_gen_cfg_t ttp_gen_cfg;
};
union afe_dec_config_data {
struct asm_sbc_dec_cfg_t sbc_config;
struct asm_aac_dec_cfg_v2_t aac_config;
struct asm_mp3_dec_cfg_t mp3_config;
struct asm_aptx_ad_dec_cfg_t aptx_ad_config;
struct asm_aptx_ad_speech_dec_cfg_t aptx_ad_speech_config;
struct asm_lc3_dec_cfg_t lc3_dec_config;
};
struct afe_dec_config {
u32 format;
struct afe_abr_dec_cfg_t abr_dec_cfg;
union afe_dec_config_data data;
};
struct afe_enc_cfg_blk_param_t {
uint32_t enc_cfg_blk_size;
/*
*Size of the encoder configuration block that follows this member
*/
union afe_enc_config_data enc_blk_config;
};
struct afe_enc_aptx_ad_speech_cfg_blk_param_t {
uint32_t enc_cfg_blk_size;
/*
* Size of the encoder configuration block that follows this member
*/
struct asm_custom_enc_cfg_t custom_cfg;
};
/*
* Payload of the AVS_DECODER_PARAM_ID_DEC_MEDIA_FMT parameter used by
* AVS_MODULE_ID_DECODER.
*/
struct afe_dec_media_fmt_t {
union afe_dec_config_data dec_media_config;
} __packed;
/*
* Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
*/
struct avs_enc_packetizer_id_param_t {
/*
* Supported values:
* #AVS_MODULE_ID_PACKETIZER_COP
* Any OpenDSP supported values
*/
uint32_t enc_packetizer_id;
};
/*
* Payload of the AVS_ENCODER_PARAM_ID_ENABLE_SCRAMBLING parameter.
*/
struct avs_enc_set_scrambler_param_t {
/*
* Supported values:
* 1 : enable scrambler
* 0 : disable scrambler
*/
uint32_t enable_scrambler;
};
/*
* Payload of the AVS_ENCODER_PARAM_ID_BIT_RATE_LEVEL_MAP parameter.
*/
struct afe_enc_level_to_bitrate_map_param_t {
/*
* Parameter for mapping link quality level to bitrate.
*/
struct afe_quality_level_to_bitrate_info mapping_table;
};
/*
* Payload of the AVS_ENCDEC_PARAM_ID_DEC_TO_ENC_COMMUNICATION parameter.
*/
struct afe_enc_dec_imc_info_param_t {
/*
* Parameter to set up Inter Module Communication (IMC) between
* AFE Decoder and Encoder.
*/
struct afe_imc_dec_enc_info imc_info;
};
/*
* Payload of the AVS_DECODER_PARAM_ID_DEPACKETIZER_ID parameter.
*/
struct avs_dec_depacketizer_id_param_t {
/*
* Supported values:
* #AFE_MODULE_ID_DEPACKETIZER_COP
* #AFE_MODULE_ID_DEPACKETIZER_COP_V1
* Any OpenDSP supported values
*/
uint32_t dec_depacketizer_id;
};
struct avs_dec_congestion_buffer_param_t {
uint32_t version;
uint16_t max_nr_buffers;
/*
* Maximum number of 1ms buffers:
* 0 - 256
*/
uint16_t pre_buffer_size;
/*
* Pre-buffering size in 1ms:
* 1 - 128
*/
};
/*
* ID of the parameter used by #AVS_MODULE_ID_DECODER to configure
* the decoder mode for the AFE module.
* This parameter cannot be set at runtime.
*/
#define AVS_DECODER_PARAM_ID_DEC_MEDIA_FMT 0x00013232
/*
* ID of the parameter used by #AVS_MODULE_ID_DECODER to configure
* the decoder mode of adaptive speech and byte swap mode
*/
#define AVS_DECODER_PARAM_ID_APTX_AD_SPEECH_DEC_INIT 0x0001334D
/*
* ID of the parameter used by #AVS_MODULE_ID_ENCODER to configure
* the encoder mode of adaptive speech and byte swap mode
*/
#define AVS_DECODER_PARAM_ID_APTX_AD_SPEECH_ENC_INIT 0x0001332B
/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to configure
* the island mode for a given AFE port.
*/
#define AFE_PARAM_ID_ISLAND_CONFIG 0x000102B4
/* Version information used to handle future additions to codec DMA
* configuration (for backward compatibility).
*/
#define AFE_API_VERSION_ISLAND_CONFIG 0x1
/* Payload of the AFE_PARAM_ID_ISLAND_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
#define AFE_PARAM_ID_POWER_MODE_CONFIG 0x0002002c
#define AFE_API_VERSION_POWER_MODE_CONFIG 0x1
struct afe_param_id_island_cfg_t {
uint32_t island_cfg_minor_version;
/* Tracks the configuration of this parameter.
* Supported values: #AFE_API_VERSION_ISLAND_CONFIG
*/
uint32_t island_enable;
/* Specifies whether island mode should be enabled or disabled for the
* use-case being setup.
* Supported values: 0 - Disable, 1 - Enable
*/
} __packed;
struct afe_param_id_power_mode_cfg_t {
uint32_t power_mode_cfg_minor_version;
/* Tracks the configuration of this parameter
* Supported values: #AFE_API_VERSION_POWER_MODE_CONFIG
*/
uint32_t power_mode_enable;
/* Specifies whether island mode should be enabled or disabled for the
* use-case being setup.
* Supported values: 0 - Disable, 1 - Enable
*/
} __packed;
/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to configure
* the Codec DMA interface.
*/
#define AFE_PARAM_ID_CODEC_DMA_CONFIG 0x000102B8
/* Version information used to handle future additions to codec DMA
* configuration (for backward compatibility).
*/
#define AFE_API_VERSION_CODEC_DMA_CONFIG 0x1
/* Payload of the AFE_PARAM_ID_CODEC_DMA_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_cdc_dma_cfg_t {
uint32_t cdc_dma_cfg_minor_version;
/* Tracks the configuration of this parameter.
* Supported values: #AFE_API_VERSION_CODEC_DMA_CONFIG
*/
uint32_t sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_11_025K
* - #AFE_PORT_SAMPLE_RATE_12K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_22_05K
* - #AFE_PORT_SAMPLE_RATE_24K
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_44_1K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_88_2K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_176_4K
* - #AFE_PORT_SAMPLE_RATE_192K
* - #AFE_PORT_SAMPLE_RATE_352_8K
* - #AFE_PORT_SAMPLE_RATE_384K
*/
uint16_t bit_width;
/* Bit width of the sample.
* Supported values: 16, 24, 32
*/
uint16_t data_format;
/* Data format supported by the codec DMA interface.
* Supported values:
* - #AFE_LINEAR_PCM_DATA
* - #AFE_LINEAR_PCM_DATA_PACKED_16BIT
*/
uint16_t num_channels;
/* Number of channels.
* Supported values: 1 to Maximum number of channels supported
* for each interface
*/
uint16_t active_channels_mask;
/* Active channels mask to denote the bit mask for active channels.
* Bits 0 to 7 denote channels 0 to 7. A 1 denotes the channel is active
* while a 0 denotes a channel is inactive.
* Supported values:
* Any mask with number of active bits equal to num_channels
*/
} __packed;
union afe_port_config {
struct afe_param_id_pcm_cfg pcm;
struct afe_param_id_i2s_cfg i2s;
struct afe_param_id_meta_i2s_cfg meta_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_v2 spdif;
struct afe_param_id_set_topology_cfg topology;
struct afe_param_id_tdm_cfg tdm;
struct afe_param_id_usb_audio_cfg usb_audio;
struct afe_param_id_aptx_sync_mode sync_mode_param;
struct asm_aac_frame_size_control_t frame_ctl_param;
struct afe_enc_fmt_id_param_t enc_fmt;
struct afe_port_media_type_t media_type;
struct afe_enc_cfg_blk_param_t enc_blk_param;
struct avs_enc_packetizer_id_param_t enc_pkt_id_param;
struct avs_enc_set_scrambler_param_t enc_set_scrambler_param;
struct avs_dec_depacketizer_id_param_t dec_depkt_id_param;
struct afe_dec_media_fmt_t dec_media_fmt;
struct afe_enc_level_to_bitrate_map_param_t map_param;
struct afe_enc_dec_imc_info_param_t imc_info_param;
struct afe_param_id_cdc_dma_cfg_t cdc_dma;
} __packed;
/*
* AFE event registration related APIs and corresponding payloads
*/
#define AFE_SVC_CMD_EVENT_CFG 0x000100FE
#define AFE_CMD_APPS_WAKEUP_IRQ_REGISTER_MINOR_VERSION 0x1
/* Flag to indicate AFE to register APPS wakeup Interrupt */
#define AFE_APPS_WAKEUP_IRQ_REGISTER_FLAG 1
/* Flag to indicate AFE to de-register APPS wakeup Interrupt */
#define AFE_APPS_WAKEUP_IRQ_DEREGISTER_FLAG 0
/* Default interrupt trigger value. */
#define DEFAULT_SETTINGS 0x00000001
/* Interrupt is triggered only if the input signal at the source is high. */
#define LEVEL_HIGH_TRIGGER 0x00000002
/* Interrupt is triggered only if the input signal at the source is low. */
#define LEVEL_LOW_TRIGGER 0x00000003
/* Interrupt is triggered only if the input signal at the source transitions
*from low to high.
*/
#define RISING_EDGE_TRIGGER 0x00000004
/* Interrupt is triggered only if the input signal at the source transitions
*from high to low.
*/
#define FALLING_EDGE_TRIGGER 0x00000005
/* Macro for invalid trigger type. This should not be used. */
#define INVALID_TRIGGER 0x00000006
#define AFE_EVENT_ID_MBHC_DETECTION_SW_WA 0x1
/* @weakgroup weak_afe_svc_cmd_evt_cfg_payload
*
* This is payload of each event that is to be
* registered with AFE service.
*/
struct afe_svc_cmd_evt_cfg_payload {
struct apr_hdr hdr;
uint32_t event_id;
/* Unique ID of the event.
*
* @values
* -# AFE_EVENT_ID_MBHC_DETECTION_SW_WA
*/
uint32_t reg_flag;
/* Flag for registering or de-registering an event.
* @values
* - #AFE_SVC_REGISTER_EVENT_FLAG
* - #AFE_SVC_DEREGISTER_EVENT_FLAG
*/
} __packed;
#define AFE_EVENT_MBHC_DETECTION_SW_WA 0x0001010F
#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;
/* Used by RTAC */
struct afe_rtac_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_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 {
struct apr_hdr apr_hdr;
/* Port interface and direction (Rx or Tx) to start. */
u16 port_id;
/* Maximum data size of the parameter ID/module ID combination.
* This is a multiple of four bytes
* Supported values: > 0
*/
u16 payload_size;
/* The memory mapping header to be used when requesting outband */
struct mem_mapping_hdr mem_hdr;
/* The module ID of the parameter data requested */
u32 module_id;
/* The parameter ID of the parameter data requested */
u32 param_id;
/* The header information for the parameter data */
struct param_hdr_v1 param_hdr;
} __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;
struct param_hdr_v1 param_hdr;
u8 param_data[0];
} __packed;
#define AFE_PORT_CMD_GET_PARAM_V3 0x000100FB
struct afe_port_cmd_get_param_v3 {
/* APR Header */
struct apr_hdr apr_hdr;
/* Port ID of the AFE port to configure. Port interface and direction
* (Rx or Tx) to configure. An even number represents the Rx direction,
* and an odd number represents the Tx direction.
*/
u16 port_id;
/* Reserved. This field must be set to zero. */
u16 reserved;
/* The memory mapping header to be used when requesting outband */
struct mem_mapping_hdr mem_hdr;
/* The header information for the parameter data */
struct param_hdr_v3 param_hdr;
} __packed;
#define AFE_PORT_CMDRSP_GET_PARAM_V3 0x00010108
struct afe_port_cmdrsp_get_param_v3 {
/* The status of the command */
uint32_t status;
/* The header information for the parameter data */
struct param_hdr_v3 param_hdr;
/* The parameter data to be filled when sent inband */
u8 param_data[0];
} __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;
/* 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
#define ADSP_MEMORY_MAP_MDF_SHMEM_4K_POOL 4
/* 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 COMPRESSED_PASSTHROUGH_NONE_TOPOLOGY 0x00010774
#define VPM_TX_SM_ECNS_V2_COPP_TOPOLOGY 0x00010F89
#define VPM_TX_VOICE_SMECNS_V2_COPP_TOPOLOGY 0x10000003
#define VPM_TX_VOICE_FLUENCE_SM_COPP_TOPOLOGY 0x10000004
#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
#define VPM_TX_DM_FLUENCE_EF_COPP_TOPOLOGY 0x10000005
#define AUDIO_RX_MONO_VOIP_COPP_TOPOLOGY 0x11000101
#define VPM_TX_VOICE_FLUENCE_NN_COPP_TOPOLOGY 0x10028008
/* 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;
#define AVS_MDF_MDSP_PROC_ID 0x2
#define AVS_MDF_SSC_PROC_ID 0x3
#define AVS_MDF_CDSP_PROC_ID 0x4
/* Shared memory map command payload used by the
* #AVCS_CMD_MAP_MDF_SHARED_MEMORY.
*
* This structure allows clients to map multiple shared memory
* regions with remote processor ID. All mapped regions must be
* from the same memory pool. Following this structure are
* num_regions of avs_shared_map_region_payload.
*/
struct avs_cmd_map_mdf_shared_memory {
struct apr_hdr hdr;
uint32_t mem_map_handle;
/* Unique identifier for the shared memory address.
*
* The aDSP returns this handle for
* #AVCS_CMD_SHARED_MEM_MAP_REGIONS
*
* Supported values:
* Any 32-bit value
*
* The aDSP uses this handle to retrieve the shared memory
* attributes. This handle can be an abstract representation
* of the shared memory regions that are being mapped.
*/
uint32_t proc_id;
/* Supported values:
* #AVS_MDF_MDSP_PROC_ID
* #AVS_MDF_SSC_PROC_ID
* #AVS_MDF_CDSP_PROC_ID
*/
uint32_t num_regions;
/* Number of regions to be mapped with the remote DSP processor
* mentioned by proc_id field.
*
* Array of structures of avs_shared_map_region_payload will follow.
* The address fields in those arrays should correspond to the remote
* processor mentioned by proc_id.
* In case of DSPs with SMMU enabled, the address should be IOVA.
* And for DSPs without SMMU, the address should be physical address.
*/
} __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_g711_dec_cfg {
u32 sample_rate;
};
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_dsd_cfg {
u16 num_version;
u16 is_bitwise_big_endian;
u16 dsd_channel_block_size;
u16 num_channels;
u8 channel_mapping[8];
u32 dsd_data_rate;
};
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_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
/* Second low frequency channel. */
#define PCM_CHANNEL_LFE2 17
/* Side left channel. */
#define PCM_CHANNEL_SL 18
/* Side right channel. */
#define PCM_CHANNEL_SR 19
/* Top front left channel. */
#define PCM_CHANNEL_TFL 20
/* Left vertical height channel. */
#define PCM_CHANNEL_LVH 20
/* Top front right channel. */
#define PCM_CHANNEL_TFR 21
/* Right vertical height channel. */
#define PCM_CHANNEL_RVH 21
/* Top center channel. */
#define PCM_CHANNEL_TC 22
/* Top back left channel. */
#define PCM_CHANNEL_TBL 23
/* Top back right channel. */
#define PCM_CHANNEL_TBR 24
/* Top side left channel. */
#define PCM_CHANNEL_TSL 25
/* Top side right channel. */
#define PCM_CHANNEL_TSR 26
/* Top back center channel. */
#define PCM_CHANNEL_TBC 27
/* Bottom front center channel. */
#define PCM_CHANNEL_BFC 28
/* Bottom front left channel. */
#define PCM_CHANNEL_BFL 29
/* Bottom front right channel. */
#define PCM_CHANNEL_BFR 30
/* Left wide channel. */
#define PCM_CHANNEL_LW 31
/* Right wide channel. */
#define PCM_CHANNEL_RW 32
/* Left side direct channel. */
#define PCM_CHANNEL_LSD 33
/* Right side direct channel. */
#define PCM_CHANNEL_RSD 34
/* Mark unused channel. */
#define PCM_CHANNEL_UNUSED 47
#define PCM_CUSTOM_CHANNEL_MAP_1 48
#define PCM_CUSTOM_CHANNEL_MAP_2 49
#define PCM_CUSTOM_CHANNEL_MAP_3 50
#define PCM_CUSTOM_CHANNEL_MAP_4 51
#define PCM_CUSTOM_CHANNEL_MAP_5 52
#define PCM_CUSTOM_CHANNEL_MAP_6 53
#define PCM_CUSTOM_CHANNEL_MAP_7 54
#define PCM_CUSTOM_CHANNEL_MAP_8 55
#define PCM_CUSTOM_CHANNEL_MAP_9 56
#define PCM_CUSTOM_CHANNEL_MAP_10 57
#define PCM_CUSTOM_CHANNEL_MAP_11 58
#define PCM_CUSTOM_CHANNEL_MAP_12 59
#define PCM_CUSTOM_CHANNEL_MAP_13 60
#define PCM_CUSTOM_CHANNEL_MAP_14 61
#define PCM_CUSTOM_CHANNEL_MAP_15 62
#define PCM_CUSTOM_CHANNEL_MAP_16 63
#define PCM_MAX_CHANNEL_MAP 63
/* Max valid channel map index */
#define PCM_MAX_CHMAP_ID PCM_MAX_CHANNEL_MAP
#define PCM_FORMAT_MAX_NUM_CHANNEL 8
#define PCM_FORMAT_MAX_CHANNELS_9 9
/* Used for ADM_CMD_DEVICE_OPEN_V8 */
#define PCM_FORMAT_MAX_NUM_CHANNEL_V8 32
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 0x00013222
#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
#define ASM_MEDIA_FMT_GENERIC_COMPRESSED 0x00013212
#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_generic_compressed_fmt_blk_t {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
/*
* Channel mapping array of bitstream output.
* Channel[i] mapping describes channel i inside the buffer, where
* i < num_channels. All valid used channels must be
* present at the beginning of the array.
*/
uint8_t channel_mapping[8];
/*
* Number of channels of the incoming bitstream.
* Supported values: 1,2,3,4,5,6,7,8
*/
uint16_t num_channels;
/*
* Nominal bits per sample value of the incoming bitstream.
* Supported values: 16, 32
*/
uint16_t bits_per_sample;
/*
* Nominal sampling rate of the incoming bitstream.
* Supported values: 8000, 11025, 16000, 22050, 24000, 32000,
* 44100, 48000, 88200, 96000, 176400, 192000,
* 352800, 384000
*/
uint32_t sampling_rate;
} __packed;
/* Command to send sample rate & channels for IEC61937 (compressed) or IEC60958
* (pcm) streams. Both audio standards use the same format and are used for
* HDMI or SPDIF output.
*/
#define ASM_DATA_CMD_IEC_60958_MEDIA_FMT 0x0001321E
struct asm_iec_compressed_fmt_blk_t {
struct apr_hdr hdr;
/*
* Nominal sampling rate of the incoming bitstream.
* Supported values: 8000, 11025, 16000, 22050, 24000, 32000,
* 44100, 48000, 88200, 96000, 176400, 192000,
* 352800, 384000
*/
uint32_t sampling_rate;
/*
* Number of channels of the incoming bitstream.
* Supported values: 1,2,3,4,5,6,7,8
*/
uint32_t num_channels;
} __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_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 0 @le I <
* num_channels. An unused channel is set to zero.
*/
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v3 {
uint16_t num_channels;
/*
* Number of channels
* Supported values: 1 to 8
*/
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
* Supported values: 2000 to 48000, 96000,192000 Hz
*/
uint16_t is_signed;
/* Flag that indicates that PCM samples are signed (1) */
uint16_t sample_word_size;
/*
* Size in bits of the word that holds a sample of a channel.
* Supported values: 12,24,32
*/
uint8_t channel_mapping[8];
/*
* Each element, i, in the array describes channel i inside the buffer where
* 0 <= i < num_channels. Unused channels are set to 0.
*/
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v4 {
uint16_t num_channels;
/*
* Number of channels
* Supported values: 1 to 8
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel
* Supported values: 16, 24, 32
*/
uint32_t sample_rate;
/*
* Number of samples per second
* Supported values: 2000 to 48000, 96000,192000 Hz
*/
uint16_t is_signed;
/* Flag that indicates that PCM samples are signed (1) */
uint16_t sample_word_size;
/*
* Size in bits of the word that holds a sample of a channel.
* Supported values: 12,24,32
*/
uint8_t channel_mapping[8];
/*
* Each element, i, in the array describes channel i inside the buffer where
* 0 <= i < num_channels. Unused channels are set to 0.
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
* 31 - for 24b unpacked or 32bit
*/
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v5 {
uint16_t num_channels;
/*
* Number of channels
* Supported values: 1 to 32
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel
* Supported values: 16, 24, 32
*/
uint32_t sample_rate;
/*
* Number of samples per second
* Supported values: 2000 to 48000, 96000,192000 Hz
*/
uint16_t is_signed;
/* Flag that indicates that PCM samples are signed (1) */
uint16_t sample_word_size;
/*
* Size in bits of the word that holds a sample of a channel.
* Supported values: 12,24,32
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
* 31 - for 24b unpacked or 32bit
*/
uint8_t channel_mapping[32];
/*
* Each element, i, in the array describes channel i inside the buffer where
* 0 <= i < num_channels. Unused channels are set to 0.
*/
} __packed;
/*
* Payload of the multichannel PCM configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
*/
struct asm_multi_channel_pcm_fmt_blk_param_v3 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
struct asm_multi_channel_pcm_fmt_blk_v3 param;
} __packed;
/*
* Payload of the multichannel PCM configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
*/
struct asm_multi_channel_pcm_fmt_blk_param_v4 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
struct asm_multi_channel_pcm_fmt_blk_v4 param;
} __packed;
/*
* Payload of the multichannel PCM configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format.
*/
struct asm_multi_channel_pcm_fmt_blk_param_v5 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
struct asm_multi_channel_pcm_fmt_blk_v5 param;
} __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;
struct asm_custom_enc_cfg_t_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint32_t sample_rate;
uint16_t num_channels;
uint16_t reserved;
/* num_ch == 1, then PCM_CHANNEL_C,
* num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
*/
uint8_t channel_mapping[8];
uint32_t custom_size;
uint8_t custom_data[15];
} __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;
/*
* Payload of the multichannel PCM encoder configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format.
*/
struct asm_multi_channel_pcm_enc_cfg_v5 {
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.
* @values
* - 0 -- Native mode
* - 1 -- 8 channels
* 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.
* @values 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second.
* @values 0, 8000 to 48000 Hz
* A value of 0 indicates the native sampling rate. Encoding is
* performed at the input sampling rate.
*/
uint16_t is_signed;
/*
* Flag that indicates the PCM samples are signed (1). Currently, only
* signed PCM samples are supported.
*/
uint16_t sample_word_size;
/*
* The size in bits of the word that holds a sample of a channel.
* @values 16, 24, 32
* 16-bit samples are always placed in 16-bit words:
* sample_word_size = 1.
* 24-bit samples can be placed in 32-bit words or in consecutive
* 24-bit words.
* - If sample_word_size = 32, 24-bit samples are placed in the
* most significant 24 bits of a 32-bit word.
* - If sample_word_size = 24, 24-bit samples are placed in
* 24-bit words. @tablebulletend
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
*/
uint8_t channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL_V8];
/*
* Channel mapping array expected at the encoder output.
* Channel[i] mapping describes channel i inside the buffer, where
* 0 @le i < num_channels. All valid used channels must be present at
* the beginning of the array.
* If Native mode is set for the channels, this field is ignored.
* @values See Section @xref{dox:PcmChannelDefs}
*/
} __packed;
/*
* Payload of the multichannel PCM encoder configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
*/
struct asm_multi_channel_pcm_enc_cfg_v4 {
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.
* @values
* - 0 -- Native mode
* - 1 -- 8 channels
* 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.
* @values 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second.
* @values 0, 8000 to 48000 Hz
* A value of 0 indicates the native sampling rate. Encoding is
* performed at the input sampling rate.
*/
uint16_t is_signed;
/*
* Flag that indicates the PCM samples are signed (1). Currently, only
* signed PCM samples are supported.
*/
uint16_t sample_word_size;
/*
* The size in bits of the word that holds a sample of a channel.
* @values 16, 24, 32
* 16-bit samples are always placed in 16-bit words:
* sample_word_size = 1.
* 24-bit samples can be placed in 32-bit words or in consecutive
* 24-bit words.
* - If sample_word_size = 32, 24-bit samples are placed in the
* most significant 24 bits of a 32-bit word.
* - If sample_word_size = 24, 24-bit samples are placed in
* 24-bit words. @tablebulletend
*/
uint8_t channel_mapping[8];
/*
* Channel mapping array expected at the encoder output.
* Channel[i] mapping describes channel i inside the buffer, where
* 0 @le i < num_channels. All valid used channels must be present at
* the beginning of the array.
* If Native mode is set for the channels, this field is ignored.
* @values See Section @xref{dox:PcmChannelDefs}
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
* 31 - for 24b unpacked or 32bit
*/
} __packed;
/*
* Payload of the multichannel PCM encoder configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
*/
struct asm_multi_channel_pcm_enc_cfg_v3 {
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.
* @values
* - 0 -- Native mode
* - 1 -- 8 channels
* 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.
* @values 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second.
* @values 0, 8000 to 48000 Hz
* A value of 0 indicates the native sampling rate. Encoding is
* performed at the input sampling rate.
*/
uint16_t is_signed;
/*
* Flag that indicates the PCM samples are signed (1). Currently, only
* signed PCM samples are supported.
*/
uint16_t sample_word_size;
/*
* The size in bits of the word that holds a sample of a channel.
* @values 16, 24, 32
* 16-bit samples are always placed in 16-bit words:
* sample_word_size = 1.
* 24-bit samples can be placed in 32-bit words or in consecutive
* 24-bit words.
* - If sample_word_size = 32, 24-bit samples are placed in the
* most significant 24 bits of a 32-bit word.
* - If sample_word_size = 24, 24-bit samples are placed in
* 24-bit words. @tablebulletend
*/
uint8_t channel_mapping[8];
/*
* Channel mapping array expected at the encoder output.
* Channel[i] mapping describes channel i inside the buffer, where
* 0 @le i < num_channels. All valid used channels must be present at
* the beginning of the array.
* If Native mode is set for the channels, this field is ignored.
* @values See Section @xref{dox:PcmChannelDefs}
*/
};
/* @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
/* Enumeration for the AAC LATM format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM 4
#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;
#define ASM_MEDIA_FMT_G711_ALAW_FS 0x00010BF7
#define ASM_MEDIA_FMT_G711_MLAW_FS 0x00010C2E
struct asm_g711_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 sample_rate;
/*
* Number of samples per second.
* Supported values: 8000, 16000 Hz
*/
} __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_g711_dec_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 sample_rate;
} __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;
struct asm_dsd_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 num_version;
u16 is_bitwise_big_endian;
u16 dsd_channel_block_size;
u16 num_channels;
u8 channel_mapping[8];
u32 dsd_data_rate;
} __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
#define ASM_MEDIA_FMT_DSD 0x00012F3E
#define ASM_MEDIA_FMT_TRUEHD 0x00013215
/* 0x0 is used for fomat ID since ADSP dynamically determines the
* format encapsulated in the IEC61937 (compressed) or IEC60958
* (pcm) packets.
*/
#define ASM_MEDIA_FMT_IEC 0x00000000
/* 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_CMD_EOS_V2 0x00012F3C
#define ASM_DATA_EVENT_RENDERED_EOS_V2 0x00012F3D
#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 doesn't 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 doesn't 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_PULL_MODE_WRITE 0x00010DD9
/* Bitmask for the stream_perf_mode subfield. */
#define ASM_BIT_MASK_STREAM_PERF_FLAG_PULL_MODE_WRITE 0xE0000000UL
/* Bitmask for the stream_perf_mode subfield. */
#define ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE 29
#define ASM_STREAM_CMD_OPEN_PUSH_MODE_READ 0x00010DDA
#define ASM_BIT_MASK_STREAM_PERF_FLAG_PUSH_MODE_READ 0xE0000000UL
#define ASM_SHIFT_STREAM_PERF_FLAG_PUSH_MODE_READ 29
#define ASM_DATA_EVENT_WATERMARK 0x00010DDB
struct asm_shared_position_buffer {
volatile uint32_t frame_counter;
/* Counter used to handle interprocessor synchronization issues.
* When frame_counter is 0: read_index, wall_clock_us_lsw, and
* wall_clock_us_msw are invalid.
* Supported values: >= 0.
*/
volatile uint32_t index;
/* Index in bytes from where the aDSP is reading/writing.
* Supported values: 0 to circular buffer size - 1
*/
volatile uint32_t wall_clock_us_lsw;
/* Lower 32 bits of the 64-bit wall clock time in microseconds when the
* read index was updated.
* Supported values: >= 0
*/
volatile uint32_t wall_clock_us_msw;
/* Upper 32 bits of the 64 bit wall clock time in microseconds when the
* read index was updated
* Supported values: >= 0
*/
} __packed;
struct asm_shared_watermark_level {
uint32_t watermark_level_bytes;
} __packed;
struct asm_stream_cmd_open_shared_io {
struct apr_hdr hdr;
uint32_t mode_flags;
uint16_t endpoint_type;
uint16_t topo_bits_per_sample;
uint32_t topo_id;
uint32_t fmt_id;
uint32_t shared_pos_buf_phy_addr_lsw;
uint32_t shared_pos_buf_phy_addr_msw;
uint16_t shared_pos_buf_mem_pool_id;
uint16_t shared_pos_buf_num_regions;
uint32_t shared_pos_buf_property_flag;
uint32_t shared_circ_buf_start_phy_addr_lsw;
uint32_t shared_circ_buf_start_phy_addr_msw;
uint32_t shared_circ_buf_size;
uint16_t shared_circ_buf_mem_pool_id;
uint16_t shared_circ_buf_num_regions;
uint32_t shared_circ_buf_property_flag;
uint32_t num_watermark_levels;
struct asm_multi_channel_pcm_fmt_blk_v3 fmt;
struct avs_shared_map_region_payload map_region_pos_buf;
struct avs_shared_map_region_payload map_region_circ_buf;
struct asm_shared_watermark_level watermark[0];
} __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_OPEN_TRANSCODE_LOOPBACK 0x00010DBA
/* Bitmask for the stream's Performance mode. */
#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \
(0x70000000UL)
/* Bit shift for the stream's Performance mode. */
#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28
/* Bitmask for the decoder converter enable flag. */
#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL)
/* Shift value for the decoder converter enable flag. */
#define ASM_SHIFT_DECODER_CONVERTER_FLAG 3
/* Converter mode is None (Default). */
#define ASM_CONVERTER_MODE_NONE 0
/* Converter mode is DDP-to-DD. */
#define ASM_DDP_DD_CONVERTER_MODE 1
/* Identifies a special converter mode where source and sink formats
* are the same but postprocessing must applied. Therefore, Decode
* @rarrow Re-encode is necessary.
*/
#define ASM_POST_PROCESS_CONVERTER_MODE 2
struct asm_stream_cmd_open_transcode_loopback_t {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode Flags specifies the performance mode in which this stream
* is to be opened.
* Supported values{for bits 30 to 28}(stream_perf_mode flag)
*
* #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward
* compatibility to the original behavior
* of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK
*
* #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using
* shortened buffers in low latency POPP
* - Recommendation: Do not enable high latency algorithms. They might
* negate the benefits of opening a low latency stream, and they
* might also suffer quality degradation from unexpected jitter.
* - This Low Latency mode is supported only for PCM In and PCM Out
* loopbacks. An error is returned if Low Latency mode is opened for
* other transcode loopback modes.
* - To configure this subfield, use
* ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and
* ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK.
*
* Supported values{for bits 6 to 3} (decoder-converter compatibility)
* #ASM_CONVERTER_MODE_NONE (0x0) -- Default
* #ASM_DDP_DD_CONVERTER_MODE (0x1)
* #ASM_POST_PROCESS_CONVERTER_MODE (0x2)
* 0x3-0xF -- Reserved for future use
* - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and
* ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit
* All other bits are reserved; clients must set them to 0.
*/
u32 src_format_id;
/* Specifies the media format of the input audio stream.
*
* Supported values
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
* - #ASM_MEDIA_FMT_DTS
* - #ASM_MEDIA_FMT_EAC3_DEC
* - #ASM_MEDIA_FMT_EAC3
* - #ASM_MEDIA_FMT_AC3_DEC
* - #ASM_MEDIA_FMT_AC3
*/
u32 sink_format_id;
/* Specifies the media format of the output stream.
*
* Supported values
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
* - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode)
*/
u32 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_MCH_PEAK_VOL
* - #ASM_STREAM_POSTPROC_TOPO_ID_NONE
* Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES.
* This field is ignored for the Converter mode, in which no
* postprocessing is performed.
*/
u16 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 0. Otherwise, an error is returned.
*/
u16 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 0. Otherwise, an error is returned.
*/
u16 bits_per_sample;
/* Number of bits per sample processed by the ASM modules.
* Supported values 16, 24
*/
u16 reserved;
/* This field must be set to 0.
*/
} __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
#define ASM_STREAM_CMD_SET_PP_PARAMS_V3 0x0001320D
/*
* Structure for the ASM Stream Set PP Params command. Parameter data must be
* pre-packed with the correct header for either V2 or V3 when sent in-band.
* Use q6core_pack_pp_params to pack the header and data correctly depending on
* Instance ID support.
*/
struct asm_stream_cmd_set_pp_params {
/* APR Header */
struct apr_hdr apr_hdr;
/* The memory mapping header to be used when sending out of band */
struct mem_mapping_hdr mem_hdr;
/* The total size of the payload, including the parameter header */
u32 payload_size;
/* The parameter data to be filled when sent inband. Parameter data
* must be pre-packed with parameter header and then copied here. Use
* q6core_pack_pp_params to pack the header and param data correctly.
*/
u32 param_data[0];
} __packed;
#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2
#define ASM_STREAM_CMD_GET_PP_PARAMS_V3 0x0001320E
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_STREAM_CMD_SET_ENCDEC_PARAM_V2 0x00013218
struct asm_stream_cmd_set_encdec_param_v2 {
u16 service_id;
/* 0 - ASM_ENCODER_SVC; 1 - ASM_DECODER_SVC */
u16 reserved;
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;
#define ASM_STREAM_CMD_REGISTER_ENCDEC_EVENTS 0x00013219
#define ASM_STREAM_CMD_ENCDEC_EVENTS 0x0001321A
#define AVS_PARAM_ID_RTIC_SHARED_MEMORY_ADDR 0x00013237
struct avs_rtic_shared_mem_addr {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param_v2 encdec;
u32 shm_buf_addr_lsw;
/* Lower 32 bit of the RTIC shared memory */
u32 shm_buf_addr_msw;
/* Upper 32 bit of the RTIC shared memory */
u32 buf_size;
/* Size of buffer */
u16 shm_buf_mem_pool_id;
/* ADSP_MEMORY_MAP_SHMEM8_4K_POOL */
u16 shm_buf_num_regions;
/* number of regions to map */
u32 shm_buf_flag;
/* buffer property flag */
struct avs_shared_map_region_payload map_region;
/* memory map region*/
} __packed;
#define AVS_PARAM_ID_RTIC_EVENT_ACK 0x00013238
struct avs_param_rtic_event_ack {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param_v2 encdec;
} __packed;
#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
#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V3 0x0001320F
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 command */
/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
* command, 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
/* Bitmask for the IEC 61937 to 61937 pass-through capture. */
#define ASM_BIT_MASK_IEC_61937_PASS_THROUGH_FLAG (0x00000001UL)
/* Shift value for the IEC 61937 to 61937 pass-through capture. */
#define ASM_SHIFT_IEC_61937_PASS_THROUGH_FLAG 0
/* Bitmask for the DSD pass-through capture. */
#define ASM_BIT_MASK_COMPRESSED_FORMAT_FLAG (0x00000003UL)
/* Shift value for the DSD pass-through capture. */
#define ASM_SHIFT_DSD_COMPRESSED_FORMAT_FLAG 0
#define ASM_DSD_FORMAT_FLAG 2
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 and packaging.
* Supported values for bit 0: (IEC 61937 pass-through mode)
* - 0 -- Unpack the IEC 61937 format stream to RAW compressed format
* - 1 -- Pass-through transfer of the IEC 61937 format stream
* - Use #ASM_BIT_MASK_IEC_61937_PASS_THROUGH_FLAG to set the bitmask
* and #ASM_SHIFT_IEC_61937_PASS_THROUGH_FLAG to set the shift value
* for this bit.
* Supported values for bit 1: (DSD native pass-through mode)
* 0 -- non DSD operation
* 1 -- Pass-through transfer of the DSD format stream
* To set this bit, use #ASM_BIT_MASK_DSD_PASS_THROUGH_FLAG and
* use #ASM_SHIFT_DSD_PASS_THROUGH_FLAG to set the shift value for
* this bit
* 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 for IEC to RAW compressed
* unpack mode.
* Value is don't care for IEC 61937 pass-through mode.
* @values
* - >0 -- For IEC 61937-to-RAW Compressed Unpack mode
* - 1 -- For IEC 61937 or DSD Pass-through mode
*/
} __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 ADM_CMD_GET_PP_TOPO_MODULE_LIST_V2 0x00010360
#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST_V2 0x00010361
#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 {
struct apr_hdr apr_hdr;
/* The memory mapping header to be used when requesting out of band */
struct mem_mapping_hdr mem_hdr;
/*
* Maximum data size of the list of modules. This
* field is a multiple of 4 bytes.
*/
uint32_t param_max_size;
} __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 {
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 {
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 {
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 {
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 {
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_CHANNEL_TS
* - #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 {
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 {
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_CHANNEL_TS
* - #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 {
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 {
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
/* Get & set params */
#define VSS_ICOMMON_CMD_SET_PARAM_V2 0x0001133D
#define VSS_ICOMMON_CMD_GET_PARAM_V2 0x0001133E
#define VSS_ICOMMON_RSP_GET_PARAM 0x00011008
#define VSS_ICOMMON_CMD_SET_PARAM_V3 0x00013245
#define VSS_ICOMMON_CMD_GET_PARAM_V3 0x00013246
#define VSS_ICOMMON_RSP_GET_PARAM_V3 0x00013247
#define VSS_MAX_AVCS_NUM_SERVICES 25
/* 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_SP_RX_LIMITER_TH 0x000102B1
#define AFE_PARAM_ID_FBSP_MODE_RX_CFG 0x0001021D
#define AFE_PARAM_ID_FBSP_PTONE_RAMP_CFG 0x00010260
#define AFE_PARAM_ID_SP_RX_TMAX_XMAX_LOGGING 0x000102BC
#define AFE_PARAM_ID_CPS_LPASS_HW_INTF_CFG 0x000102EF
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;
#define AFE_MODULE_SPEAKER_PROTECTION_V2_TH_VI 0x0001026A
#define AFE_PARAM_ID_SP_V2_TH_VI_MODE_CFG 0x0001026B
#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_CFG 0x0001029F
#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_PARAMS 0x000102A0
#define AFE_PARAM_ID_SP_V2_TH_VI_V_VALI_CFG 0x000102BF
#define AFE_PARAM_ID_SP_V2_TH_VI_V_VALI_PARAMS 0x000102C0
struct afe_sp_th_vi_mode_cfg {
uint32_t minor_version;
uint32_t operation_mode;
/*
* Operation mode of thermal VI module.
* 0 -- Normal Running mode
* 1 -- Calibration mode
* 2 -- FTM mode
*/
uint32_t r0t0_selection_flag[SP_V2_NUM_MAX_SPKR];
/*
* Specifies which set of R0, T0 values the algorithm will use.
* This field is valid only in Normal mode (operation_mode = 0).
* 0 -- Use calibrated R0, T0 value
* 1 -- Use safe R0, T0 value
*/
int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
/*
* Calibration point resistance per device. This field is valid
* only in Normal mode (operation_mode = 0).
* values 33554432 to 1073741824 Ohms (in Q24 format)
*/
int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR];
/*
* Calibration point temperature per device. This field is valid
* in both Normal mode and Calibration mode.
* values -1920 to 5120 degrees C (in Q6 format)
*/
uint32_t quick_calib_flag;
/*
* Indicates whether calibration is to be done in quick mode or not.
* This field is valid only in Calibration mode (operation_mode = 1).
* 0 -- Disabled
* 1 -- Enabled
*/
} __packed;
struct afe_sp_th_vi_ftm_cfg {
uint32_t minor_version;
uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
struct afe_sp_th_vi_ftm_params {
uint32_t minor_version;
int32_t dc_res_q24[SP_V2_NUM_MAX_SPKR];
/*
* DC resistance value in q24 format
* values 0 to 2147483647 Ohms (in Q24 format)
*/
int32_t temp_q22[SP_V2_NUM_MAX_SPKR];
/*
* temperature value in q22 format
* values -125829120 to 2147483647 degC (in Q22 format)
*/
uint32_t status[SP_V2_NUM_MAX_SPKR];
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
struct afe_sp_th_vi_get_param {
struct param_hdr_v3 pdata;
struct afe_sp_th_vi_ftm_params param;
} __packed;
struct afe_sp_th_vi_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_sp_th_vi_ftm_params param;
} __packed;
struct afe_sp_th_vi_v_vali_cfg {
uint32_t minor_version;
uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* Wait time to heat up speaker before collecting statistics
* for V validation mode in ms.
* values 100 to 1000 ms
*/
uint32_t vali_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* duration for which V VALIDATION statistics are collected in ms.
* values 1000 to 3000 ms
*/
} __packed;
struct afe_sp_th_vi_v_vali_params {
uint32_t minor_version;
uint32_t vrms_q24[SP_V2_NUM_MAX_SPKR];
/*
* Vrms value in q24 format
* values [0 33554432] Q24 (0 - 2Vrms)
*/
uint32_t status[SP_V2_NUM_MAX_SPKR];
/*
* v-vali packet status
* 0 - Failed.
* 1 - Success.
* 2 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non v-vali Mode
* 3 - Inactive mode -- Port is not yet started.
* 4 - Wait state. wait_time_ms has not yet elapsed
* 5 - In progress state. ftm_time_ms has not yet elapsed.
*/
} __packed;
struct afe_sp_th_vi_v_vali_get_param {
struct param_hdr_v3 pdata;
struct afe_sp_th_vi_v_vali_params param;
} __packed;
struct afe_sp_th_vi_v_vali_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_sp_th_vi_v_vali_params param;
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION_V2_EX_VI 0x0001026F
#define AFE_PARAM_ID_SP_V2_EX_VI_MODE_CFG 0x000102A1
#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_CFG 0x000102A2
#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_PARAMS 0x000102A3
struct afe_sp_ex_vi_mode_cfg {
uint32_t minor_version;
uint32_t operation_mode;
/*
* Operation mode of Excursion VI module.
* 0 - Normal Running mode
* 2 - FTM mode
*/
} __packed;
struct afe_sp_ex_vi_ftm_cfg {
uint32_t minor_version;
uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
struct afe_sp_ex_vi_ftm_params {
uint32_t minor_version;
int32_t freq_q20[SP_V2_NUM_MAX_SPKR];
/*
* Resonance frequency in q20 format
* values 0 to 2147483647 Hz (in Q20 format)
*/
int32_t resis_q24[SP_V2_NUM_MAX_SPKR];
/*
* Mechanical resistance in q24 format
* values 0 to 2147483647 Ohms (in Q24 format)
*/
int32_t qmct_q24[SP_V2_NUM_MAX_SPKR];
/*
* Mechanical Qfactor in q24 format
* values 0 to 2147483647 (in Q24 format)
*/
uint32_t status[SP_V2_NUM_MAX_SPKR];
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode.
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
struct afe_sp_rx_tmax_xmax_logging_param {
/*
* Maximum excursion since the last grasp of xmax in mm.
*/
int32_t max_excursion[SP_V2_NUM_MAX_SPKR];
/*
* Number of periods when the monitored excursion exceeds to and
* stays at Xmax during logging_count_period.
*/
uint32_t count_exceeded_excursion[SP_V2_NUM_MAX_SPKR];
/*
* Maximum temperature since the last grasp of tmax in C.
*/
int32_t max_temperature[SP_V2_NUM_MAX_SPKR];
/*
* Number of periods when the monitored temperature exceeds to and
* stays at Tmax during logging_count_period
*/
uint32_t count_exceeded_temperature[SP_V2_NUM_MAX_SPKR];
} __packed;
struct afe_sp_rx_tmax_xmax_logging_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_sp_rx_tmax_xmax_logging_param param;
} __packed;
struct afe_sp_ex_vi_get_param {
struct param_hdr_v3 pdata;
struct afe_sp_ex_vi_ftm_params param;
} __packed;
struct afe_sp_ex_vi_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_sp_ex_vi_ftm_params param;
} __packed;
struct afe_sp_rx_limiter_th_param {
uint32_t minor_version;
uint32_t lim_thr_per_calib_q27[SP_V2_NUM_MAX_SPKR];
} __packed;
struct afe_spkr_prot_get_vi_calib {
struct apr_hdr hdr;
struct mem_mapping_hdr mem_hdr;
struct param_hdr_v3 pdata;
struct asm_calib_res_cfg res_cfg;
} __packed;
struct afe_spkr_prot_calib_get_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct asm_calib_res_cfg res_cfg;
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION_V4_RX 0x000102C7
#define AFE_PARAM_ID_SP_V4_OP_MODE 0x000102C9
#define AFE_PARAM_ID_SP_V4_RX_TMAX_XMAX_LOGGING 0x000102D2
struct afe_sp_v4_param_op_mode {
uint32_t mode;
} __packed;
struct afe_sp_v4_channel_tmax_xmax_params {
/*
* Maximum excursion since the last grasp of xmax in mm.
*/
int32_t max_excursion;
/*
* Number of periods when the monitored excursion exceeds to and
* stays at Xmax during logging_count_period.
*/
uint32_t count_exceeded_excursion;
/*
* Maximum temperature since the last grasp of tmax in C.
*/
int32_t max_temperature;
/*
* Number of periods when the monitored temperature exceeds to and
* stays at Tmax during logging_count_period
*/
uint32_t count_exceeded_temperature;
} __packed;
/* This structure is followed by 'num_ch' number of structures of
* type afe_sp_v4_channel_tmax_xmax_params.
*/
struct afe_sp_v4_param_tmax_xmax_logging {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_tmax_xmax_params ch_tmax_xmax[0];
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION_V4_VI 0x000102D3
#define AFE_PARAM_ID_SP_V4_VI_OP_MODE_CFG 0x000102D4
#define AFE_PARAM_ID_SP_V4_VI_R0T0_CFG 0x000102D5
#define AFE_PARAM_ID_SP_V4_CALIB_RES_CFG 0x000102D8
#define AFE_PARAM_ID_SP_V4_TH_VI_FTM_CFG 0x000102D9
#define AFE_PARAM_ID_SP_V4_TH_VI_FTM_PARAMS 0x000102DA
#define AFE_PARAM_ID_SP_V4_TH_VI_V_VALI_CFG 0x000102DB
#define AFE_PARAM_ID_SP_V4_TH_VI_V_VALI_PARAMS 0x000102DC
#define AFE_PARAM_ID_SP_V4_EX_VI_MODE_CFG 0x000102DF
#define AFE_PARAM_ID_SP_V4_EX_VI_FTM_CFG 0x000102E0
#define AFE_PARAM_ID_SP_V4_EX_VI_FTM_PARAMS 0x000102E1
#define AFE_PARAM_ID_SP_V4_VI_CHANNEL_MAP_CFG 0x000102E5
struct afe_sp_v4_param_vi_op_mode_cfg {
uint32_t num_speakers;
/* Number of channels for Rx signal.
*/
uint32_t th_operation_mode;
/*
* Operation mode of thermal VI module.
* 0 -- Normal Running mode
* 1 -- Calibration mode
* 2 -- FTM mode
* 3 -- V-Validation mode
*/
uint32_t th_quick_calib_flag;
/*
* Indicates whether calibration is to be done in quick mode or not.
* This field is valid only in Calibration mode (operation_mode = 1).
* 0 -- Disabled
* 1 -- Enabled
*/
uint32_t th_r0t0_selection_flag[SP_V2_NUM_MAX_SPKR];
/*
* Specifies which set of R0, T0 values the algorithm will use.
* This field is valid only in Normal mode (operation_mode = 0).
* 0 -- Use calibrated R0, T0 value
* 1 -- Use safe R0, T0 value
*/
} __packed;
struct afe_sp_v4_channel_r0t0 {
int32_t r0_cali_q24;
/*
* Calibration point resistance per device. This field is valid
* only in Normal mode (operation_mode = 0).
* values 33554432 to 1073741824 Ohms (in Q24 format)
*/
int16_t t0_cali_q6;
/*
* Calibration point temperature per device. This field is valid
* in both Normal mode and Calibration mode.
* values -1920 to 5120 degrees C (in Q6 format)
*/
uint16_t reserved;
} __packed;
/* Followed by this structure are 'num_speaakers' number of structures
* of type afe_sp_v4_channel_r0t0.
*/
struct afe_sp_v4_param_th_vi_r0t0_cfg {
uint32_t num_speakers;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_r0t0 ch_r0t0[0];
} __packed;
struct afe_sp_v4_param_th_vi_calib_res_cfg {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
uint32_t th_vi_ca_state;
/*
* Represents the calibration state for both speakers.
* 0 -- Incorrect operation mode.
* 1 -- Inactive mode.
* 2 -- Wait state.
* 3 -- Calibration in progress state.
* 4 -- Calibration success.
* 5 -- Calibration failed.
*/
int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
/* Calibration resistance per device.
*/
} __packed;
struct afe_sp_v4_th_vi_calib_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_sp_v4_param_th_vi_calib_res_cfg res_cfg;
} __packed;
struct afe_sp_v4_channel_ftm_cfg {
uint32_t wait_time_ms;
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms;
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures
* of type afe_sp_v4_channel_ftm_cfg.
*/
struct afe_sp_v4_param_th_vi_ftm_cfg {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_ftm_cfg ch_ftm_cfg[0];
} __packed;
struct afe_sp_v4_channel_ftm_params {
int32_t dc_res_q24;
/*
* DC resistance value in q24 format
* values 0 to 2147483647 Ohms (in Q24 format)
*/
int32_t temp_q22;
/*
* temperature value in q22 format
* values -125829120 to 2147483647 degC (in Q22 format)
*/
uint32_t status;
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures
* of type afe_sp_v4_channel_ftm_params.
*/
struct afe_sp_v4_param_th_vi_ftm_params {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_ftm_params ch_ftm_params[0];
} __packed;
struct afe_sp_v4_gen_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
} __packed;
struct afe_sp_v4_channel_v_vali_cfg {
uint32_t wait_time_ms;
/*
* Wait time to heat up speaker before collecting statistics
* for V validation mode in ms.
* values 100 to 1000 ms
*/
uint32_t vali_time_ms;
/*
* duration for which V VALIDATION statistics are collected in ms.
* values 1000 to 3000 ms
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures
* of type afe_sp_v4_channel_v_vali_cfg.
*/
struct afe_sp_v4_param_th_vi_v_vali_cfg {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_v_vali_cfg ch_v_vali_cfg[0];
} __packed;
struct afe_sp_v4_channel_v_vali_params {
uint32_t vrms_q24;
/*
* Vrms value in q24 format
* values [0 33554432] Q24 (0 - 2Vrms)
*/
uint32_t status;
/*
* v-vali packet status
* 0 - Failed.
* 1 - Success.
* 2 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non v-vali Mode
* 3 - Inactive mode -- Port is not yet started.
* 4 - Wait state. wait_time_ms has not yet elapsed
* 5 - In progress state. ftm_time_ms has not yet elapsed.
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures
* of type afe_sp_v4_channel_v_vali_params.
*/
struct afe_sp_v4_param_th_vi_v_vali_params {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_v_vali_params ch_v_vali_params[0];
} __packed;
struct afe_sp_v4_param_ex_vi_mode_cfg {
uint32_t operation_mode;
/*
* Operation mode of Excursion VI module.
* 0 - Normal Running mode
* 2 - FTM mode
*/
} __packed;
struct afe_sp_v4_channel_ex_vi_ftm {
uint32_t wait_time_ms;
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms;
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures
* of type afe_sp_v4_channel_ex_vi_ftm.
*/
struct afe_sp_v4_param_ex_vi_ftm_cfg {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_ex_vi_ftm ch_ex_vi_ftm[0];
} __packed;
struct afe_sp_v4_channel_ex_vi_ftm_params {
int32_t ftm_re_q24;
/*
* DC resistance of voice coil at room temperature
* or small signal level in Ohm.
*/
int32_t ftm_Bl_q24;
/* Force factor.
*/
int32_t ftm_Rms_q24;
/* Mechanical damping or resistance of loudspeaker in Kg/sec.
*/
int32_t ftm_Kms_q24;
/* Mechanical stiffness of driver suspension in N/mm.
*/
int32_t ftm_Fres_q20;
/* Resonance frequency in Hz.
*/
int32_t ftm_Qms_q24;
/* Mechanical Q-factor.
*/
uint32_t status;
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode.
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
/* This structure is followed by 'num_ch' number of structures of
* type afe_sp_v4_channel_ex_vi_ftm_params.
*/
struct afe_sp_v4_param_ex_vi_ftm_params {
uint32_t num_ch;
/* Number of channels for Rx signal.
*/
struct afe_sp_v4_channel_ex_vi_ftm_params ch_ex_vi_ftm_params[0];
} __packed;
struct afe_sp_v4_param_vi_channel_map_cfg {
int32_t num_channels;
int32_t chan_info[4];
} __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;
struct afe_sp_th_vi_mode_cfg th_vi_mode_cfg;
struct afe_sp_th_vi_ftm_cfg th_vi_ftm_cfg;
struct afe_sp_th_vi_v_vali_cfg th_vi_v_vali_cfg;
struct afe_sp_ex_vi_mode_cfg ex_vi_mode_cfg;
struct afe_sp_ex_vi_ftm_cfg ex_vi_ftm_cfg;
struct afe_sp_rx_limiter_th_param limiter_th_cfg;
struct afe_sp_v4_param_op_mode v4_op_mode;
struct afe_sp_v4_param_vi_op_mode_cfg v4_vi_op_mode;
struct afe_sp_v4_param_th_vi_r0t0_cfg v4_r0t0_cfg;
struct afe_sp_v4_param_th_vi_ftm_cfg v4_th_vi_ftm_cfg;
struct afe_sp_v4_param_th_vi_v_vali_cfg v4_v_vali_cfg;
struct afe_sp_v4_param_ex_vi_mode_cfg v4_ex_vi_mode_cfg;
struct afe_sp_v4_param_ex_vi_ftm_cfg v4_ex_vi_ftm_cfg;
struct afe_sp_v4_param_vi_channel_map_cfg v4_ch_map_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
/* Opcode to set BT address and license for aptx decoder */
#define APTX_DECODER_BT_ADDRESS 0x00013201
#define APTX_CLASSIC_DEC_LICENSE_ID 0x00013202
struct aptx_dec_bt_addr_cfg {
uint32_t lap;
uint32_t uap;
uint32_t nap;
} __packed;
struct aptx_dec_bt_dev_addr {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct aptx_dec_bt_addr_cfg bt_addr_cfg;
} __packed;
struct asm_aptx_dec_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 sample_rate;
/* Number of samples per second.
* Supported values: 44100 and 48000 Hz
*/
} __packed;
/* Q6Core Specific */
#define AVCS_CMD_GET_FWK_VERSION (0x0001292C)
#define AVCS_CMDRSP_GET_FWK_VERSION (0x0001292D)
#define AVCS_SERVICE_ID_ALL (0xFFFFFFFF)
#define APRV2_IDS_SERVICE_ID_ADSP_CVP_V (0xB)
#define APRV2_IDS_SERVICE_ID_ADSP_AFE_V (0x4)
struct avcs_get_fwk_version {
/*
* Indicates the major version of the AVS build.
* This value is incremented on chipset family boundaries.
*/
uint32_t build_major_version;
/*
* Minor version of the AVS build.
* This value represents the mainline to which the AVS build belongs.
*/
uint32_t build_minor_version;
/* Indicates the AVS branch version to which the image belongs. */
uint32_t build_branch_version;
/* Indicates the AVS sub-branch or customer product line information. */
uint32_t build_subbranch_version;
/* Number of supported AVS services in the current build. */
uint32_t num_services;
};
struct avs_svc_api_info {
/*
* APRV2 service IDs for the individual static services.
*
* @values
* - APRV2_IDS_SERVICE_ID_ADSP_CORE_V
* - APRV2_IDS_SERVICE_ID_ADSP_AFE_V
* - APRV2_IDS_SERVICE_ID_ADSP_ASM_V
* - APRV2_IDS_SERVICE_ID_ADSP_ADM_V
* - APRV2_IDS_SERVICE_ID_ADSP_MVM_V
* - APRV2_IDS_SERVICE_ID_ADSP_CVS_V
* - APRV2_IDS_SERVICE_ID_ADSP_CVP_V
* - APRV2_IDS_SERVICE_ID_ADSP_LSM_V
*/
uint32_t service_id;
/*
* Indicates the API version of the service.
*
* Each new API update that warrants a change on the HLOS side triggers
* an increment in the version.
*/
uint32_t api_version;
/*
* Indicates the API increments on a sub-branch (not on the mainline).
*
* API branch version numbers can increment independently on different
* sub-branches.
*/
uint32_t api_branch_version;
};
struct avcs_fwk_ver_info {
struct avcs_get_fwk_version avcs_fwk_version;
struct avs_svc_api_info services[0];
} __packed;
/* LSM Specific */
#define VW_FEAT_DIM (39)
#define APRV2_IDS_SERVICE_ID_ADSP_ASM_V (0x7)
#define APRV2_IDS_SERVICE_ID_ADSP_ADM_V (0x8)
#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_SET_PARAMS_V3 (0x00012A92)
#define LSM_SESSION_CMD_GET_PARAMS_V2 (0x00012A90)
#define LSM_SESSION_CMDRSP_GET_PARAMS_V2 (0x00012A91)
#define LSM_SESSION_CMD_GET_PARAMS_V3 (0x00012A93)
#define LSM_SESSION_CMDRSP_GET_PARAMS_V3 (0x00012A94)
#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_SESSION_CMD_OPEN_TX_V3 (0x00012A95)
#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_SESSION_EVENT_DETECTION_STATUS_V3 (0x00012B04)
#define LSM_SESSION_DETECTION_ENGINE_GENERIC_EVENT (0x00012B06)
#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)
#define LSM_PARAM_ID_SWMAD_CFG (0x00012C18)
#define LSM_PARAM_ID_SWMAD_MODEL (0x00012C19)
#define LSM_PARAM_ID_SWMAD_ENABLE (0x00012C1A)
#define LSM_PARAM_ID_POLLING_ENABLE (0x00012C1B)
#define LSM_PARAM_ID_MEDIA_FMT (0x00012C1E)
#define LSM_PARAM_ID_FWK_MODE_CONFIG (0x00012C27)
#define LSM_PARAM_ID_MEDIA_FMT_V2 (0x00012C32)
#define LSM_PARAM_ID_LAB_OUTPUT_CHANNEL_CONFIG (0x00012C2D)
/* 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_SVC_CMD_SET_PARAM_V2 (0x000100fc)
#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;
#define AFE_PARAM_ID_AANC_NOISE_REDUCTION 0x000102AB
struct afe_param_id_aanc_noise_reduction {
/* Minor version used for tracking the version of the module's
* hw version
*/
uint32_t minor_version;
/* Target noise level */
int32_t ad_beta;
} __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_11_P2896_MHZ 0xAC4400
#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_P8224_MHZ 0x2b1100
#define Q6AFE_LPASS_IBIT_CLK_2_P048_MHZ 0x1F4000
#define Q6AFE_LPASS_IBIT_CLK_1_P536_MHZ 0x177000
#define Q6AFE_LPASS_IBIT_CLK_1_P4112_MHZ 0x158880
#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 INT0 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
/* Clock ID for INT1 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
/* Clock ID for INT2 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
/* Clock ID for INT3 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
/* Clock ID for INT4 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
/* Clock ID for INT5 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
/* Clock ID for INT6 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
/* Clock ID for QUINARY MI2S OSR CLK */
#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR 0x116
/* 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 Quinary PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_IBIT 0x208
/* Clock ID for Quinary PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_EBIT 0x209
/* Clock ID for QUINARY PCM OSR */
#define Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR 0x20A
/* Clock ID for Senary PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_PCM_IBIT 0x20B
/* Clock ID for Senary PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_PCM_EBIT 0x20C
/** 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 Quinary TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_IBIT 0x208
/** Clock ID for Quinary TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT 0x209
/** Clock ID for Quinary TDM OSR */
#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_OSR 0x20A
/** Clock ID for Senary TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_TDM_IBIT 0x20B
/** Clock ID for Senary TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_TDM_EBIT 0x20C
/* 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 MCLK4 */
#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
/* Clock ID for Internal Digital Codec Core */
#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
/* Clock ID for INT MCLK0 */
#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
/* Clock ID for INT MCLK1 */
#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
/*
* Clock ID for soundwire NPL.
* This is the clock to be used to enable NPL clock for internal Soundwire.
*/
#define AFE_CLOCK_SET_CLOCK_ID_SWR_NPL_CLK 0x307
/* Clock ID for MCLK5 */
#define Q6AFE_LPASS_CLK_ID_MCLK_5 0x308
/* Clock ID for AHB HDMI input */
#define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT 0x400
/* Clock ID for the primary SPDIF output core. */
#define AFE_CLOCK_SET_CLOCK_ID_PRI_SPDIF_OUTPUT_CORE 0x500
/* Clock ID for the secondary SPDIF output core. */
#define AFE_CLOCK_SET_CLOCK_ID_SEC_SPDIF_OUTPUT_CORE 0x501
/* Clock ID for the primary SPDIF input core. */
#define AFE_CLOCK_SET_CLOCK_ID_PRI_SPDIF_INPUT_CORE 0x502
/* Clock ID for the secondary SPDIF input core. */
#define AFE_CLOCK_SET_CLOCK_ID_SEC_SPDIF_INPUT_CORE 0x503
/* Clock ID for the secondary SPDIF output NPL clk. */
#define AFE_CLOCK_SET_CLOCK_ID_PRI_SPDIF_OUTPUT_NPL 0x504
/* Clock ID for the primary SPDIF output NPL clk. */
#define AFE_CLOCK_SET_CLOCK_ID_SEC_SPDIF_OUTPUT_NPL 0x505
/* 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;
};
#define AVS_BUILD_MAJOR_VERSION_V2 2
#define AVS_BUILD_MINOR_VERSION_V9 9
#define AVS_BUILD_BRANCH_VERSION_V0 0
#define AVS_BUILD_BRANCH_VERSION_V3 3
#define AFE_PARAM_ID_CLOCK_SET_V2 0x000102E6
#define AFE_API_VERSION_CLOCK_SET_V2 0x1
struct afe_param_id_clock_set_v2_t {
uint32_t clk_set_minor_version;
uint32_t clk_id;
uint32_t clk_freq_in_hz;
uint16_t clk_attri;
uint16_t clk_root;
uint32_t enable;
uint32_t divider_2x;
uint32_t m;
uint32_t n;
uint32_t d;
};
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
#define CLK_SRC_NAME_MAX 32
enum {
CLK_SRC_INTEGRAL,
CLK_SRC_FRACT,
CLK_SRC_MAX
};
struct afe_set_clk_drift {
/*
* Clock drift (in PPB) to be set.
* @values
* - need to get values from DSP team
*/
int32_t clk_drift;
/*
* Clock reset.
* @values
* - 1 -- Reset PLL with the original frequency
* - 0 -- Adjust the clock with the clk drift value
*/
uint32_t clk_reset;
/*
* Clock src name.
* @values
* - values to be set from machine driver
* - LPAPLL0 -- integral clk src
* - LPAPLL2 -- fractional clk src
*/
char clk_src_name[CLK_SRC_NAME_MAX];
} __packed;
/* This param id is used to adjust audio interface PLL*/
#define AFE_PARAM_ID_CLOCK_ADJUST 0x000102C6
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;
/*
* 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_v1 {
/* APR Header */
struct apr_hdr apr_hdr;
/* The total size of the payload, including param_hdr_v3 */
uint32_t payload_size;
/* The memory mapping header to be used when sending outband */
struct mem_mapping_hdr mem_hdr;
/* The parameter data to be filled when sent inband */
u32 param_data[0];
} __packed;
struct afe_svc_cmd_set_param_v2 {
/* APR Header */
struct apr_hdr apr_hdr;
/* The memory mapping header to be used when sending outband */
struct mem_mapping_hdr mem_hdr;
/* The total size of the payload, including param_hdr_v3 */
u32 payload_size;
/* The parameter data to be filled when sent inband */
u32 param_data[0];
} __packed;
struct afe_param_hw_mad_ctrl {
uint32_t minor_version;
uint16_t mad_type;
uint16_t mad_enable;
} __packed;
struct afe_port_cmd_set_aanc_acdb_table {
struct apr_hdr hdr;
struct mem_mapping_hdr mem_hdr;
} __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
/* 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)
#define AFE_GROUP_DEVICE_ID_QUINARY_TDM_RX \
(AFE_PORT_ID_QUINARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_QUINARY_TDM_TX \
(AFE_PORT_ID_QUINARY_TDM_TX + 0x100)
#define AFE_GROUP_DEVICE_ID_SENARY_TDM_RX \
(AFE_PORT_ID_SENARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_SENARY_TDM_TX \
(AFE_PORT_ID_SENARY_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 active channels = num of active slots * num of active lanes.
* @values 1 to 64
*/
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;
/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to specify
* the timing statistics of the corresponding device interface.
* Client can periodically query for the device time statistics to help adjust
* the PLL based on the drift value. The get param command must be sent to
* AFE port ID corresponding to device interface
* This parameter ID supports following get param commands:
* #AFE_PORT_CMD_GET_PARAM_V2 and
* #AFE_PORT_CMD_GET_PARAM_V3.
*/
#define AFE_PARAM_ID_DEV_TIMING_STATS 0x000102AD
/* Version information used to handle future additions to AFE device
* interface timing statistics (for backward compatibility).
*/
#define AFE_API_VERSION_DEV_TIMING_STATS 0x1
/* Enumeration for specifying a sink(Rx) device */
#define AFE_SINK_DEVICE 0x0
/* Enumeration for specifying a source(Tx) device */
#define AFE_SOURCE_DEVICE 0x1
/* Enumeration for specifying the drift reference is of type AV Timer */
#define AFE_REF_TIMER_TYPE_AVTIMER 0x0
/* Message payload structure for the
* AFE_PARAM_ID_DEV_TIMING_STATS parameter.
*/
struct afe_param_id_dev_timing_stats {
/* Minor version used to track the version of device interface timing
* statistics. Currently, the supported version is 1.
* @values #AFE_API_VERSION_DEV_TIMING_STATS
*/
u32 minor_version;
/* Indicates the device interface direction as either
* source (Tx) or sink (Rx).
* @values
* #AFE_SINK_DEVICE
* #AFE_SOURCE_DEVICE
*/
u16 device_direction;
/* Reference timer for drift accumulation and time stamp information.
* @values
* #AFE_REF_TIMER_TYPE_AVTIMER @tablebulletend
*/
u16 reference_timer;
/*
* Flag to indicate if resync is required on the client side for
* drift correction. Flag is set to TRUE for the first get_param
* response after device interface starts. This flag value can be
* used by client to identify if device interface restart has
* happened and if any re-sync is required at their end for drift
* correction.
* @values
* 0: FALSE (Resync not required)
* 1: TRUE (Resync required) @tablebulletend
*/
u32 resync_flag;
/* Accumulated drift value in microseconds. This value is updated
* every 100th ms.
* Positive drift value indicates AV timer is running faster than device
* Negative drift value indicates AV timer is running slower than device
* @values Any valid int32 number
*/
s32 acc_drift_value;
/* Lower 32 bits of the 64-bit absolute timestamp of reference
* timer in microseconds.
* This timestamp corresponds to the time when the drift values
* are accumlated for every 100th ms.
* @values Any valid uint32 number
*/
u32 ref_timer_abs_ts_lsw;
/* Upper 32 bits of the 64-bit absolute timestamp of reference
* timer in microseconds.
* This timestamp corresponds to the time when the drift values
* are accumlated for every 100th ms.
* @values Any valid uint32 number
*/
u32 ref_timer_abs_ts_msw;
} __packed;
struct afe_av_dev_drift_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct afe_param_id_dev_timing_stats timing_stats;
} __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)
*/
};
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
* audio client choose the rendering decision that the audio DSP should use.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD 0x00012F0D
/* Indicates that rendering decision will be based on default rate
* (session clock based rendering, device driven).
* 1. The default session clock based rendering is inherently driven
* by the timing of the device.
* 2. After the initial decision is made (first buffer after a run
* command), subsequent data rendering decisions are made with
* respect to the rate at which the device is rendering, thus deriving
* its timing from the device.
* 3. While this decision making is simple, it has some inherent limitations
* (mentioned in the next section).
* 4. If this API is not set, the session clock based rendering will be assumed
* and this will ensure that the DSP is backward compatible.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT 0
/* Indicates that rendering decision will be based on local clock rate.
* 1. In the DSP loopback/client loopback use cases (frame based
* inputs), the incoming data into audio DSP is time-stamped at the
* local clock rate (STC).
* 2. This TS rate may match the incoming data rate or maybe different
* from the incoming data rate.
* 3. Regardless, the data will be time-stamped with local STC and
* therefore, the client is recommended to set this mode for these
* use cases. This method is inherently more robust to sequencing
* (AFE Start/Stop) and device switches, among other benefits.
* 4. This API will inform the DSP to compare every incoming buffer TS
* against local STC.
* 5. DSP will continue to honor render windows APIs, as before.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC 1
/* Structure for rendering decision parameter */
struct asm_session_mtmx_strtr_param_render_mode_t {
/* Specifies the type of rendering decision the audio DSP should use.
*
* @values
* - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT
* - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC
*/
u32 flags;
} __packed;
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
* audio client to specify the clock recovery mechanism that the audio DSP
* should use.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD 0x00012F0E
/* Indicates that default clock recovery will be used (no clock recovery).
* If the client wishes that no clock recovery be done, the client can
* choose this. This means that no attempt will made by the DSP to try and
* match the rates of the input and output audio.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE 0
/* Indicates that independent clock recovery needs to be used.
* 1. In the DSP loopback/client loopback use cases (frame based inputs),
* the client should choose the independent clock recovery option.
* 2. This basically de-couples the audio and video from knowing each others
* clock sources and lets the audio DSP independently rate match the input
* and output rates.
* 3. After drift detection, the drift correction is achieved by either pulling
* the PLLs (if applicable) or by stream to device rate matching
* (for PCM use cases) by comparing drift with respect to STC.
* 4. For passthrough use cases, since the PLL pulling is the only option,
* a best effort will be made.
* If PLL pulling is not possible / available, the rendering will be
* done without rate matching.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO 1
/* Payload of the #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC parameter.
*/
struct asm_session_mtmx_strtr_param_clk_rec_t {
/* Specifies the type of clock recovery that the audio DSP should
* use for rate matching.
*/
/* @values
* #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_DEFAULT
* #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_INDEPENDENT
*/
u32 flags;
} __packed;
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to
* realize smoother adjustment of audio session clock for a specified session.
* The desired audio session clock adjustment(in micro seconds) is specified
* using the command #ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2.
* Delaying/Advancing the session clock would be implemented by inserting
* interpolated/dropping audio samples in the playback path respectively.
* Also, this parameter has to be configured before the Audio Session is put
* to RUN state to avoid cold start latency/glitches in the playback.
*/
#define ASM_SESSION_MTMX_PARAM_ADJUST_SESSION_TIME_CTL 0x00013217
struct asm_session_mtmx_param_adjust_session_time_ctl_t {
/* Specifies whether the module is enabled or not
* @values
* 0 -- disabled
* 1 -- enabled
*/
u32 enable;
};
union asm_session_mtmx_strtr_param_config {
struct asm_session_mtmx_strtr_param_window_v2_t window_param;
struct asm_session_mtmx_strtr_param_render_mode_t render_param;
struct asm_session_mtmx_strtr_param_clk_rec_t clk_rec_param;
struct asm_session_mtmx_param_adjust_session_time_ctl_t adj_time_param;
} __packed;
struct asm_mtmx_strtr_params {
struct apr_hdr hdr;
struct asm_session_cmd_set_mtmx_strstr_params_v2 param;
struct param_hdr_v1 data;
union asm_session_mtmx_strtr_param_config config;
} __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 param_hdr_v1 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,
COMPRESSED_PASSTHROUGH_DSD,
LISTEN,
COMPRESSED_PASSTHROUGH_GEN,
COMPRESSED_PASSTHROUGH_IEC61937
};
#define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770
#define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771
struct adm_set_compressed_device_mute {
u32 mute_on;
} __packed;
#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E
#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F
struct adm_set_compressed_device_latency {
u32 latency;
} __packed;
#define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6
#define VOICEPROC_MODULE_ID_FLUENCE_PRO_VC_TX 0x00010F35
#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
#define MAX_DOA_TRACKING_ANGLES 2
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 doa_tracking_mon_param {
uint16_t target_angle_L16[MAX_DOA_TRACKING_ANGLES];
uint16_t interf_angle_L16[MAX_DOA_TRACKING_ANGLES];
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_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;
struct adm_param_doa_tracking_mon_t {
uint16_t target_angle_L16[MAX_DOA_TRACKING_ANGLES];
uint16_t interf_angle_L16[MAX_DOA_TRACKING_ANGLES];
uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
} __packed;
struct afe_doa_tracking_mon_get_param_resp {
uint32_t status;
struct param_hdr_v3 pdata;
struct doa_tracking_mon_param doa;
} __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
#define AUDPROC_MODULE_ID_VOICE_TX_SECNS 0x10027059
#define AUDPROC_PARAM_IDX_SEC_PRIMARY_MIC_CH 0x10014444
#define AUDPROC_MODULE_ID_FFECNS 0x00010952
#define AUDPROC_MODULE_ID_FFNS 0x00010962
#define AUDPROC_PARAM_ID_FFV_DOA_TRACKING_PARAM 0x0001097C
/*
* ID of the DTMF Detection Module.
*/
#define AUDPROC_MODULE_ID_DTMF_DETECTION 0x00010940
#define AUDPROC_PARAM_ID_FFV_DOA_TRACKING_MONITOR 0x0001097D
struct admx_sec_primary_mic_ch {
uint16_t version;
uint16_t reserved;
uint16_t sec_primary_mic_ch;
uint16_t reserved1;
} __packed;
#define FFECNS_MODULE_ID 0x00010952
#define FLUENCE_CMN_GLOBAL_EFFECT_PARAM_ID 0x00010EAF
#define FFECNS_TOPOLOGY 0X10028003
struct ffecns_effect {
uint32_t payload;
};
/** ID of the Voice Activity Detection (VAD) module, which is used to
* configure AFE VAD.
*/
#define AFE_MODULE_VAD 0x000102B9
/** ID of the parameter used by #AFE_MODULE_VAD to configure the VAD.
*/
#define AFE_PARAM_ID_VAD_CFG 0x000102BA
#define AFE_API_VERSION_VAD_CFG 0x1
/* Payload of the AFE_PARAM_ID_VAD_CONFIG parameter used by
* AFE_MODULE_VAD.
*/
struct afe_param_id_vad_cfg_t {
uint32_t vad_cfg_minor_version;
/** Tracks the configuration of this parameter.
* Supported Values: #AFE_API_VERSION_VAD_CFG
*/
uint32_t pre_roll_in_ms;
/** Pre-roll period in ms.
* Supported Values: 0x0 to 0x3E8
*/
} __packed;
#define AFE_PARAM_ID_VAD_CORE_CFG 0x000102BB
/**
* This parameter is should be used to configure the AFE TDM
* interface lane configuration.
* Regular TDM interface ports:
* This parameter ID must be used with module ID
* AFE_MODULE_AUDIO_DEV_INTERFACE and set using the AFE_PORT_CMD_SET_PARAM_V3
* command for the AFE TDM interface port IDs.
* Group device TDM interface ports:
* This parameter ID must be used with module ID AFE_MODULE_GROUP_DEVICE
* and set using the AFE_SVC_CMD_SET_PARAM_V2 command for the AFE TDM group IDs.
*/
#define AFE_PARAM_ID_TDM_LANE_CONFIG 0x000102C1
/* Payload of the AFE_PARAM_ID_TDM_LANE_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_tdm_lane_cfg {
uint16_t port_id;
/** ID of the TDM interface.
* For regular TDM interfaces value corresponds to valid port ID.
* For group devices TDM interface value corresponds to valid group device ID.
*/
uint16_t lane_mask;
/** Position of the active lanes. Bits 0 to N correspond to lanes 0 to N.
* 1 to 2^N-1 When a bit is set, the corresponding lane is active.
* The number of active lanes can be inferred from the number of bits
* set in the mask.
*/
};
/** ID of the parameter used to set the AFE port data logging to enable or disable state.
* For non-group device use cases, #AFE_MODULE_AUDIO_DEV_INTERFACE uses this
* parameter to configure the flag used for data logging in afe_data_logging_t
* of the respective port to enabled or disabled state.
* The HLOS client can use this parameter to configure the data logging
* disable flag for it's respective port.
* The reason for this parameter addition is if a number of ports are
* configured and running, Upon enabling logging through 0x1586 tap point,
* we will get input/output logs for all the enabled ports.
* In order to disabled logging for a specific port for which data logging
* is not needed, the HLOS client can make use of AFE_PORT_DATA_LOGGING_DISABLE flag.
* This flag will set to AFE_PORT_DATA_LOGGING_ENABLE during port initialization and also
* during port stop. If port is restarted, the set param should be called again
* by the HLOS client if needed to disable data logging.
* @par
* If HLOS client doesn't set this paramter, by default the disable flag = AFE_PORT_DATA_LOGGING_ENABLE.
* If HLOS client sets the flag = AFE_PORT_DATA_LOGGING_DISABLE, the respective port will be disabled for data logging.
*/
#define AFE_PARAM_ID_PORT_DATA_LOGGING_DISABLE 0x000102E9
/** Enable flag for port data logging. */
#define AFE_PORT_DATA_LOGGING_ENABLE 0
/** Disable flag for port data logging. */
#define AFE_PORT_DATA_LOGGING_DISABLE 1
/*
* Payload of the AFE_PARAM_ID_PORT_DATA_LOGGING_DISABLE parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE
*/
struct afe_param_id_port_data_log_disable_t
{
uint32_t disable_logging_flag;
/** Flag for enabling or disabling data logging.
* @values
* - AFE_PORT_DATA_LOGGING_ENABLE - enable data logging.
* - AFE_PORT_DATA_LOGGING_DISABLE - disable data logging.
*/
} __packed;
#endif /*_APR_AUDIO_V2_H_ */
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