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gerrit-public.fairphone.software / fp2-dev / kernel / msm / 45ab7751c04df1f49b4a4f1b62dfcb17cbb4e7a0 / . / include / sound / apr_audio-v2.h
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/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
#ifndef _APR_AUDIO_V2_H_
#define _APR_AUDIO_V2_H_
#include <mach/qdsp6v2/apr.h>
#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
/* Enumeration for an audio Rx matrix ID.*/
#define ADM_MATRIX_ID_AUDIO_RX 0
#define ADM_MATRIX_ID_AUDIO_TX 1
/* Enumeration for an audio Tx matrix ID.*/
#define ADM_MATRIX_ID_AUDIOX 1
#define ADM_MAX_COPPS 5
/* 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
/* Indicates that endpoint_id_2 is to be ignored.*/
#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3
/* Indicates that an audio COPP is to send/receive a mono PCM
* stream to/from
* END_POINT_ID_1.
*/
#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1
/* Indicates that an audio COPP is to send/receive a
* stereo PCM stream to/from END_POINT_ID_1.
*/
#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2
/* Sample rate is 8000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000
/* Sample rate is 16000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000
/* Sample rate is 48000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000
/* Definition for a COPP live input flag bitmask.*/
#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U)
/* Definition for a COPP live shift value bitmask.*/
#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0
/* Definition for the COPP ID bitmask.*/
#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL)
/* Definition for the COPP ID shift value.*/
#define ADM_SHIFT_COPP_ID 0
/* Definition for the service ID bitmask.*/
#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
/* Definition for the service ID shift value.*/
#define ADM_SHIFT_SERVICE_ID 16
/* Definition for the domain ID bitmask.*/
#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
/* Definition for the domain ID shift value.*/
#define ADM_SHIFT_DOMAIN_ID 24
/* ADM device open command payload of the
#ADM_CMD_DEVICE_OPEN_V5 command.
*/
struct adm_cmd_device_open_v5 {
struct apr_hdr hdr;
u16 flags;
/* Reserved for future use. Clients must set this field
* to zero.
*/
u16 mode_of_operation;
/* Specifies whether the COPP must be opened on the Tx or Rx
* path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
* supported values and interpretation.
* Supported values:
* - 0x1 -- Rx path COPP
* - 0x2 -- Tx path live COPP
* - 0x3 -- Tx path nonlive COPP
* Live connections cause sample discarding in the Tx device
* matrix if the destination output ports do not pull them
* fast enough. Nonlive connections queue the samples
* indefinitely.
*/
u16 endpoint_id_1;
/* Logical and physical endpoint ID of the audio path.
* If the ID is a voice processor Tx block, it receives near
* samples. Supported values: Any pseudoport, AFE Rx port,
* or AFE Tx port For a list of valid IDs, refer to
* @xhyperref{Q4,[Q4]}.
* Q4 = Hexagon Multimedia: AFE Interface Specification
*/
u16 endpoint_id_2;
/* Logical and physical endpoint ID 2 for a voice processor
* Tx block.
* This is not applicable to audio COPP.
* Supported values:
* - AFE Rx port
* - 0xFFFF -- Endpoint 2 is unavailable and the voice
* processor Tx
* block ignores this endpoint
* When the voice processor Tx block is created on the audio
* record path,
* it can receive far-end samples from an AFE Rx port if the
* voice call
* is active. The ID of the AFE port is provided in this
* field.
* For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
*/
u32 topology_id;
/* Audio COPP topology ID; 32-bit GUID. */
u16 dev_num_channel;
/* Number of channels the audio COPP sends to/receives from
* the endpoint.
* Supported values: 1 to 8.
* The value is ignored for the voice processor Tx block,
* where channel
* configuration is derived from the topology ID.
*/
u16 bit_width;
/* Bit width (in bits) that the audio COPP sends to/receives
* from the
* endpoint. The value is ignored for the voice processing
* Tx block,
* where the PCM width is 16 bits.
*/
u32 sample_rate;
/* Sampling rate at which the audio COPP/voice processor
* Tx block
* interfaces with the endpoint.
* Supported values for voice processor Tx: 8000, 16000,
* 48000 Hz
* Supported values for audio COPP: >0 and <=192 kHz
*/
u8 dev_channel_mapping[8];
/* Array of channel mapping of buffers that the audio COPP
* sends to the endpoint. Channel[i] mapping describes channel
* I inside the buffer, where 0 < i < dev_num_channel.
* This value is relevent only for an audio Rx COPP.
* For the voice processor block and Tx audio block, this field
* is set to zero and is ignored.
*/
} __packed;
/*
* This command allows the client to close a COPP and disconnect
* the device session.
*/
#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
/* Sets one or more parameters to a COPP.
*/
#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328
/* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command.
* If the data_payload_addr_lsw and data_payload_addr_msw element
* are NULL, a series of adm_param_datastructures immediately
* follows, whose total size is data_payload_size bytes.
*/
struct adm_cmd_set_pp_params_v5 {
struct apr_hdr hdr;
u32 payload_addr_lsw;
/* LSW of parameter data payload address.*/
u32 payload_addr_msw;
/* MSW of parameter data payload address.*/
u32 mem_map_handle;
/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS
* command */
/* If mem_map_handle is zero implies the message is in
* the payload */
u32 payload_size;
/* Size in bytes of the variable payload accompanying this
* message or
* in shared memory. This is used for parsing the parameter
* payload.
*/
} __packed;
/* Payload format for COPP parameter data.
* Immediately following this structure are param_size bytes
* of parameter
* data.
*/
struct adm_param_data_v5 {
u32 module_id;
/* Unique ID of the module. */
u32 param_id;
/* Unique ID of the parameter. */
u16 param_size;
/* Data size of the param_id/module_id combination.
This value is a
multiple of 4 bytes. */
u16 reserved;
/* Reserved for future enhancements.
* This field must be set to zero.
*/
} __packed;
/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
*/
#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message,
* which returns the
* status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
*/
struct adm_cmd_rsp_device_open_v5 {
u32 status;
/* Status message (error code).*/
u16 copp_id;
/* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/
u16 reserved;
/* Reserved. This field must be set to zero.*/
} __packed;
/* This command allows a query of one COPP parameter.
*/
#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A
/* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command.
*/
struct adm_cmd_get_pp_params_v5 {
u32 data_payload_addr_lsw;
/* LSW of parameter data payload address.*/
u32 data_payload_addr_msw;
/* MSW of parameter data payload address.*/
/* If the mem_map_handle is non zero,
* on ACK, the ParamData payloads begin at
* the address specified (out-of-band).
*/
u32 mem_map_handle;
/* Memory map handle returned
* by ADM_CMD_SHARED_MEM_MAP_REGIONS command.
* If the mem_map_handle is 0, it implies that
* the ACK's payload will contain the ParamData (in-band).
*/
u32 module_id;
/* Unique ID of the module. */
u32 param_id;
/* Unique ID of the parameter. */
u16 param_max_size;
/* Maximum data size of the parameter
*ID/module ID combination. This
* field is a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for future enhancements.
* This field must be set to zero.
*/
} __packed;
/* Returns parameter values
* in response to an #ADM_CMD_GET_PP_PARAMS_V5 command.
*/
#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B
/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message,
* which returns parameter values in response
* to an #ADM_CMD_GET_PP_PARAMS_V5 command.
* Immediately following this
* structure is the adm_param_data_v5
* structure containing the pre/postprocessing
* parameter data. For an in-band
* scenario, the variable payload depends
* on the size of the parameter.
*/
struct adm_cmd_rsp_get_pp_params_v5 {
u32 status;
/* Status message (error code).*/
} __packed;
/* Allows a client to control the gains on various session-to-COPP paths.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C
/* Indicates that the target gain in the
* current adm_session_copp_gain_v5
* structure is to be applied to all
* the session-to-COPP paths that exist for
* the specified session.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
/* Indicates that the target gain is
* to be immediately applied to the
* specified session-to-COPP path,
* without a ramping fashion.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000
/* Enumeration for a linear ramping curve.*/
#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000
/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
* Immediately following this structure are num_gains of the
* adm_session_copp_gain_v5structure.
*/
struct adm_cmd_matrix_ramp_gains_v5 {
u32 matrix_id;
/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
* Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
* macros to set this field.
*/
u16 num_gains;
/* Number of gains being applied. */
u16 reserved_for_align;
/* Reserved. This field must be set to zero.*/
} __packed;
/* Session-to-COPP path gain structure, used by the
* #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
* This structure specifies the target
* gain (per channel) that must be applied
* to a particular session-to-COPP path in
* the audio matrix. The structure can
* also be used to apply the gain globally
* to all session-to-COPP paths that
* exist for the given session.
* The aDSP uses device channel mapping to
* determine which channel gains to
* use from this command. For example,
* if the device is configured as stereo,
* the aDSP uses only target_gain_ch_1 and
* target_gain_ch_2, and it ignores
* the others.
*/
struct adm_session_copp_gain_v5 {
u16 session_id;
/* Handle of the ASM session.
* Supported values: 1 to 8.
*/
u16 copp_id;
/* Handle of the COPP. Gain will be applied on the Session ID
* COPP ID path.
*/
u16 ramp_duration;
/* Duration (in milliseconds) of the ramp over
* which target gains are
* to be applied. Use
* #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE
* to indicate that gain must be applied immediately.
*/
u16 step_duration;
/* Duration (in milliseconds) of each step in the ramp.
* This parameter is ignored if ramp_duration is equal to
* #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE.
* Supported value: 1
*/
u16 ramp_curve;
/* Type of ramping curve.
* Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR
*/
u16 reserved_for_align;
/* Reserved. This field must be set to zero. */
u16 target_gain_ch_1;
/* Target linear gain for channel 1 in Q13 format; */
u16 target_gain_ch_2;
/* Target linear gain for channel 2 in Q13 format; */
u16 target_gain_ch_3;
/* Target linear gain for channel 3 in Q13 format; */
u16 target_gain_ch_4;
/* Target linear gain for channel 4 in Q13 format; */
u16 target_gain_ch_5;
/* Target linear gain for channel 5 in Q13 format; */
u16 target_gain_ch_6;
/* Target linear gain for channel 6 in Q13 format; */
u16 target_gain_ch_7;
/* Target linear gain for channel 7 in Q13 format; */
u16 target_gain_ch_8;
/* Target linear gain for channel 8 in Q13 format; */
} __packed;
/* Allows to set mute/unmute on various session-to-COPP paths.
* For every session-to-COPP path (stream-device interconnection),
* mute/unmute can be set individually on the output channels.
*/
#define ADM_CMD_MATRIX_MUTE_V5 0x0001032D
/* Indicates that mute/unmute in the
* current adm_session_copp_mute_v5structure
* is to be applied to all the session-to-COPP
* paths that exist for the specified session.
*/
#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
/* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/
struct adm_cmd_matrix_mute_v5 {
u32 matrix_id;
/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
* Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
* macros to set this field.
*/
u16 session_id;
/* Handle of the ASM session.
* Supported values: 1 to 8.
*/
u16 copp_id;
/* Handle of the COPP.
* Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS
* to indicate that mute/unmute must be applied to
* all the COPPs connected to session_id.
* Supported values:
* - 0xFFFF -- Apply mute/unmute to all connected COPPs
* - Other values -- Valid COPP ID
*/
u8 mute_flag_ch_1;
/* Mute flag for channel 1 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_2;
/* Mute flag for channel 2 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_3;
/* Mute flag for channel 3 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_4;
/* Mute flag for channel 4 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_5;
/* Mute flag for channel 5 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_6;
/* Mute flag for channel 6 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_7;
/* Mute flag for channel 7 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_8;
/* Mute flag for channel 8 is set to unmute (0) or mute (1). */
u16 ramp_duration;
/* Period (in milliseconds) over which the soft mute/unmute will be
* applied.
* Supported values: 0 (Default) to 0xFFFF
* The default of 0 means mute/unmute will be applied immediately.
*/
u16 reserved_for_align;
/* Clients must set this field to zero.*/
} __packed;
/* 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 /* index = 0 */
#define PRIMARY_I2S_TX 1 /* index = 1 */
#define PCM_RX 2 /* index = 2 */
#define PCM_TX 3 /* index = 3 */
#define SECONDARY_I2S_RX 4 /* index = 4 */
#define SECONDARY_I2S_TX 5 /* index = 5 */
#define MI2S_RX 6 /* index = 6 */
#define MI2S_TX 7 /* index = 7 */
#define HDMI_RX 8 /* index = 8 */
#define RSVD_2 9 /* index = 9 */
#define RSVD_3 10 /* index = 10 */
#define DIGI_MIC_TX 11 /* index = 11 */
#define VOICE_RECORD_RX 0x8003 /* index = 12 */
#define VOICE_RECORD_TX 0x8004 /* index = 13 */
#define VOICE_PLAYBACK_TX 0x8005 /* index = 14 */
/* Slimbus Multi channel port id pool */
#define SLIMBUS_0_RX 0x4000 /* index = 15 */
#define SLIMBUS_0_TX 0x4001 /* index = 16 */
#define SLIMBUS_1_RX 0x4002 /* index = 17 */
#define SLIMBUS_1_TX 0x4003 /* index = 18 */
#define SLIMBUS_2_RX 0x4004
#define SLIMBUS_2_TX 0x4005
#define SLIMBUS_3_RX 0x4006
#define SLIMBUS_3_TX 0x4007
#define SLIMBUS_4_RX 0x4008
#define SLIMBUS_4_TX 0x4009 /* index = 24 */
#define INT_BT_SCO_RX 0x3000 /* index = 25 */
#define INT_BT_SCO_TX 0x3001 /* index = 26 */
#define INT_BT_A2DP_RX 0x3002 /* index = 27 */
#define INT_FM_RX 0x3004 /* index = 28 */
#define INT_FM_TX 0x3005 /* index = 29 */
#define RT_PROXY_PORT_001_RX 0x2000 /* index = 30 */
#define RT_PROXY_PORT_001_TX 0x2001 /* index = 31 */
#define AFE_PORT_INVALID 0xFFFF
#define SLIMBUS_INVALID AFE_PORT_INVALID
#define AFE_PORT_CMD_START 0x000100ca
#define AFE_EVENT_RTPORT_START 0
#define AFE_EVENT_RTPORT_STOP 1
#define AFE_EVENT_RTPORT_LOW_WM 2
#define AFE_EVENT_RTPORT_HI_WM 3
#define ADSP_AFE_VERSION 0x00200000
/* Size of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE 0xF
/* Size of the range of port IDs for internal BT-FM ports. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE 0x6
/* Size of the range of port IDs for SLIMbus<sup>&reg;
* </sup> multichannel
* ports.
*/
#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE 0xA
/* Size of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE 0x2
/* Size of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE 0x5
/* Start of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START 0x1000
/* End of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \
(AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\
AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1)
/* Start of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_START 0x2000
/* End of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \
(AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\
AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1)
/* Start of the range of port IDs for internal BT-FM devices. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START 0x3000
/* End of the range of port IDs for internal BT-FM devices. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \
(AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\
AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1)
/* Start of the range of port IDs for SLIMbus devices. */
#define AFE_PORT_ID_SLIMBUS_RANGE_START 0x4000
/* End of the range of port IDs for SLIMbus devices. */
#define AFE_PORT_ID_SLIMBUS_RANGE_END \
(AFE_PORT_ID_SLIMBUS_RANGE_START +\
AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1)
/* Start of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_START 0x8001
/* End of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \
(AFE_PORT_ID_PSEUDOPORT_RANGE_START +\
AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1)
#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_RT_PROXY_PORT_001_RX 0x2000
#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001
#define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000
#define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001
#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002
#define AFE_PORT_ID_INTERNAL_FM_RX 0x3004
#define AFE_PORT_ID_INTERNAL_FM_TX 0x3005
/* SLIMbus Rx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
/* SLIMbus Tx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
/* SLIMbus Rx port on channel 1. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
/* SLIMbus Tx port on channel 1. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
/* SLIMbus Rx port on channel 2. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
/* SLIMbus Tx port on channel 2. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
/* SLIMbus Rx port on channel 3. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
/* SLIMbus Tx port on channel 3. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
/* SLIMbus Rx port on channel 4. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
/* SLIMbus Tx port on channel 4. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
/* SLIMbus Rx port on channel 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
/* 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_VOICE_PLAYBACK_TX 0x8005
#define AFE_PORT_ID_INVALID 0xFFFF
#define AAC_ENC_MODE_AAC_LC 0x02
#define AAC_ENC_MODE_AAC_P 0x05
#define AAC_ENC_MODE_EAAC_P 0x1D
#define AFE_PSEUDOPORT_CMD_START 0x000100cf
struct afe_pseudoport_start_command {
struct apr_hdr hdr;
u16 port_id; /* Pseudo Port 1 = 0x8000 */
/* Pseudo Port 2 = 0x8001 */
/* Pseudo Port 3 = 0x8002 */
u16 timing; /* FTRT = 0 , AVTimer = 1, */
} __packed;
#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
struct afe_pseudoport_stop_command {
struct apr_hdr hdr;
u16 port_id; /* Pseudo Port 1 = 0x8000 */
/* Pseudo Port 2 = 0x8001 */
/* Pseudo Port 3 = 0x8002 */
u16 reserved;
} __packed;
#define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202
#define AFE_PARAM_ID_ENABLE 0x00010203
/* Payload of the #AFE_PARAM_ID_ENABLE
* parameter, which enables or
* disables any module.
* The fixed size of this structure is four bytes.
*/
struct afe_mod_enable_param {
u16 enable;
/* Enables (1) or disables (0) the module. */
u16 reserved;
/* This field must be set to zero.
*/
} __packed;
/* ID of the configuration parameter used by the
* #AFE_MODULE_SIDETONE_IIR_FILTER module.
*/
#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204
struct afe_sidetone_iir_filter_config_params {
u16 num_biquad_stages;
/* Number of stages.
* Supported values: Minimum of 5 and maximum of 10
*/
u16 pregain;
/* Pregain for the compensating filter response.
* Supported values: Any number in Q13 format
*/
} __packed;
#define AFE_MODULE_LOOPBACK 0x00010205
#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206
/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter,
* which gets/sets loopback gain of a port to an Rx port.
* The Tx port ID of the loopback is part of the set_param command.
*/
/* Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's
* configuration/calibration settings for the AFE port.
*/
struct afe_port_cmd_set_param_v2 {
u16 port_id;
/* Port interface and direction (Rx or Tx) to start.
*/
u16 payload_size;
/* Actual size of the payload in bytes.
* This is used for parsing the parameter payload.
* Supported values: > 0
*/
u32 payload_address_lsw;
/* LSW of 64 bit Payload address.
* Address should be 32-byte,
* 4kbyte aligned and must be contiguous memory.
*/
u32 payload_address_msw;
/* MSW of 64 bit Payload address.
* In case of 32-bit shared memory address,
* this field must be set to zero.
* In case of 36-bit shared memory address,
* bit-4 to bit-31 must be set to zero.
* Address should be 32-byte, 4kbyte aligned
* and must be contiguous memory.
*/
u32 mem_map_handle;
/* Memory map handle returned by
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
* Supported Values:
* - NULL -- Message. The parameter data is in-band.
* - Non-NULL -- The parameter data is Out-band.Pointer to
* the physical address
* in shared memory of the payload data.
* An optional field is available if parameter
* data is in-band:
* afe_param_data_v2 param_data[...].
* For detailed payload content, see the
* afe_port_param_data_v2 structure.
*/
} __packed;
#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
struct afe_port_param_data_v2 {
u32 module_id;
/* ID of the module to be configured.
* Supported values: Valid module ID
*/
u32 param_id;
/* ID of the parameter corresponding to the supported parameters
* for the module ID.
* Supported values: Valid parameter ID
*/
u16 param_size;
/* Actual size of the data for the
* module_id/param_id pair. The size is a
* multiple of four bytes.
* Supported values: > 0
*/
u16 reserved;
/* This field must be set to zero.
*/
} __packed;
struct afe_loopback_gain_per_path_param {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
u16 rx_port_id;
/* Rx port of the loopback. */
u16 gain;
/* Loopback gain per path of the port.
* Supported values: Any number in Q13 format
*/
} __packed;
/* Parameter ID used to configure and enable/disable the
* loopback path. The difference with respect to the existing
* API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be
* configured as source port in loopback path. Port-id in
* AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be
* Tx or Rx port. In addition, we can configure the type of
* routing mode to handle different use cases.
*/
#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
enum afe_loopback_routing_mode {
LB_MODE_DEFAULT = 1,
/* Regular loopback from source to destination port */
LB_MODE_SIDETONE,
/* Sidetone feed from Tx source to Rx destination port */
LB_MODE_EC_REF_VOICE_AUDIO,
/* Echo canceller reference, voice + audio + DTMF */
LB_MODE_EC_REF_VOICE
/* Echo canceller reference, voice alone */
} __packed;
/* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG ,
* which enables/disables one AFE loopback.
*/
struct afe_loopback_cfg_v1 {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
u32 loopback_cfg_minor_version;
/* Minor version used for tracking the version of the RMC module
* configuration interface.
* Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
*/
u16 dst_port_id;
/* Destination Port Id. */
u16 routing_mode;
/* Specifies data path type from src to dest port.
* Supported values:
* #LB_MODE_DEFAULT
* #LB_MODE_SIDETONE
* #LB_MODE_EC_REF_VOICE_AUDIO
* #LB_MODE_EC_REF_VOICE_A
* #LB_MODE_EC_REF_VOICE
*/
u16 enable;
/* Specifies whether to enable (1) or
* disable (0) an AFE loopback.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.
*/
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION 0x00010209
#define AFE_PARAM_ID_SPKR_PROT_CONFIG 0x0001020a
#define AFE_API_VERSION_SPKR_PROT_CONFIG 0x1
#define AFE_SPKR_PROT_EXCURSIONF_LEN 512
struct afe_spkr_prot_cfg_param_v1 {
u32 spkr_prot_minor_version;
/*
* Minor version used for tracking the version of the
* speaker protection module configuration interface.
* Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG
*/
int16_t win_size;
/* Analysis and synthesis window size (nWinSize).
* Supported values: 1024, 512, 256 samples
*/
int16_t margin;
/* Allowable margin for excursion prediction,
* in L16Q15 format. This is a
* control parameter to allow
* for overestimation of peak excursion.
*/
int16_t spkr_exc_limit;
/* Speaker excursion limit, in L16Q15 format.*/
int16_t spkr_resonance_freq;
/* Resonance frequency of the speaker; used
* to define a frequency range
* for signal modification.
*
* Supported values: 0 to 2000 Hz */
int16_t limhresh;
/* Threshold of the hard limiter; used to
* prevent overshooting beyond a
* signal level that was set by the limiter
* prior to speaker protection.
* Supported values: 0 to 32767
*/
int16_t hpf_cut_off_freq;
/* High pass filter cutoff frequency.
* Supported values: 100, 200, 300 Hz
*/
int16_t hpf_enable;
/* Specifies whether the high pass filter
* is enabled (0) or disabled (1).
*/
int16_t reserved;
/* This field must be set to zero. */
int32_t amp_gain;
/* Amplifier gain in L32Q15 format.
* This is the RMS voltage at the
* loudspeaker when a 0dBFS tone
* is played in the digital domain.
*/
int16_t excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN];
/* Array of the excursion transfer function.
* The peak excursion of the
* loudspeaker diaphragm is
* measured in millimeters for 1 Vrms Sine
* tone at all FFT bin frequencies.
* Supported values: Q15 format
*/
} __packed;
#define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER 0x000100E0
/* Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER
* command, which registers a real-time port driver
* with the AFE service.
*/
struct afe_service_cmd_register_rt_port_driver {
struct apr_hdr hdr;
u16 port_id;
/* Port ID with which the real-time driver exchanges data
* (registers for events).
* Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
* #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
#define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER 0x000100E1
/* Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER
* command, which unregisters a real-time port driver from
* the AFE service.
*/
struct afe_service_cmd_unregister_rt_port_driver {
struct apr_hdr hdr;
u16 port_id;
/* Port ID from which the real-time
* driver unregisters for events.
* Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
* #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105
#define AFE_EVENTYPE_RT_PROXY_PORT_START 0
#define AFE_EVENTYPE_RT_PROXY_PORT_STOP 1
#define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK 2
#define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK 3
#define AFE_EVENTYPE_RT_PROXY_PORT_INVALID 0xFFFF
/* Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS
* message, which sends an event from the AFE service
* to a registered client.
*/
struct afe_event_rt_proxy_port_status {
u16 port_id;
/* Port ID to which the event is sent.
* Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
* #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
*/
u16 eventype;
/* Type of event.
* Supported values:
* - #AFE_EVENTYPE_RT_PROXY_PORT_START
* - #AFE_EVENTYPE_RT_PROXY_PORT_STOP
* - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK
* - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK
*/
} __packed;
#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED
struct afe_port_data_cmd_rt_proxy_port_write_v2 {
struct apr_hdr hdr;
u16 port_id;
/* Tx (mic) proxy port ID with which the real-time
* driver exchanges data.
* Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
* #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
*/
u16 reserved;
/* This field must be set to zero. */
u32 buffer_address_lsw;
/* LSW Address of the buffer containing the
* data from the real-time source
* device on a client.
*/
u32 buffer_address_msw;
/* MSW Address of the buffer containing the
* data from the real-time source
* device on a client.
*/
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory
* attributes is returned if
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS
* command is successful.
* Supported Values:
* - Any 32 bit value
*/
u32 available_bytes;
/* Number of valid bytes available
* in the buffer (including all
* channels: number of bytes per
* channel = availableBytesumChannels).
* Supported values: > 0
*
* This field must be equal to the frame
* size specified in the #AFE_PORT_AUDIO_IF_CONFIG
* command that was sent to configure this
* port.
*/
} __packed;
#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 0x000100EE
/* Payload of the
* #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which
* delivers an empty buffer to the AFE service. On
* acknowledgment, data is filled in the buffer.
*/
struct afe_port_data_cmd_rt_proxy_port_read_v2 {
struct apr_hdr hdr;
u16 port_id;
/* Rx proxy port ID with which the real-time
* driver exchanges data.
* Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
* #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
* (This must be an Rx (speaker) port.)
*/
u16 reserved;
/* This field must be set to zero. */
u32 buffer_address_lsw;
/* LSW Address of the buffer containing the data sent from the AFE
* service to a real-time sink device on the client.
*/
u32 buffer_address_msw;
/* MSW Address of the buffer containing the data sent from the AFE
* service to a real-time sink device on the client.
*/
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory attributes is
* returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
* successful.
* Supported Values:
* - Any 32 bit value
*/
u32 available_bytes;
/* Number of valid bytes available in the buffer (including all
* channels).
* Supported values: > 0
* This field must be equal to the frame size specified in the
* #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure
* this port.
*/
} __packed;
/* This module ID is related to device configuring like I2S,PCM,
* HDMI, SLIMBus etc. This module supports follwing parameter ids.
* - #AFE_PARAM_ID_I2S_CONFIG
* - #AFE_PARAM_ID_PCM_CONFIG
* - #AFE_PARAM_ID_DIGI_MIC_CONFIG
* - #AFE_PARAM_ID_HDMI_CONFIG
* - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
* - #AFE_PARAM_ID_SLIMBUS_CONFIG
* - #AFE_PARAM_ID_RT_PROXY_CONFIG
*/
#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
#define AFE_PORT_SAMPLE_RATE_8K 8000
#define AFE_PORT_SAMPLE_RATE_16K 16000
#define AFE_PORT_SAMPLE_RATE_48K 48000
#define AFE_PORT_SAMPLE_RATE_96K 96000
#define AFE_PORT_SAMPLE_RATE_192K 192000
#define AFE_LINEAR_PCM_DATA 0x0
#define AFE_NON_LINEAR_DATA 0x1
#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2
#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3
/* This param id is used to configure I2S interface */
#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
#define AFE_API_VERSION_I2S_CONFIG 0x1
/* Enumeration for setting the I2S configuration
* channel_mode parameter to
* serial data wire number 1-3 (SD3).
*/
#define AFE_PORT_I2S_SD0 0x1
#define AFE_PORT_I2S_SD1 0x2
#define AFE_PORT_I2S_SD2 0x3
#define AFE_PORT_I2S_SD3 0x4
#define AFE_PORT_I2S_QUAD01 0x5
#define AFE_PORT_I2S_QUAD23 0x6
#define AFE_PORT_I2S_6CHS 0x7
#define AFE_PORT_I2S_8CHS 0x8
#define AFE_PORT_I2S_MONO 0x0
#define AFE_PORT_I2S_STEREO 0x1
#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
/* Payload of the #AFE_PARAM_ID_I2S_CONFIG
* command's (I2S configuration
* parameter).
*/
struct afe_param_id_i2s_cfg {
u32 i2s_cfg_minor_version;
/* Minor version used for tracking the version of the I2S
* configuration interface.
* Supported values: #AFE_API_VERSION_I2S_CONFIG
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 channel_mode;
/* I2S lines and multichannel operation.
* Supported values:
* - #AFE_PORT_I2S_SD0
* - #AFE_PORT_I2S_SD1
* - #AFE_PORT_I2S_SD2
* - #AFE_PORT_I2S_SD3
* - #AFE_PORT_I2S_QUAD01
* - #AFE_PORT_I2S_QUAD23
* - #AFE_PORT_I2S_6CHS
* - #AFE_PORT_I2S_8CHS
*/
u16 mono_stereo;
/* Specifies mono or stereo. This applies only when
* a single I2S line is used.
* Supported values:
* - #AFE_PORT_I2S_MONO
* - #AFE_PORT_I2S_STEREO
*/
u16 ws_src;
/* Word select source: internal or external.
* Supported values:
* - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL
* - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
u16 data_format;
/* data format
* Supported values:
* - #LINEAR_PCM_DATA
* - #NON_LINEAR_DATA
* - #LINEAR_PCM_DATA_PACKED_IN_60958
* - #NON_LINEAR_DATA_PACKED_IN_60958
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
/*
* This param id is used to configure PCM interface
*/
#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E
#define AFE_API_VERSION_PCM_CONFIG 0x1
/* Enumeration for the auxiliary PCM synchronization signal
* provided by an external source.
*/
#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0
/* Enumeration for the auxiliary PCM synchronization signal
* provided by an internal source.
*/
#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1
/* Enumeration for the PCM configuration aux_mode parameter,
* which configures the auxiliary PCM interface to use
* short synchronization.
*/
#define AFE_PORT_PCM_AUX_MODE_PCM 0x0
/*
* Enumeration for the PCM configuration aux_mode parameter,
* which configures the auxiliary PCM interface to use long
* synchronization.
*/
#define AFE_PORT_PCM_AUX_MODE_AUX 0x1
/*
* Enumeration for setting the PCM configuration frame to 8.
*/
#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0
/*
* Enumeration for setting the PCM configuration frame to 16.
*/
#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1
/* Enumeration for setting the PCM configuration frame to 32.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2
/* Enumeration for setting the PCM configuration frame to 64.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3
/* Enumeration for setting the PCM configuration frame to 128.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4
/* Enumeration for setting the PCM configuration frame to 256.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5
/* Enumeration for setting the PCM configuration
* quantype parameter to A-law with no padding.
*/
#define AFE_PORT_PCM_ALAW_NOPADDING 0x0
/* Enumeration for setting the PCM configuration quantype
* parameter to mu-law with no padding.
*/
#define AFE_PORT_PCM_MULAW_NOPADDING 0x1
/* Enumeration for setting the PCM configuration quantype
* parameter to linear with no padding.
*/
#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2
/* Enumeration for setting the PCM configuration quantype
* parameter to A-law with padding.
*/
#define AFE_PORT_PCM_ALAW_PADDING 0x3
/* Enumeration for setting the PCM configuration quantype
* parameter to mu-law with padding.
*/
#define AFE_PORT_PCM_MULAW_PADDING 0x4
/* Enumeration for setting the PCM configuration quantype
* parameter to linear with padding.
*/
#define AFE_PORT_PCM_LINEAR_PADDING 0x5
/* Enumeration for disabling the PCM configuration
* ctrl_data_out_enable parameter.
* The PCM block is the only master.
*/
#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0
/*
* Enumeration for enabling the PCM configuration
* ctrl_data_out_enable parameter. The PCM block shares
* the signal with other masters.
*/
#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1
/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
* (PCM configuration parameter).
*/
struct afe_param_id_pcm_cfg {
u32 pcm_cfg_minor_version;
/* Minor version used for tracking the version of the AUX PCM
* configuration interface.
* Supported values: #AFE_API_VERSION_PCM_CONFIG
*/
u16 aux_mode;
/* PCM synchronization setting.
* Supported values:
* - #AFE_PORT_PCM_AUX_MODE_PCM
* - #AFE_PORT_PCM_AUX_MODE_AUX
*/
u16 sync_src;
/* Synchronization source.
* Supported values:
* - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL
* - #AFE_PORT_PCM_SYNC_SRC_INTERNAL
*/
u16 frame_setting;
/* Number of bits per frame.
* Supported values:
* - #AFE_PORT_PCM_BITS_PER_FRAME_8
* - #AFE_PORT_PCM_BITS_PER_FRAME_16
* - #AFE_PORT_PCM_BITS_PER_FRAME_32
* - #AFE_PORT_PCM_BITS_PER_FRAME_64
* - #AFE_PORT_PCM_BITS_PER_FRAME_128
* - #AFE_PORT_PCM_BITS_PER_FRAME_256
*/
u16 quantype;
/* PCM quantization type.
* Supported values:
* - #AFE_PORT_PCM_ALAW_NOPADDING
* - #AFE_PORT_PCM_MULAW_NOPADDING
* - #AFE_PORT_PCM_LINEAR_NOPADDING
* - #AFE_PORT_PCM_ALAW_PADDING
* - #AFE_PORT_PCM_MULAW_PADDING
* - #AFE_PORT_PCM_LINEAR_PADDING
*/
u16 ctrl_data_out_enable;
/* Specifies whether the PCM block shares the data-out
* signal to the drive with other masters.
* Supported values:
* - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE
* - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE
*/
u16 reserved;
/* This field must be set to zero. */
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to 4
*/
u16 slot_number_mapping[4];
/* Specifies the slot number for the each channel in
* multi channel scenario.
* Supported values: 1 to 32
*/
} __packed;
/*
* This param id is used to configure DIGI MIC interface
*/
#define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to left 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1
/*Enumeration for setting the digital mic configuration
* channel_mode parameter to right 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to left 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to right 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to stereo 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to stereo 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to quad.
*/
#define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7
/* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's
* (DIGI MIC configuration
* parameter).
*/
struct afe_param_id_digi_mic_cfg {
u32 digi_mic_cfg_minor_version;
/* Minor version used for tracking the version of the DIGI Mic
* configuration interface.
* Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 channel_mode;
/* Digital mic and multichannel operation.
* Supported values:
* - #AFE_PORT_DIGI_MIC_MODE_LEFT0
* - #AFE_PORT_DIGI_MIC_MODE_RIGHT0
* - #AFE_PORT_DIGI_MIC_MODE_LEFT1
* - #AFE_PORT_DIGI_MIC_MODE_RIGHT1
* - #AFE_PORT_DIGI_MIC_MODE_STEREO0
* - #AFE_PORT_DIGI_MIC_MODE_STEREO1
* - #AFE_PORT_DIGI_MIC_MODE_QUAD
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
*/
} __packed;
/*
* This param id is used to configure HDMI interface
*/
#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_HDMI_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command,
* which configures a multichannel HDMI audio interface.
*/
struct afe_param_id_hdmi_multi_chan_audio_cfg {
u32 hdmi_cfg_minor_version;
/* Minor version used for tracking the version of the HDMI
* configuration interface.
* Supported values: #AFE_API_VERSION_HDMI_CONFIG
*/
u16 datatype;
/* data type
* Supported values:
* - #LINEAR_PCM_DATA
* - #NON_LINEAR_DATA
* - #LINEAR_PCM_DATA_PACKED_IN_60958
* - #NON_LINEAR_DATA_PACKED_IN_60958
*/
u16 channel_allocation;
/* HDMI channel allocation information for programming an HDMI
* frame. The default is 0 (Stereo).
*
* This information is defined in the HDMI standard, CEA 861-D
* (refer to @xhyperref{S1,[S1]}). The number of channels is also
* inferred from this parameter.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - 22050, 44100, 176400 for compressed streams
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
/*
* This param id is used to configure BT or FM(RIVA) interface
*/
#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
* command's BT voice/BT audio/FM configuration parameter.
*/
struct afe_param_id_internal_bt_fm_cfg {
u32 bt_fm_cfg_minor_version;
/* Minor version used for tracking the version of the BT and FM
* configuration interface.
* Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to 2
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO)
* - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO)
* - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP)
*/
} __packed;
/* This param id is used to configure SLIMBUS interface using
* shared channel approach.
*/
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
/* Enumeration for setting SLIMbus device ID 1.
*/
#define AFE_SLIMBUS_DEVICE_1 0x0
/* Enumeration for setting SLIMbus device ID 2.
*/
#define AFE_SLIMBUS_DEVICE_2 0x1
/* Enumeration for setting the SLIMbus data formats.
*/
#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0
/* Enumeration for setting the maximum number of streams per
* device.
*/
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus
* port configuration parameter.
*/
struct afe_param_id_slimbus_cfg {
u32 sb_cfg_minor_version;
/* Minor version used for tracking the version of the SLIMBUS
* configuration interface.
* Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
*/
u16 slimbus_dev_id;
/* SLIMbus hardware device ID, which is required to handle
* multiple SLIMbus hardware blocks.
* Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 data_format;
/* Data format supported by the SLIMbus hardware. The default is
* 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
* hardware does not perform any format conversions before the data
* transfer.
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
/* Mapping of shared channel IDs (128 to 255) to which the
* master port is to be connected.
* Shared_channel_mapping[i] represents the shared channel assigned
* for audio channel i in multichannel audio data.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
} __packed;
/*
* This param id is used to configure Real Time Proxy interface.
*/
#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG
* command (real-time proxy port configuration parameter).
*/
struct afe_param_id_rt_proxy_port_cfg {
u32 rt_proxy_cfg_minor_version;
/* Minor version used for tracking the version of rt-proxy
* config interface.
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 interleaved;
/* Specifies whether the data exchanged between the AFE
* interface and real-time port is interleaved.
* Supported values: - 0 -- Non-interleaved (samples from each
* channel are contiguous in the buffer) - 1 -- Interleaved
* (corresponding samples from each input channel are interleaved
* within the buffer)
*/
u16 frame_size;
/* Size of the frames that are used for PCM exchanges with this
* port.
* Supported values: > 0, in bytes
* For example, 5 ms buffers of 16 bits and 16 kHz stereo samples
* is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320
* bytes.
*/
u16 jitter_allowance;
/* Configures the amount of jitter that the port will allow.
* Supported values: > 0
* For example, if +/-10 ms of jitter is anticipated in the timing
* of sending frames to the port, and the configuration is 16 kHz
* mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2
* bytes/sample = 320.
*/
u16 low_water_mark;
/* Low watermark in bytes (including all channels).
* Supported values:
* - 0 -- Do not send any low watermark events
* - > 0 -- Low watermark for triggering an event
* If the number of bytes in an internal circular buffer is lower
* than this low_water_mark parameter, a LOW_WATER_MARK event is
* sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
* event).
* Use of watermark events is optional for debugging purposes.
*/
u16 high_water_mark;
/* High watermark in bytes (including all channels).
* Supported values:
* - 0 -- Do not send any high watermark events
* - > 0 -- High watermark for triggering an event
* If the number of bytes in an internal circular buffer exceeds
* TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event
* is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
* event).
* The use of watermark events is optional and for debugging
* purposes.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
u16 reserved;
/* For 32 bit alignment. */
} __packed;
union afe_port_config {
struct afe_param_id_pcm_cfg pcm;
struct afe_param_id_i2s_cfg i2s;
struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
struct afe_param_id_slimbus_cfg slim_sch;
struct afe_param_id_rt_proxy_port_cfg rtproxy;
struct afe_param_id_internal_bt_fm_cfg int_bt_fm;
} __packed;
struct afe_audioif_config_command_no_payload {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
} __packed;
struct afe_audioif_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
union afe_port_config port;
} __packed;
#define AFE_PORT_CMD_DEVICE_START 0x000100E5
/* Payload of the #AFE_PORT_CMD_DEVICE_START.*/
struct afe_port_cmd_device_start {
struct apr_hdr hdr;
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. An even
* number represents the Rx direction, and an odd number represents
* the Tx direction.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.*/
} __packed;
#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
/* Payload of the #AFE_PORT_CMD_DEVICE_STOP.
*/
struct afe_port_cmd_device_stop {
struct apr_hdr hdr;
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. An even
* number represents the Rx direction, and an odd number represents
* the Tx direction.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.*/
} __packed;
#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA
/* Memory map regions command payload used by the
* #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS .
* This structure allows clients to map multiple shared memory
* regions in a single command. Following this structure are
* num_regions of afe_service_shared_map_region_payload.
*/
struct afe_service_cmd_shared_mem_map_regions {
struct apr_hdr hdr;
u16 mem_pool_id;
/* Type of memory on which this memory region is mapped.
* Supported values:
* - #ADSP_MEMORY_MAP_EBI_POOL
* - #ADSP_MEMORY_MAP_SMI_POOL
* - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL
* - Other values are reserved
*
* The memory pool ID implicitly defines the characteristics of the
* memory. Characteristics may include alignment type, permissions,
* etc.
*
* ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory
* ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory
* ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte
* addressable, and 4 KB aligned.
*/
u16 num_regions;
/* Number of regions to map.
* Supported values:
* - Any value greater than zero
*/
u32 property_flag;
/* Configures one common property for all the regions in the
* payload.
*
* Supported values: - 0x00000000 to 0x00000001
*
* b0 - bit 0 indicates physical or virtual mapping 0 Shared memory
* address provided in afe_service_shared_map_region_payloadis a
* physical address. The shared memory needs to be mapped( hardware
* TLB entry) and a software entry needs to be added for internal
* book keeping.
*
* 1 Shared memory address provided in
* afe_service_shared_map_region_payloadis a virtual address. The
* shared memory must not be mapped (since hardware TLB entry is
* already available) but a software entry needs to be added for
* internal book keeping. This can be useful if two services with in
* ADSP is communicating via APR. They can now directly communicate
* via the Virtual address instead of Physical address. The virtual
* regions must be contiguous. num_regions must be 1 in this case.
*
* b31-b1 - reserved bits. must be set to zero
*/
} __packed;
/* Map region payload used by the
* afe_service_shared_map_region_payloadstructure.
*/
struct afe_service_shared_map_region_payload {
u32 shm_addr_lsw;
/* least significant word of starting address in the memory
* region to map. It must be contiguous memory, and it must be 4 KB
* aligned.
* Supported values: - Any 32 bit value
*/
u32 shm_addr_msw;
/* most significant word of startng address in the memory region
* to map. For 32 bit shared memory address, this field must be set
* to zero. For 36 bit shared memory address, bit31 to bit 4 must be
* set to zero
*
* Supported values: - For 32 bit shared memory address, this field
* must be set to zero. - For 36 bit shared memory address, bit31 to
* bit 4 must be set to zero - For 64 bit shared memory address, any
* 32 bit value
*/
u32 mem_size_bytes;
/* Number of bytes in the region. The aDSP will always map the
* regions as virtual contiguous memory, but the memory size must be
* in multiples of 4 KB to avoid gaps in the virtually contiguous
* mapped memory.
*
* Supported values: - multiples of 4KB
*/
} __packed;
#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB
struct afe_service_cmdrsp_shared_mem_map_regions {
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory attributes is
* returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
* successful. In the case of failure , a generic APR error response
* is returned to the client.
*
* Supported Values: - Any 32 bit value
*/
} __packed;
#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC
/* Memory unmap regions command payload used by the
* #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS
*
* This structure allows clients to unmap multiple shared memory
* regions in a single command.
*/
struct afe_service_cmd_shared_mem_unmap_regions {
struct apr_hdr hdr;
u32 mem_map_handle;
/* memory map handle returned by
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands
*
* Supported Values:
* - Any 32 bit value
*/
} __packed;
#define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0
/* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command,
* which queries for one post/preprocessing parameter of a
* stream.
*/
struct afe_port_cmd_get_param_v2 {
struct apr_hdr hdr;
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. */
u16 payload_size;
/* Maximum data size of the parameter ID/module ID combination.
* This is a multiple of four bytes
* Supported values: > 0
*/
u32 payload_address_lsw;
/* LSW of 64 bit Payload address. Address should be 32-byte,
* 4kbyte aligned and must be contig memory.
*/
u32 payload_address_msw;
/* MSW of 64 bit Payload address. In case of 32-bit shared
* memory address, this field must be set to zero. In case of 36-bit
* shared memory address, bit-4 to bit-31 must be set to zero.
* Address should be 32-byte, 4kbyte aligned and must be contiguous
* memory.
*/
u32 mem_map_handle;
/* Memory map handle returned by
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
* Supported Values: - NULL -- Message. The parameter data is
* in-band. - Non-NULL -- The parameter data is Out-band.Pointer to
* - the physical address in shared memory of the payload data.
* For detailed payload content, see the afe_port_param_data_v2
* structure
*/
u32 module_id;
/* ID of the module to be queried.
* Supported values: Valid module ID
*/
u32 param_id;
/* ID of the parameter to be queried.
* Supported values: Valid parameter ID
*/
} __packed;
#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which
* responds to an #AFE_PORT_CMD_GET_PARAM_V2 command.
*
* Immediately following this structure is the parameters structure
* (afe_port_param_data) containing the response(acknowledgment)
* parameter payload. This payload is included for an in-band
* scenario. For an address/shared memory-based set parameter, this
* payload is not needed.
*/
struct afe_port_cmdrsp_get_param_v2 {
u32 status;
} __packed;
/* adsp_afe_service_commands.h */
#define ADSP_MEMORY_MAP_EBI_POOL 0
#define ADSP_MEMORY_MAP_SMI_POOL 1
#define ADSP_MEMORY_MAP_IMEM_POOL 2
#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
/*
* Definition of virtual memory flag
*/
#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1
/*
* Definition of physical memory flag
*/
#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0
#define DEFAULT_COPP_TOPOLOGY 0x00010be3
#define DEFAULT_POPP_TOPOLOGY 0x00010be4
#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75
/* Memory map regions command payload used by the
* #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS
* commands.
*
* This structure allows clients to map multiple shared memory
* regions in a single command. Following this structure are
* num_regions of avs_shared_map_region_payload.
*/
struct avs_cmd_shared_mem_map_regions {
struct apr_hdr hdr;
u16 mem_pool_id;
/* Type of memory on which this memory region is mapped.
*
* Supported values: - #ADSP_MEMORY_MAP_EBI_POOL -
* #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL
* (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values
* are reserved
*
* The memory ID implicitly defines the characteristics of the
* memory. Characteristics may include alignment type, permissions,
* etc.
*
* SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned.
*/
u16 num_regions;
/* Number of regions to map.*/
u32 property_flag;
/* Configures one common property for all the regions in the
* payload. No two regions in the same memory map regions cmd can
* have differnt property. Supported values: - 0x00000000 to
* 0x00000001
*
* b0 - bit 0 indicates physical or virtual mapping 0 shared memory
* address provided in avs_shared_map_regions_payload is physical
* address. The shared memory needs to be mapped( hardware TLB
* entry)
*
* and a software entry needs to be added for internal book keeping.
*
* 1 Shared memory address provided in MayPayload[usRegions] is
* virtual address. The shared memory must not be mapped (since
* hardware TLB entry is already available) but a software entry
* needs to be added for internal book keeping. This can be useful
* if two services with in ADSP is communicating via APR. They can
* now directly communicate via the Virtual address instead of
* Physical address. The virtual regions must be contiguous.
*
* b31-b1 - reserved bits. must be set to zero
*/
} __packed;
struct avs_shared_map_region_payload {
u32 shm_addr_lsw;
/* least significant word of shared memory address of the memory
* region to map. It must be contiguous memory, and it must be 4 KB
* aligned.
*/
u32 shm_addr_msw;
/* most significant word of shared memory address of the memory
* region to map. For 32 bit shared memory address, this field must
* tbe set to zero. For 36 bit shared memory address, bit31 to bit 4
* must be set to zero
*/
u32 mem_size_bytes;
/* Number of bytes in the region.
*
* The aDSP will always map the regions as virtual contiguous
* memory, but the memory size must be in multiples of 4 KB to avoid
* gaps in the virtually contiguous mapped memory.
*/
} __packed;
struct avs_cmd_shared_mem_unmap_regions {
struct apr_hdr hdr;
u32 mem_map_handle;
/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS
* , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands
*/
} __packed;
/* Memory map command response payload used by the
* #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS
* ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS
*/
struct avs_cmdrsp_shared_mem_map_regions {
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory attributes is
* returned
*/
} __packed;
/*adsp_audio_memmap_api.h*/
/* ASM related data structures */
struct asm_wma_cfg {
u16 format_tag;
u16 ch_cfg;
u32 sample_rate;
u32 avg_bytes_per_sec;
u16 block_align;
u16 valid_bits_per_sample;
u32 ch_mask;
u16 encode_opt;
u16 adv_encode_opt;
u32 adv_encode_opt2;
u32 drc_peak_ref;
u32 drc_peak_target;
u32 drc_ave_ref;
u32 drc_ave_target;
} __packed;
struct asm_wmapro_cfg {
u16 format_tag;
u16 ch_cfg;
u32 sample_rate;
u32 avg_bytes_per_sec;
u16 block_align;
u16 valid_bits_per_sample;
u32 ch_mask;
u16 encode_opt;
u16 adv_encode_opt;
u32 adv_encode_opt2;
u32 drc_peak_ref;
u32 drc_peak_target;
u32 drc_ave_ref;
u32 drc_ave_target;
} __packed;
struct asm_aac_cfg {
u16 format;
u16 aot;
u16 ep_config;
u16 section_data_resilience;
u16 scalefactor_data_resilience;
u16 spectral_data_resilience;
u16 ch_cfg;
u16 reserved;
u32 sample_rate;
} __packed;
struct asm_softpause_params {
u32 enable;
u32 period;
u32 step;
u32 rampingcurve;
} __packed;
struct asm_softvolume_params {
u32 period;
u32 step;
u32 rampingcurve;
} __packed;
#define ASM_END_POINT_DEVICE_MATRIX 0
#define PCM_CHANNEL_NULL 0
/* Front left channel. */
#define PCM_CHANNEL_FL 1
/* Front right channel. */
#define PCM_CHANNEL_FR 2
/* Front center channel. */
#define PCM_CHANNEL_FC 3
/* Left surround channel.*/
#define PCM_CHANNEL_LS 4
/* Right surround channel.*/
#define PCM_CHANNEL_RS 5
/* Low frequency effect channel. */
#define PCM_CHANNEL_LFE 6
/* Center surround channel; Rear center channel. */
#define PCM_CHANNEL_CS 7
/* Left back channel; Rear left channel. */
#define PCM_CHANNEL_LB 8
/* Right back channel; Rear right channel. */
#define PCM_CHANNEL_RB 9
/* Top surround channel. */
#define PCM_CHANNELS 10
/* Center vertical height channel.*/
#define PCM_CHANNEL_CVH 11
/* Mono surround channel.*/
#define PCM_CHANNEL_MS 12
/* Front left of center. */
#define PCM_CHANNEL_FLC 13
/* Front right of center. */
#define PCM_CHANNEL_FRC 14
/* Rear left of center. */
#define PCM_CHANNEL_RLC 15
/* Rear right of center. */
#define PCM_CHANNEL_RRC 16
#define PCM_FORMAT_MAX_NUM_CHANNEL 8
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT 0x00010BE4
#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
#define ASM_MAX_EQ_BANDS 12
#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
struct asm_data_cmd_media_fmt_update_v2 {
u32 fmt_blk_size;
/* Media format block size in bytes.*/
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 num_channels;
/* Number of channels. Supported values: 1 to 8 */
u16 bits_per_sample;
/* Number of bits per sample per channel. * Supported values:
* 16, 24 * When used for playback, the client must send 24-bit
* samples packed in 32-bit words. The 24-bit samples must be placed
* in the most significant 24 bits of the 32-bit word. When used for
* recording, the aDSP sends 24-bit samples packed in 32-bit words.
* The 24-bit samples are placed in the most significant 24 bits of
* the 32-bit word.
*/
u32 sample_rate;
/* Number of samples per second (in Hertz).
* Supported values: 2000 to 48000
*/
u16 is_signed;
/* Flag that indicates the samples are signed (1). */
u16 reserved;
/* reserved field for 32 bit alignment. must be set to zero. */
u8 channel_mapping[8];
/* Channel array of size 8.
* Supported values:
* - #PCM_CHANNEL_L
* - #PCM_CHANNEL_R
* - #PCM_CHANNEL_C
* - #PCM_CHANNEL_LS
* - #PCM_CHANNEL_RS
* - #PCM_CHANNEL_LFE
* - #PCM_CHANNEL_CS
* - #PCM_CHANNEL_LB
* - #PCM_CHANNEL_RB
* - #PCM_CHANNELS
* - #PCM_CHANNEL_CVH
* - #PCM_CHANNEL_MS
* - #PCM_CHANNEL_FLC
* - #PCM_CHANNEL_FRC
* - #PCM_CHANNEL_RLC
* - #PCM_CHANNEL_RRC
*
* Channel[i] mapping describes channel I. Each element i of the
* array describes channel I inside the buffer where 0 @le I <
* num_channels. An unused channel is set to zero.
*/
} __packed;
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
/* ID of the parameter. */
u32 param_size;
/* Data size of this parameter, in bytes. The size is a multiple
* of 4 bytes.
*/
} __packed;
struct asm_enc_cfg_blk_param_v2 {
u32 frames_per_buf;
/* Number of encoded frames to pack into each buffer.
*
* @note1hang This is only guidance information for the aDSP. The
* number of encoded frames put into each buffer (specified by the
* client) is less than or equal to this number.
*/
u32 enc_cfg_blk_size;
/* Size in bytes of the encoder configuration block that follows
* this member.
*/
} __packed;
/* @brief Multichannel PCM encoder configuration structure used
* in the #ASM_STREAM_CMD_OPEN_READ_V2 command.
*/
struct asm_multi_channel_pcm_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint16_t num_channels;
/*< Number of PCM channels.
*
* Supported values: - 0 -- Native mode - 1 -- 8 Native mode
* indicates that encoding must be performed with the number of
* channels at the input.
*/
uint16_t bits_per_sample;
/*< Number of bits per sample per channel.
* Supported values: 16, 24
*/
uint32_t sample_rate;
/*< Number of samples per second (in Hertz).
*
* Supported values: 0, 8000 to 48000 A value of 0 indicates the
* native sampling rate. Encoding is performed at the input sampling
* rate.
*/
uint16_t is_signed;
/*< Specifies whether the samples are signed (1). Currently,
* only signed samples are supported.
*/
uint16_t reserved;
/*< reserved field for 32 bit alignment. must be set to zero.*/
uint8_t channel_mapping[8];
} __packed;
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6
/* @xreflabel
* {hdr:AsmMediaFmtDolbyAac} Media format ID for the
* Dolby AAC decoder. This format ID is be used if the client wants
* to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC
* contents.
*/
#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86
/* Enumeration for the audio data transport stream AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
/* Enumeration for low overhead audio stream AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS 1
/* Enumeration for the audio data interchange format
* AAC format.
*/
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF 2
/* Enumeration for the raw AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
#define ASM_MEDIA_FMT_AAC_AOT_LC 2
#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
#define ASM_MEDIA_FMT_AAC_AOT_PS 29
#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22
struct asm_aac_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 aac_fmt_flag;
/* Bitstream format option.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
u16 audio_objype;
/* Audio Object Type (AOT) present in the AAC stream.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_AOT_LC
* - #ASM_MEDIA_FMT_AAC_AOT_SBR
* - #ASM_MEDIA_FMT_AAC_AOT_BSAC
* - #ASM_MEDIA_FMT_AAC_AOT_PS
* - Otherwise -- Not supported
*/
u16 channel_config;
/* Number of channels present in the AAC stream.
* Supported values:
* - 1 -- Mono
* - 2 -- Stereo
* - 6 -- 5.1 content
*/
u16 total_size_of_PCE_bits;
/* greater or equal to zero. * -In case of RAW formats and
* channel config = 0 (PCE), client can send * the bit stream
* containing PCE immediately following this structure * (in-band).
* -This number does not include bits included for 32 bit alignment.
* -If zero, then the PCE info is assumed to be available in the
* audio -bit stream & not in-band.
*/
u32 sample_rate;
/* Number of samples per second (in Hertz).
*
* Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000,
* 44100, 48000
*
* This field must be equal to the sample rate of the AAC-LC
* decoder's output. - For MP4 or 3GP containers, this is indicated
* by the samplingFrequencyIndex field in the AudioSpecificConfig
* element. - For ADTS format, this is indicated by the
* samplingFrequencyIndex in the ADTS fixed header. - For ADIF
* format, this is indicated by the samplingFrequencyIndex in the
* program_config_element present in the ADIF header.
*/
} __packed;
struct asm_aac_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 bit_rate;
/* Encoding rate in bits per second. */
u32 enc_mode;
/* Encoding mode.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_AOT_LC
* - #ASM_MEDIA_FMT_AAC_AOT_SBR
* - #ASM_MEDIA_FMT_AAC_AOT_PS
*/
u16 aac_fmt_flag;
/* AAC format flag.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
u16 channel_cfg;
/* Number of channels to encode.
* Supported values:
* - 0 -- Native mode
* - 1 -- Mono
* - 2 -- Stereo
* - Other values are not supported.
* @note1hang The eAAC+ encoder mode supports only stereo.
* Native mode indicates that encoding must be performed with the
* number of channels at the input.
* The number of channels must not change during encoding.
*/
u32 sample_rate;
/* Number of samples per second.
* Supported values: - 0 -- Native mode - For other values,
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
} __packed;
#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_STREAM_CMD_OPEN_READ_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_DEC 0x00010BF6
#define ASM_MEDIA_FMT_EAC3_DEC 0x00010C3C
#define ASM_MEDIA_FMT_DTS 0x00010D88
/* Media format ID for adaptive transform acoustic coding. This
* ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED command
* only.
*/
#define ASM_MEDIA_FMT_ATRAC 0x00010D89
/* Media format ID for metadata-enhanced audio transmission.
* This ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
* command only.
*/
#define ASM_MEDIA_FMT_MAT 0x00010D8A
/* adsp_media_fmt.h */
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
struct asm_data_cmd_write_v2 {
struct apr_hdr hdr;
u32 buf_addr_lsw;
/* The 64 bit address msw-lsw should be a valid, mapped address.
* 64 bit address should be a multiple of 32 bytes
*/
u32 buf_addr_msw;
/* The 64 bit address msw-lsw should be a valid, mapped address.
* 64 bit address should be a multiple of 32 bytes.
* -Address of the buffer containing the data to be decoded.
* The buffer should be aligned to a 32 byte boundary.
* -In the case of 32 bit Shared memory address, msw field must
* -be set to zero.
* -In the case of 36 bit shared memory address, bit 31 to bit 4
* -of msw must be set to zero.
*/
u32 mem_map_handle;
/* memory map handle returned by DSP through
* ASM_CMD_SHARED_MEM_MAP_REGIONS command
*/
u32 buf_size;
/* Number of valid bytes available in the buffer for decoding. The
* first byte starts at buf_addr.
*/
u32 seq_id;
/* Optional buffer sequence ID. */
u32 timestamp_lsw;
/* Lower 32 bits of the 64-bit session time in microseconds of the
* first buffer sample.
*/
u32 timestamp_msw;
/* Upper 32 bits of the 64-bit session time in microseconds of the
* first buffer sample.
*/
u32 flags;
/* Bitfield of flags.
* Supported values for bit 31:
* - 1 -- Valid timestamp.
* - 0 -- Invalid timestamp.
* - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG as the bitmask and
* #ASM_SHIFTIMESTAMP_VALID_FLAG as the shift value to set this bit.
* Supported values for bit 30:
* - 1 -- Last buffer.
* - 0 -- Not the last buffer.
*
* Supported values for bit 29:
* - 1 -- Continue the timestamp from the previous buffer.
* - 0 -- Timestamp of the current buffer is not related
* to the timestamp of the previous buffer.
* - Use #ASM_BIT_MASKS_CONTINUE_FLAG and #ASM_SHIFTS_CONTINUE_FLAG
* to set this bit.
*
* Supported values for bit 4:
* - 1 -- End of the frame.
* - 0 -- Not the end of frame, or this information is not known.
* - Use #ASM_BIT_MASK_EOF_FLAG as the bitmask and #ASM_SHIFT_EOF_FLAG
* as the shift value to set this bit.
*
* All other bits are reserved and must be set to 0.
*
* If bit 31=0 and bit 29=1: The timestamp of the first sample in
* this buffer continues from the timestamp of the last sample in
* the previous buffer. If there is no previous buffer (i.e., this
* is the first buffer sent after opening the stream or after a
* flush operation), or if the previous buffer does not have a valid
* timestamp, the samples in the current buffer also do not have a
* valid timestamp. They are played out as soon as possible.
*
*
* If bit 31=0 and bit 29=0: No timestamp is associated with the
* first sample in this buffer. The samples are played out as soon
* as possible.
*
*
* If bit 31=1 and bit 29 is ignored: The timestamp specified in
* this payload is honored.
*
*
* If bit 30=0: Not the last buffer in the stream. This is useful
* in removing trailing samples.
*
*
* For bit 4: The client can set this flag for every buffer sent in
* which the last byte is the end of a frame. If this flag is set,
* the buffer can contain data from multiple frames, but it should
* always end at a frame boundary. Restrictions allow the aDSP to
* detect an end of frame without requiring additional processing.
*/
} __packed;
#define ASM_DATA_CMD_READ_V2 0x00010DAC
struct asm_data_cmd_read_v2 {
struct apr_hdr hdr;
u32 buf_addr_lsw;
/* the 64 bit address msw-lsw should be a valid mapped address
* and should be a multiple of 32 bytes
*/
u32 buf_addr_msw;
/* the 64 bit address msw-lsw should be a valid mapped address
* and should be a multiple of 32 bytes.
* - Address of the buffer where the DSP puts the encoded data,
* potentially, at an offset specified by the uOffset field in
* ASM_DATA_EVENT_READ_DONE structure. The buffer should be aligned
* to a 32 byte boundary.
*- In the case of 32 bit Shared memory address, msw field must
*- be set to zero.
*- In the case of 36 bit shared memory address, bit 31 to bit
*- 4 of msw must be set to zero.
*/
u32 mem_map_handle;
/* memory map handle returned by DSP through
* ASM_CMD_SHARED_MEM_MAP_REGIONS command.
*/
u32 buf_size;
/* Number of bytes available for the aDSP to write. The aDSP
* starts writing from buf_addr.
*/
u32 seq_id;
/* Optional buffer sequence ID.
*/
} __packed;
#define ASM_DATA_CMD_EOS 0x00010BDB
#define ASM_DATA_EVENT_RENDERED_EOS 0x00010C1C
#define ASM_DATA_EVENT_EOS 0x00010BDD
#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
struct asm_data_event_write_done_v2 {
u32 buf_addr_lsw;
/* lsw of the 64 bit address */
u32 buf_addr_msw;
/* msw of the 64 bit address. address given by the client in
* ASM_DATA_CMD_WRITE_V2 command.
*/
u32 mem_map_handle;
/* memory map handle in the ASM_DATA_CMD_WRITE_V2 */
u32 status;
/* Status message (error code) that indicates whether the
* referenced buffer has been successfully consumed.
* Supported values: Refer to @xhyperref{Q3,[Q3]}
*/
} __packed;
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
/* Definition of the frame metadata flag bitmask.*/
#define ASM_BIT_MASK_FRAME_METADATA_FLAG (0x40000000UL)
/* Definition of the frame metadata flag shift value. */
#define ASM_SHIFT_FRAME_METADATA_FLAG 30
struct asm_data_event_read_done_v2 {
u32 status;
/* Status message (error code).
* Supported values: Refer to @xhyperref{Q3,[Q3]}
*/
u32 buf_addr_lsw;
/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
* address is a multiple of 32 bytes.
*/
u32 buf_addr_msw;
/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
* address is a multiple of 32 bytes.
*
* -Same address provided by the client in ASM_DATA_CMD_READ_V2
* -In the case of 32 bit Shared memory address, msw field is set to
* zero.
* -In the case of 36 bit shared memory address, bit 31 to bit 4
* -of msw is set to zero.
*/
u32 mem_map_handle;
/* memory map handle in the ASM_DATA_CMD_READ_V2 */
u32 enc_framesotal_size;
/* Total size of the encoded frames in bytes.
* Supported values: >0
*/
u32 offset;
/* Offset (from buf_addr) to the first byte of the first encoded
* frame. All encoded frames are consecutive, starting from this
* offset.
* Supported values: > 0
*/
u32 timestamp_lsw;
/* Lower 32 bits of the 64-bit session time in microseconds of
* the first sample in the buffer. If Bit 5 of mode_flags flag of
* ASM_STREAM_CMD_OPEN_READ_V2 is 1 then the 64 bit timestamp is
* absolute capture time otherwise it is relative session time. The
* absolute timestamp doesnt reset unless the system is reset.
*/
u32 timestamp_msw;
/* Upper 32 bits of the 64-bit session time in microseconds of
* the first sample in the buffer.
*/
u32 flags;
/* Bitfield of flags. Bit 30 indicates whether frame metadata is
* present. If frame metadata is present, num_frames consecutive
* instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start
* at the buffer address.
* Supported values for bit 31:
* - 1 -- Timestamp is valid.
* - 0 -- Timestamp is invalid.
* - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and
* #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit.
*
* Supported values for bit 30:
* - 1 -- Frame metadata is present.
* - 0 -- Frame metadata is absent.
* - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and
* #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit.
*
* All other bits are reserved; the aDSP sets them to 0.
*/
u32 num_frames;
/* Number of encoded frames in the buffer. */
u32 seq_id;
/* Optional buffer sequence ID. */
} __packed;
struct asm_data_read_buf_metadata_v2 {
u32 offset;
/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to
* the frame associated with this metadata.
* Supported values: > 0
*/
u32 frm_size;
/* Size of the encoded frame in bytes.
* Supported values: > 0
*/
u32 num_encoded_pcm_samples;
/* Number of encoded PCM samples (per channel) in the frame
* associated with this metadata.
* Supported values: > 0
*/
u32 timestamp_lsw;
/* Lower 32 bits of the 64-bit session time in microseconds of the
* first sample for this frame.
* If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1
* then the 64 bit timestamp is absolute capture time otherwise it
* is relative session time. The absolute timestamp doesnt reset
* unless the system is reset.
*/
u32 timestamp_msw;
/* Lower 32 bits of the 64-bit session time in microseconds of the
* first sample for this frame.
*/
u32 flags;
/* Frame flags.
* Supported values for bit 31:
* - 1 -- Time stamp is valid
* - 0 -- Time stamp is not valid
* - All other bits are reserved; the aDSP sets them to 0.
*/
} __packed;
/* Notifies the client of a change in the data sampling rate or
* Channel mode. This event is raised by the decoder service. The
* event is enabled through the mode flags of
* #ASM_STREAM_CMD_OPEN_WRITE_V2 or
* #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
* in the output sampling frequency or the number/positioning of
* output channels, or if it is the first frame decoded.The new
* sampling frequency or the new channel configuration is
* communicated back to the client asynchronously.
*/
#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event.
* This event is raised when the following conditions are both true:
* - The event is enabled through the mode_flags of
* #ASM_STREAM_CMD_OPEN_WRITE_V2 or
* #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
* in either the output sampling frequency or the number/positioning
* of output channels, or if it is the first frame decoded.
* This event is not raised (even if enabled) if the decoder is
* MIDI, because
*/
struct asm_data_event_sr_cm_change_notify {
u32 sample_rate;
/* New sampling rate (in Hertz) after detecting a change in the
* bitstream.
* Supported values: 2000 to 48000
*/
u16 num_channels;
/* New number of channels after detecting a change in the
* bitstream.
* Supported values: 1 to 8
*/
u16 reserved;
/* Reserved for future use. This field must be set to 0.*/
u8 channel_mapping[8];
} __packed;
/* Notifies the client of a data sampling rate or channel mode
* change. This event is raised by the encoder service.
* This event is raised when :
* - Native mode encoding was requested in the encoder
* configuration (i.e., the channel number was 0), the sample rate
* was 0, or both were 0.
*
* - The input data frame at the encoder is the first one, or the
* sampling rate/channel mode is different from the previous input
* data frame.
*
*/
#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE
struct asm_data_event_enc_sr_cm_change_notify {
u32 sample_rate;
/* New sampling rate (in Hertz) after detecting a change in the
* input data.
* Supported values: 2000 to 48000
*/
u16 num_channels;
/* New number of channels after detecting a change in the input
* data. Supported values: 1 to 8
*/
u16 bits_per_sample;
/* New bits per sample after detecting a change in the input
* data.
* Supported values: 16, 24
*/
u8 channel_mapping[8];
} __packed;
#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87
/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command,
* which is used to indicate the IEC 60958 frame rate of a given
* packetized audio stream.
*/
struct asm_data_cmd_iec_60958_frame_rate {
u32 frame_rate;
/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream.
* Supported values: Any valid frame rate
*/
} __packed;
/* adsp_asm_data_commands.h*/
#define ASM_SVC_CMD_GET_STREAM_HANDLES 0x00010C0B
#define ASM_SVC_CMDRSP_GET_STREAM_HANDLES 0x00010C1B
/* 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
/* Payload of the #ASM_SVC_CMDRSP_GET_STREAM_HANDLES message,
* which returns a list of currently active stream handles.
* Immediately following this structure are num_handles of uint32
* stream handles.
*/
struct asm_svc_cmdrsp_get_stream_handles {
u32 num_handles;
/* Number of active stream handles. */
} __packed;
#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 (8)
/* 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_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_V2 0x00010D8F
struct asm_stream_cmd_open_write_v2 {
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_DEC
* - #ASM_MEDIA_FMT_EAC3_DEC
* - #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_EXAMPLE
*/
} __packed;
#define ASM_STREAM_CMD_OPEN_READ_V2 0x00010D8C
/* 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
struct asm_stream_cmd_open_read_v2 {
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
#define ASM_STREAM_CMD_OPEN_READ_V2 0x00010D8C
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_DEC
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_EXAMPLE
*/
u32 enc_cfg_id;
/* Specifies the media type for the output of the stream. PCM
* indicates that no encoding must be performed, e.g., this is an NT
* decoder session.
* Supported values:
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_AAC_V2
* - #ASM_MEDIA_FMT_AMRNB_FS
* - #ASM_MEDIA_FMT_AMRWB_FS
* - #ASM_MEDIA_FMT_V13K_FS
* - #ASM_MEDIA_FMT_EVRC_FS
* - #ASM_MEDIA_FMT_EVRCB_FS
* - #ASM_MEDIA_FMT_EVRCWB_FS
* - #ASM_MEDIA_FMT_SBC
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_EXAMPLE
* - #ASM_MEDIA_FMT_WMA_V8
*/
u16 bits_per_sample;
/* Number of bits per sample processed by ASM modules.
* Supported values: 16 and 24 bits per sample
*/
u16 reserved;
/* Reserved for future use. This field must be set to zero.*/
} __packed;
#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E
struct asm_stream_cmd_open_loopback_v2 {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags.
* Bit 0-31: reserved; client should set these bits to 0
*/
u16 src_endpointype;
/* Endpoint type. 0 = Tx Matrix */
u16 sink_endpointype;
/* Endpoint type. 0 = Rx Matrix */
u32 postprocopo_id;
/* Postprocessor topology ID. Specifies the topology of
* postprocessing algorithms.
*/
u16 bits_per_sample;
/* The number of bits per sample processed by ASM modules
* Supported values: 16 and 24 bits per sample
*/
u16 reserved;
/* Reserved for future use. This field must be set to zero. */
} __packed;
#define ASM_STREAM_CMD_CLOSE 0x00010BCD
#define ASM_STREAM_CMD_FLUSH 0x00010BCE
#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1
struct asm_stream_cmd_set_pp_params_v2 {
u32 data_payload_addr_lsw;
/* LSW of parameter data payload address. Supported values: any. */
u32 data_payload_addr_msw;
/* MSW of Parameter data payload address. Supported values: any.
* - Must be set to zero for in-band data.
* - In the case of 32 bit Shared memory address, msw field must be
* - set to zero.
* - In the case of 36 bit shared memory address, bit 31 to bit 4 of
* msw
*
* - must be set to zero.
*/
u32 mem_map_handle;
/* Supported Values: Any.
* memory map handle returned by DSP through
* ASM_CMD_SHARED_MEM_MAP_REGIONS
* command.
* if mmhandle is NULL, the ParamData payloads are within the
* message payload (in-band).
* If mmhandle is non-NULL, the ParamData payloads begin at the
* address specified in the address msw and lsw (out-of-band).
*/
u32 data_payload_size;
/* Size in bytes of the variable payload accompanying the
message, or in shared memory. This field is used for parsing the
parameter payload. */
} __packed;
struct asm_stream_param_data_v2 {
u32 module_id;
/* Unique module ID. */
u32 param_id;
/* Unique parameter ID. */
u16 param_size;
/* Data size of the param_id/module_id combination. This is
* a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for future enhancements. This field must be set to
* zero.
*/
} __packed;
#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2
struct asm_stream_cmd_get_pp_params_v2 {
u32 data_payload_addr_lsw;
/* LSW of the parameter data payload address. */
u32 data_payload_addr_msw;
/* MSW of the parameter data payload address.
* - Size of the shared memory, if specified, shall be large enough
* to contain the whole ParamData payload, including Module ID,
* Param ID, Param Size, and Param Values
* - Must be set to zero for in-band data
* - In the case of 32 bit Shared memory address, msw field must be
* set to zero.
* - In the case of 36 bit shared memory address, bit 31 to bit 4 of
* msw must be set to zero.
*/
u32 mem_map_handle;
/* Supported Values: Any.
* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS
* command.
* if mmhandle is NULL, the ParamData payloads in the ACK are within the
* message payload (in-band).
* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the
* address specified in the address msw and lsw.
* (out-of-band).
*/
u32 module_id;
/* Unique module ID. */
u32 param_id;
/* Unique parameter ID. */
u16 param_max_size;
/* Maximum data size of the module_id/param_id combination. This
* is a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for backward compatibility. Clients must set this
* field to zero.
*/
} __packed;
#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
#define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13
struct asm_bitrate_param {
u32 bitrate;
/* Maximum supported bitrate. Only the AAC encoder is supported.*/
} __packed;
#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
#define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63
/* Flag to turn off both SBR and PS processing, if they are
* present in the bitstream.
*/
#define ASM_AAC_SBR_OFF_PS_OFF (2)
/* Flag to turn on SBR but turn off PS processing,if they are
* present in the bitstream.
*/
#define ASM_AAC_SBR_ON_PS_OFF (1)
/* Flag to turn on both SBR and PS processing, if they are
* present in the bitstream (default behavior).
*/
#define ASM_AAC_SBR_ON_PS_ON (0)
/* Structure for an AAC SBR PS processing flag. */
/* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_aac_sbr_ps_flag_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 sbr_ps_flag;
/* Control parameter to enable or disable SBR/PS processing in
* the AAC bitstream. Use the following macros to set this field:
* - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS
* processing, if they are present in the bitstream.
* - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS
* processing, if they are present in the bitstream.
* - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing,
* if they are present in the bitstream (default behavior).
* - All other values are invalid.
* Changes are applied to the next decoded frame.
*/
} __packed;
#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64
/* First single channel element in a dual mono bitstream.*/
#define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1)
/* Second single channel element in a dual mono bitstream.*/
#define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2)
/* Structure for AAC decoder dual mono channel mapping. */
struct asm_aac_dual_mono_mapping_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u16 left_channel_sce;
u16 right_channel_sce;
} __packed;
#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4
struct asm_stream_cmdrsp_get_pp_params_v2 {
u32 status;
} __packed;
#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73
/* Enumeration for both vocals in a karaoke stream.*/
#define AC3_KARAOKE_MODE_NO_VOCAL (0)
/* Enumeration for only the left vocal in a karaoke stream.*/
#define AC3_KARAOKE_MODE_LEFT_VOCAL (1)
/* Enumeration for only the right vocal in a karaoke stream.*/
#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2)
/* Enumeration for both vocal channels in a karaoke stream.*/
#define AC3_KARAOKE_MODE_BOTH_VOCAL (3)
#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74
/* Enumeration for the Custom Analog mode.*/
#define AC3_DRC_MODE_CUSTOM_ANALOG (0)
/* Enumeration for the Custom Digital mode.*/
#define AC3_DRC_MODE_CUSTOM_DIGITAL (1)
/* Enumeration for the Line Out mode (light compression).*/
#define AC3_DRC_MODE_LINE_OUT (2)
/* Enumeration for the RF remodulation mode (heavy compression).*/
#define AC3_DRC_MODE_RF_REMOD (3)
#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75
/* Enumeration for playing dual mono in stereo mode.*/
#define AC3_DUAL_MONO_MODE_STEREO (0)
/* Enumeration for playing left mono.*/
#define AC3_DUAL_MONO_MODE_LEFT_MONO (1)
/* Enumeration for playing right mono.*/
#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2)
/* Enumeration for mixing both dual mono channels and playing them.*/
#define AC3_DUAL_MONO_MODE_MIXED_MONO (3)
#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76
/* Enumeration for using the Downmix mode indicated in the bitstream. */
#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0)
/* Enumeration for Surround Compatible mode (preserves the
* surround information).
*/
#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1)
/* Enumeration for Mono Compatible mode (if the output is to be
* further downmixed to mono).
*/
#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2)
/* ID of the AC3 PCM scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78
/* ID of the AC3 DRC boost scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79
/* ID of the AC3 DRC cut scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A
/* Structure for AC3 Generic Parameter. */
/* Payload of the AC3 parameters in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_ac3_generic_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 generic_parameter;
/* AC3 generic parameter. Select from one of the following
* possible values.
*
* For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are:
* - AC3_KARAOKE_MODE_NO_VOCAL
* - AC3_KARAOKE_MODE_LEFT_VOCAL
* - AC3_KARAOKE_MODE_RIGHT_VOCAL
* - AC3_KARAOKE_MODE_BOTH_VOCAL
*
* For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are:
* - AC3_DRC_MODE_CUSTOM_ANALOG
* - AC3_DRC_MODE_CUSTOM_DIGITAL
* - AC3_DRC_MODE_LINE_OUT
* - AC3_DRC_MODE_RF_REMOD
*
* For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are:
* - AC3_DUAL_MONO_MODE_STEREO
* - AC3_DUAL_MONO_MODE_LEFT_MONO
* - AC3_DUAL_MONO_MODE_RIGHT_MONO
* - AC3_DUAL_MONO_MODE_MIXED_MONO
*
* For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are:
* - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT
* - AC3_STEREO_DOWNMIX_MODE_LT_RT
* - AC3_STEREO_DOWNMIX_MODE_LO_RO
*
* For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*
* For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*
* For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*/
} __packed;
/* Enumeration for Raw mode (no downmixing), which specifies
* that all channels in the bitstream are to be played out as is
* without any downmixing. (Default)
*/
#define WMAPRO_CHANNEL_MASK_RAW (-1)
/* Enumeration for setting the channel mask to 0. The 7.1 mode
* (Home Theater) is assigned.
*/
#define WMAPRO_CHANNEL_MASK_ZERO 0x0000
/* Speaker layout mask for one channel (Home Theater, mono).
* - Speaker front center
*/
#define WMAPRO_CHANNEL_MASK_1_C 0x0004
/* Speaker layout mask for two channels (Home Theater, stereo).
* - Speaker front left
* - Speaker front right
*/
#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003
/* Speaker layout mask for three channels (Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
*/
#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007
/* Speaker layout mask for two channels (stereo).
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030
/* Speaker layout mask for four channels.
* - Speaker front left
* - Speaker front right
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033
/* Speaker layout mask for four channels (Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107
/* Speaker layout mask for five channels.
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037
/* Speaker layout mask for five channels (5 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607
/* Speaker layout mask for six channels (5.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F
/* Speaker layout mask for six channels (5.1 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F
/* Speaker layout mask for six channels (5.1 mode, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137
/* Speaker layout mask for six channels (5.1 mode, Home Theater,
* no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707
/* Speaker layout mask for seven channels (6.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back left
* - Speaker back right
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F
/* Speaker layout mask for seven channels (6.1 mode, Home
* Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F
/* Speaker layout mask for seven channels (6.1 mode, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7
/* Speaker layout mask for seven channels (6.1 mode, Home
* Theater, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637
/* Speaker layout mask for eight channels (7.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker low frequency
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \
0x00FF
/* Speaker layout mask for eight channels (7.1 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
* - Speaker low frequency
* - Speaker front left of center
* - Speaker front right of center
*
*/
#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \
0x063F
#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82
/* Maximum number of decoder output channels.*/
#define MAX_CHAN_MAP_CHANNELS 16
/* Structure for decoder output channel mapping. */
/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_dec_out_chan_map_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
u32 num_channels;
/* Number of decoder output channels.
* Supported values: 0 to #MAX_CHAN_MAP_CHANNELS
*
* A value of 0 indicates native channel mapping, which is valid
* only for NT mode. This means the output of the decoder is to be
* preserved as is.
*/
u8 channel_mapping[MAX_CHAN_MAP_CHANNELS];
} __packed;
#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84
/* Bitmask for the IEC 61937 enable flag.*/
#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL)
/* Shift value for the IEC 61937 enable flag.*/
#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0
/* Bitmask for the IEC 60958 enable flag.*/
#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL)
/* Shift value for the IEC 60958 enable flag.*/
#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1
/* Payload format for open write compressed comand */
/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
* comand, which opens a stream for a given session ID and stream ID
* to be rendered in the compressed format.
*/
struct asm_stream_cmd_open_write_compressed {
struct apr_hdr hdr;
u32 flags;
/* Mode flags that configure the stream for a specific format.
* Supported values:
* - Bit 0 -- IEC 61937 compatibility
* - 0 -- Stream is not in IEC 61937 format
* - 1 -- Stream is in IEC 61937 format
* - Bit 1 -- IEC 60958 compatibility
* - 0 -- Stream is not in IEC 60958 format
* - 1 -- Stream is in IEC 60958 format
* - Bits 2 to 31 -- 0 (Reserved)
*
* For the same stream, bit 0 cannot be set to 0 and bit 1 cannot
* be set to 1. A compressed stream connot have IEC 60958
* packetization applied without IEC 61937 packetization.
* @note1hang Currently, IEC 60958 packetized input streams are not
* supported.
*/
u32 fmt_id;
/* Specifies the media type of the HDMI stream to be opened.
* Supported values:
* - #ASM_MEDIA_FMT_AC3_DEC
* - #ASM_MEDIA_FMT_EAC3_DEC
* - #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;
#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95
struct asm_stream_cmd_open_read_compressed {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags that indicate whether meta information per encoded
* frame is to be provided.
* Supported values for bit 4:
* - 0 -- Return data buffer contains all encoded frames only; it does
* not contain frame metadata.
* - 1 -- Return data buffer contains an array of metadata and encoded
* frames.
* - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and
* #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit.
* All other bits are reserved; clients must set them to zero.
*/
u32 frames_per_buf;
/* Indicates the number of frames that need to be returned per
* read buffer
* Supported values: should be greater than 0
*/
} __packed;
/* adsp_asm_stream_commands.h*/
/* adsp_asm_api.h (no changes)*/
#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \
0x00010BE4
#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \
0x00010D83
#define ASM_STREAM_POSTPROCOPO_ID_NONE \
0x00010C68
#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \
0x00010D8B
#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \
ASM_STREAM_POSTPROCOPO_ID_DEFAULT
#define ASM_STREAM_PREPROCOPO_ID_NONE \
ASM_STREAM_POSTPROCOPO_ID_NONE
#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \
0x00010312
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \
0x00010313
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \
0x00010314
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\
0x00010704
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\
0x0001070D
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\
0x0001070E
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\
0x0001070F
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_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;
/* 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_VOL_CTRL_LR_CHANNEL_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
/* @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
/* 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_VOL_CTRL_LR_CHANNEL_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
/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL
* module.
*/
#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A
/* ID of the multiple-channel volume control parameters used by the
* #ASM_MODULE_ID_VOL_CTRL module.
*/
#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713
/* ID of the multiple-channel mute configuration parameters used by the
* #ASM_MODULE_ID_VOL_CTRL module.
*/
#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714
/* Structure for the master gain parameter for a volume control
* module.
*/
/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
* parameter used by the Volume Control module.
*/
struct asm_volume_ctrl_master_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint16_t master_gain;
/*< Linear gain in Q13 format. */
uint16_t reserved;
/*< Clients must set this field to zero.
*/
} __packed;
/* Structure for the left/right channel gain parameter for a
* volume control module.
*/
/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN
* parameters used by the Volume Control module.
*/
struct asm_volume_ctrl_lr_chan_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint16_t l_chan_gain;
/*< Linear gain in Q13 format for the left channel. */
uint16_t r_chan_gain;
/*< Linear gain in Q13 format for the right channel.*/
} __packed;
/* Structure for the mute configuration parameter for a
volume control module. */
/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
* parameter used by the Volume Control module.
*/
struct asm_volume_ctrl_mute_config {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t mute_flag;
/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/
} __packed;
/*
* Supported parameters for a soft stepping linear ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0
/*
* Exponential ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1
/*
* Logarithmic ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2
/* Structure for holding soft stepping volume parameters. */
/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
* parameters used by the Volume Control module.
*/
struct asm_soft_step_volume_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t period;
/*< Period in milliseconds.
* Supported values: 0 to 15000
*/
uint32_t step;
/*< Step in microseconds.
* Supported values: 0 to 15000000
*/
uint32_t ramping_curve;
/*< Ramping curve type.
* Supported values:
* - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
* - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
* - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
*/
} __packed;
/* Structure for holding soft pause parameters. */
/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
* parameters used by the Volume Control module.
*/
struct asm_soft_pause_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t enable_flag;
/*< Specifies whether soft pause is disabled (0) or enabled
* (nonzero).
*/
uint32_t period;
/*< Period in milliseconds.
* Supported values: 0 to 15000
*/
uint32_t step;
/*< Step in microseconds.
* Supported values: 0 to 15000000
*/
uint32_t ramping_curve;
/*< Ramping curve.
* Supported values:
* - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
* - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
* - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
*/
} __packed;
/* Maximum number of channels.*/
#define VOLUME_CONTROL_MAX_CHANNELS 8
/* Structure for holding one channel type - gain pair. */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel
* type/gain pairs used by the Volume Control module. \n \n This
* structure immediately follows the
* asm_volume_ctrl_multichannel_gain structure.
*/
struct asm_volume_ctrl_channelype_gain_pair {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint8_t channelype;
/*< Channel type for which the gain setting is to be applied.
* Supported values:
* - #PCM_CHANNEL_L
* - #PCM_CHANNEL_R
* - #PCM_CHANNEL_C
* - #PCM_CHANNEL_LS
* - #PCM_CHANNEL_RS
* - #PCM_CHANNEL_LFE
* - #PCM_CHANNEL_CS
* - #PCM_CHANNEL_LB
* - #PCM_CHANNEL_RB
* - #PCM_CHANNELS
* - #PCM_CHANNEL_CVH
* - #PCM_CHANNEL_MS
* - #PCM_CHANNEL_FLC
* - #PCM_CHANNEL_FRC
* - #PCM_CHANNEL_RLC
* - #PCM_CHANNEL_RRC
*/
uint8_t reserved1;
/*< Clients must set this field to zero. */
uint8_t reserved2;
/*< Clients must set this field to zero. */
uint8_t reserved3;
/*< Clients must set this field to zero. */
uint32_t gain;
/*< Gain value for this channel in Q28 format.
* Supported values: Any
*/
} __packed;
/* Structure for the multichannel gain command */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN
* parameters used by the Volume Control module.
*/
struct asm_volume_ctrl_multichannel_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t num_channels;
/*< Number of channels for which gain values are provided. Any
* channels present in the data for which gain is not provided are
* set to unity gain.
* Supported values: 1 to 8
*/
struct asm_volume_ctrl_channelype_gain_pair
gain_data[VOLUME_CONTROL_MAX_CHANNELS];
/*< Array of channel type/gain pairs.*/
} __packed;
/* Structure for holding one channel type - mute pair. */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel
* type/mute setting pairs used by the Volume Control module. \n \n
* This structure immediately follows the
* asm_volume_ctrl_multichannel_mute structure.
*/
struct asm_volume_ctrl_channelype_mute_pair {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint8_t channelype;
/*< Channel type for which the mute setting is to be applied.
* Supported values:
* - #PCM_CHANNEL_L
* - #PCM_CHANNEL_R
* - #PCM_CHANNEL_C
* - #PCM_CHANNEL_LS
* - #PCM_CHANNEL_RS
* - #PCM_CHANNEL_LFE
* - #PCM_CHANNEL_CS
* - #PCM_CHANNEL_LB
* - #PCM_CHANNEL_RB
* - #PCM_CHANNELS
* - #PCM_CHANNEL_CVH
* - #PCM_CHANNEL_MS
* - #PCM_CHANNEL_FLC
* - #PCM_CHANNEL_FRC
* - #PCM_CHANNEL_RLC
* - #PCM_CHANNEL_RRC
*/
uint8_t reserved1;
/*< Clients must set this field to zero. */
uint8_t reserved2;
/*< Clients must set this field to zero. */
uint8_t reserved3;
/*< Clients must set this field to zero. */
uint32_t mute;
/*< Mute setting for this channel.
* Supported values:
* - 0 = Unmute
* - Nonzero = Mute
*/
} __packed;
/* Structure for the multichannel mute command */
/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE
* parameters used by the Volume Control module.
*/
struct asm_volume_ctrl_multichannel_mute {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t num_channels;
/*< Number of channels for which mute configuration is
* provided. Any channels present in the data for which mute
* configuration is not provided are set to unmute.
* Supported values: 1 to 8
*/
struct asm_volume_ctrl_channelype_mute_pair
mute_data[VOLUME_CONTROL_MAX_CHANNELS];
/*< Array of channel type/mute setting pairs.*/
} __packed;
/* end_addtogroup audio_pp_param_ids */
/* audio_pp_module_ids
* ID of the IIR Tuning Filter module.
* This module supports the following parameter IDs:
* - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
* - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
* - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
*/
#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02
/* @addtogroup audio_pp_param_ids */
/* ID of the IIR tuning filter enable parameter used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
* @messagepayload
* @structure{asm_iiruning_filter_enable}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO
* NFIG.tex}
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03
/* ID of the IIR tuning filter pregain parameter used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04
/* ID of the IIR tuning filter configuration parameters used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05
/* Structure for an enable configuration parameter for an
* IIR tuning filter module.
*/
/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
* parameter used by the IIR Tuning Filter module.
*/
struct asm_iiruning_filter_enable {
uint32_t enable_flag;
/*< Specifies whether the IIR tuning filter is disabled (0) or
* enabled (1).
*/
} __packed;
/* Structure for the pregain parameter for an IIR tuning filter module. */
/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
* parameters used by the IIR Tuning Filter module.
*/
struct asm_iiruning_filter_pregain {
uint16_t pregain;
/*< Linear gain in Q13 format. */
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* Structure for the configuration parameter for an IIR tuning filter
* module.
*/
/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
* parameters used by the IIR Tuning Filter module. \n
* \n
* This structure is followed by the IIR filter coefficients: \n
* - Sequence of int32_t FilterCoeffs \n
* Five coefficients for each band. Each coefficient is in int32_t format, in
* the order of b0, b1, b2, a1, a2.
* - Sequence of int16_t NumShiftFactor \n
* One int16_t per band. The numerator shift factor is related to the Q
* factor of the filter coefficients.
* - Sequence of uint16_t PanSetting \n
* One uint16_t per band, indicating if the filter is applied to left (0),
* right (1), or both (2) channels.
*/
struct asm_iir_filter_config_params {
uint16_t num_biquad_stages;
/*< Number of bands.
* Supported values: 0 to 20
*/
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* audio_pp_module_ids
* ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx
* paths.
* This module supports the following parameter IDs:
* - #ASM_PARAM_ID_MBDRC_ENABLE
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
*/
#define ASM_MODULE_ID_MBDRC 0x00010C06
/* audio_pp_param_ids */
/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module.
* @messagepayload
* @structure{asm_mbdrc_enable}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex}
*/
#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07
/* ID of the MBDRC configuration parameters used by the
* #ASM_MODULE_ID_MBDRC module.
* @messagepayload
* @structure{asm_mbdrc_config_params}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex}
*
* @parspace Sub-band DRC configuration parameters
* @structure{asm_subband_drc_config_params}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex}
*
* @keep{6}
* To obtain legacy ADRC from MBDRC, use the calibration tool to:
*
* - Enable MBDRC (EnableFlag = TRUE)
* - Set number of bands to 1 (uiNumBands = 1)
* - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1)
* - Clear the first band mute flag (MuteFlag[0] = 0)
* - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000)
* - Use the legacy ADRC parameters to calibrate the rest of the MBDRC
* parameters.
*/
#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08
/* end_addtogroup audio_pp_param_ids */
/* audio_pp_module_ids
* ID of the MMBDRC module version 2 pre/postprocessing block.
* This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in
* the length of the filters used in each sub-band.
* This module supports the following parameter ID:
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2
*/
#define ASM_MODULE_ID_MBDRCV2 0x0001070B
/* @addtogroup audio_pp_param_ids */
/* ID of the configuration parameters used by the
* #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure
* of the MBDRC v2 pre/postprocessing block.
* The update to this configuration structure from the original
* MBDRC is the number of filter coefficients in the filter
* structure. The sequence for is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t
* padding
* - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags +
* uint16_t padding
* This block uses the same parameter structure as
* #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.
*/
#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \
0x0001070C
/* Structure for the enable parameter for an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the
* MBDRC module.
*/
struct asm_mbdrc_enable {
uint32_t enable_flag;
/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/
} __packed;
/* Structure for the configuration parameters for an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
* parameters used by the MBDRC module. \n \n Following this
* structure is the payload for sub-band DRC configuration
* parameters (asm_subband_drc_config_params). This sub-band
* structure must be repeated for each band.
*/
struct asm_mbdrc_config_params {
uint16_t num_bands;
/*< Number of bands.
* Supported values: 1 to 5
*/
int16_t limiterhreshold;
/*< Threshold in decibels for the limiter output.
* Supported values: -72 to 18 \n
* Recommended value: 3994 (-0.22 db in Q3.12 format)
*/
int16_t limiter_makeup_gain;
/*< Makeup gain in decibels for the limiter output.
* Supported values: -42 to 42 \n
* Recommended value: 256 (0 dB in Q7.8 format)
*/
int16_t limiter_gc;
/*< Limiter gain recovery coefficient.
* Supported values: 0.5 to 0.99 \n
* Recommended value: 32440 (0.99 in Q15 format)
*/
int16_t limiter_delay;
/*< Limiter delay in samples.
* Supported values: 0 to 10 \n
* Recommended value: 262 (0.008 samples in Q15 format)
*/
int16_t limiter_max_wait;
/*< Maximum limiter waiting time in samples.
* Supported values: 0 to 10 \n
* Recommended value: 262 (0.008 samples in Q15 format)
*/
} __packed;
/* DRC configuration structure for each sub-band of an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC
* configuration parameters for each sub-band in the MBDRC module.
* After this DRC structure is configured for valid bands, the next
* MBDRC setparams expects the sequence of sub-band MBDRC filter
* coefficients (the length depends on the number of bands) plus the
* mute flag for that band plus uint16_t padding.
*
* @keep{10}
* The filter coefficient and mute flag are of type int16_t:
* - FIR coefficient = int16_t firFilter
* - Mute flag = int16_t fMuteFlag
*
* The sequence is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding
* - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding
*
* For improved filterbank, the sequence is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding
* - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding
*/
struct asm_subband_drc_config_params {
int16_t drc_stereo_linked_flag;
/*< Specifies whether all stereo channels have the same applied
* dynamics (1) or if they process their dynamics independently (0).
* Supported values:
* - 0 -- Not linked
* - 1 -- Linked
*/
int16_t drc_mode;
/*< Specifies whether DRC mode is bypassed for sub-bands.
* Supported values:
* - 0 -- Disabled
* - 1 -- Enabled
*/
int16_t drc_down_sample_level;
/*< DRC down sample level.
* Supported values: @ge 1
*/
int16_t drc_delay;
/*< DRC delay in samples.
* Supported values: 0 to 1200
*/
uint16_t drc_rmsime_avg_const;
/*< RMS signal energy time-averaging constant.
* Supported values: 0 to 2^16-1
*/
uint16_t drc_makeup_gain;
/*< DRC makeup gain in decibels.
* Supported values: 258 to 64917
*/
/* Down expander settings */
int16_t down_expdrhreshold;
/*< Down expander threshold.
* Supported Q7 format values: 1320 to up_cmpsrhreshold
*/
int16_t down_expdr_slope;
/*< Down expander slope.
* Supported Q8 format values: -32768 to 0.
*/
uint32_t down_expdr_attack;
/*< Down expander attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t down_expdr_release;
/*< Down expander release constant.
* Supported Q31 format values: 19685 to 2^31
*/
uint16_t down_expdr_hysteresis;
/*< Down expander hysteresis constant.
* Supported Q14 format values: 1 to 32690
*/
uint16_t reserved;
/*< Clients must set this field to zero. */
int32_t down_expdr_min_gain_db;
/*< Down expander minimum gain.
* Supported Q23 format values: -805306368 to 0.
*/
/* Up compressor settings */
int16_t up_cmpsrhreshold;
/*< Up compressor threshold.
* Supported Q7 format values: down_expdrhreshold to
* down_cmpsrhreshold.
*/
uint16_t up_cmpsr_slope;
/*< Up compressor slope.
* Supported Q16 format values: 0 to 64881.
*/
uint32_t up_cmpsr_attack;
/*< Up compressor attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t up_cmpsr_release;
/*< Up compressor release constant.
* Supported Q31 format values: 19685 to 2^31.
*/
uint16_t up_cmpsr_hysteresis;
/*< Up compressor hysteresis constant.
* Supported Q14 format values: 1 to 32690.
*/
/* Down compressor settings */
int16_t down_cmpsrhreshold;
/*< Down compressor threshold.
* Supported Q7 format values: up_cmpsrhreshold to 11560.
*/
uint16_t down_cmpsr_slope;
/*< Down compressor slope.
* Supported Q16 format values: 0 to 64881.
*/
uint16_t reserved1;
/*< Clients must set this field to zero. */
uint32_t down_cmpsr_attack;
/*< Down compressor attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t down_cmpsr_release;
/*< Down compressor release constant.
* Supported Q31 format values: 19685 to 2^31.
*/
uint16_t down_cmpsr_hysteresis;
/*< Down compressor hysteresis constant.
* Supported Q14 values: 1 to 32690.
*/
uint16_t reserved2;
/*< Clients must set this field to zero.*/
} __packed;
#define ASM_MODULE_ID_EQUALIZER 0x00010C27
#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28
#define ASM_MAX_EQ_BANDS 12
struct asm_eq_per_band_params {
uint32_t band_idx;
/*< Band index.
* Supported values: 0 to 11
*/
uint32_t filterype;
/*< Type of filter.
* Supported values:
* - #ASM_PARAM_EQYPE_NONE
* - #ASM_PARAM_EQ_BASS_BOOST
* - #ASM_PARAM_EQ_BASS_CUT
* - #ASM_PARAM_EQREBLE_BOOST
* - #ASM_PARAM_EQREBLE_CUT
* - #ASM_PARAM_EQ_BAND_BOOST
* - #ASM_PARAM_EQ_BAND_CUT
*/
uint32_t center_freq_hz;
/*< Filter band center frequency in Hertz. */
int32_t filter_gain;
/*< Filter band initial gain.
* Supported values: +12 to -12 dB in 1 dB increments
*/
int32_t q_factor;
/*< Filter band quality factor expressed as a Q8 number, i.e., a
* fixed-point number with q factor of 8. For example, 3000/(2^8).
*/
} __packed;
struct asm_eq_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t enable_flag;
/*< Specifies whether the equalizer module is disabled (0) or enabled
* (nonzero).
*/
uint32_t num_bands;
/*< Number of bands.
* Supported values: 1 to 12
*/
struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS];
} __packed;
/* No equalizer effect.*/
#define ASM_PARAM_EQYPE_NONE 0
/* Bass boost equalizer effect.*/
#define ASM_PARAM_EQ_BASS_BOOST 1
/*Bass cut equalizer effect.*/
#define ASM_PARAM_EQ_BASS_CUT 2
/* Treble boost equalizer effect */
#define ASM_PARAM_EQREBLE_BOOST 3
/* Treble cut equalizer effect.*/
#define ASM_PARAM_EQREBLE_CUT 4
/* Band boost equalizer effect.*/
#define ASM_PARAM_EQ_BAND_BOOST 5
/* Band cut equalizer effect.*/
#define ASM_PARAM_EQ_BAND_CUT 6
/* Voice get & set params */
#define VOICE_CMD_SET_PARAM 0x0001133D
#define VOICE_CMD_GET_PARAM 0x0001133E
#define VOICE_EVT_GET_PARAM_ACK 0x00011008
/* 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
/* Operation is finished. */
#define ADSP_ETERMINATED 0x00011174
/*bharath, adsp_error_codes.h */
#endif /*_APR_AUDIO_V2_H_ */