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gerrit-public.fairphone.software / kernel / msm-4.9 / cd76336820fc9d32aa9c8b570b594c72b5184388 / . / include / sound / apr_audio-v2.h
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/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
#ifndef _APR_AUDIO_V2_H_
#define _APR_AUDIO_V2_H_
#include <linux/qdsp6v2/apr.h>
/* size of header needed for passing data out of band */
#define APR_CMD_OB_HDR_SZ 12
/* size of header needed for getting data */
#define APR_CMD_GET_HDR_SZ 16
struct param_outband {
size_t size;
void *kvaddr;
phys_addr_t paddr;
};
#define ADSP_ADM_VERSION 0x00070000
#define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322
#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323
#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324
#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D
/* Enumeration for an audio Rx matrix ID.*/
#define ADM_MATRIX_ID_AUDIO_RX 0
#define ADM_MATRIX_ID_AUDIO_TX 1
#define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX 2
#define ADM_MATRIX_ID_LISTEN_TX 4
/* Enumeration for an audio Tx matrix ID.*/
#define ADM_MATRIX_ID_AUDIOX 1
#define ADM_MAX_COPPS 5
/* make sure this matches with msm_audio_calibration */
#define SP_V2_NUM_MAX_SPKR 2
/* Session map node structure.
* Immediately following this structure are num_copps
* entries of COPP IDs. The COPP IDs are 16 bits, so
* there might be a padding 16-bit field if num_copps
* is odd.
*/
struct adm_session_map_node_v5 {
u16 session_id;
/* Handle of the ASM session to be routed. Supported values: 1
* to 8.
*/
u16 num_copps;
/* Number of COPPs to which this session is to be routed.
* Supported values: 0 < num_copps <= ADM_MAX_COPPS.
*/
} __packed;
/* Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command.
* Immediately following this structure are num_sessions of the session map
* node payload (adm_session_map_node_v5).
*/
struct adm_cmd_matrix_map_routings_v5 {
struct apr_hdr hdr;
u32 matrix_id;
/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx
* (1). Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
* macros to set this field.
*/
u32 num_sessions;
/* Number of sessions being updated by this command (optional). */
} __packed;
/* This command allows a client to open a COPP/Voice Proc. TX module
* and sets up the device session: Matrix -> COPP -> AFE on the RX
* and AFE -> COPP -> Matrix on the TX. This enables PCM data to
* be transferred to/from the endpoint (AFEPortID).
*
* @return
* #ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and COPP ID.
*/
#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
/* This command allows a client to open a COPP/Voice Proc the
* way as ADM_CMD_DEVICE_OPEN_V5 but supports multiple endpoint2
* channels.
*
* @return
* #ADM_CMDRSP_DEVICE_OPEN_V6 with the resulting status and
* COPP ID.
*/
#define ADM_CMD_DEVICE_OPEN_V6 0x00010356
/* Definition for a low latency stream session. */
#define ADM_LOW_LATENCY_DEVICE_SESSION 0x2000
/* Definition for a ultra low latency stream session. */
#define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION 0x4000
/* Definition for a ultra low latency with Post Processing stream session. */
#define ADM_ULL_POST_PROCESSING_DEVICE_SESSION 0x8000
/* Definition for a legacy device session. */
#define ADM_LEGACY_DEVICE_SESSION 0
/* Indicates that endpoint_id_2 is to be ignored.*/
#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2
#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3
/* Indicates that an audio COPP is to send/receive a mono PCM
* stream to/from
* END_POINT_ID_1.
*/
#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1
/* Indicates that an audio COPP is to send/receive a
* stereo PCM stream to/from END_POINT_ID_1.
*/
#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2
/* Sample rate is 8000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000
/* Sample rate is 16000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000
/* Sample rate is 48000 Hz.*/
#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000
/* Definition for a COPP live input flag bitmask.*/
#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U)
/* Definition for a COPP live shift value bitmask.*/
#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0
/* Definition for the COPP ID bitmask.*/
#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL)
/* Definition for the COPP ID shift value.*/
#define ADM_SHIFT_COPP_ID 0
/* Definition for the service ID bitmask.*/
#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
/* Definition for the service ID shift value.*/
#define ADM_SHIFT_SERVICE_ID 16
/* Definition for the domain ID bitmask.*/
#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
/* Definition for the domain ID shift value.*/
#define ADM_SHIFT_DOMAIN_ID 24
/* ADM device open command payload of the
* #ADM_CMD_DEVICE_OPEN_V5 command.
*/
struct adm_cmd_device_open_v5 {
struct apr_hdr hdr;
u16 flags;
/* Reserved for future use. Clients must set this field
* to zero.
*/
u16 mode_of_operation;
/* Specifies whether the COPP must be opened on the Tx or Rx
* path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
* supported values and interpretation.
* Supported values:
* - 0x1 -- Rx path COPP
* - 0x2 -- Tx path live COPP
* - 0x3 -- Tx path nonlive COPP
* Live connections cause sample discarding in the Tx device
* matrix if the destination output ports do not pull them
* fast enough. Nonlive connections queue the samples
* indefinitely.
*/
u16 endpoint_id_1;
/* Logical and physical endpoint ID of the audio path.
* If the ID is a voice processor Tx block, it receives near
* samples. Supported values: Any pseudoport, AFE Rx port,
* or AFE Tx port For a list of valid IDs, refer to
* @xhyperref{Q4,[Q4]}.
* Q4 = Hexagon Multimedia: AFE Interface Specification
*/
u16 endpoint_id_2;
/* Logical and physical endpoint ID 2 for a voice processor
* Tx block.
* This is not applicable to audio COPP.
* Supported values:
* - AFE Rx port
* - 0xFFFF -- Endpoint 2 is unavailable and the voice
* processor Tx
* block ignores this endpoint
* When the voice processor Tx block is created on the audio
* record path,
* it can receive far-end samples from an AFE Rx port if the
* voice call
* is active. The ID of the AFE port is provided in this
* field.
* For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
*/
u32 topology_id;
/* Audio COPP topology ID; 32-bit GUID. */
u16 dev_num_channel;
/* Number of channels the audio COPP sends to/receives from
* the endpoint.
* Supported values: 1 to 8.
* The value is ignored for the voice processor Tx block,
* where channel
* configuration is derived from the topology ID.
*/
u16 bit_width;
/* Bit width (in bits) that the audio COPP sends to/receives
* from the
* endpoint. The value is ignored for the voice processing
* Tx block,
* where the PCM width is 16 bits.
*/
u32 sample_rate;
/* Sampling rate at which the audio COPP/voice processor
* Tx block
* interfaces with the endpoint.
* Supported values for voice processor Tx: 8000, 16000,
* 48000 Hz
* Supported values for audio COPP: >0 and <=192 kHz
*/
u8 dev_channel_mapping[8];
/* Array of channel mapping of buffers that the audio COPP
* sends to the endpoint. Channel[i] mapping describes channel
* I inside the buffer, where 0 < i < dev_num_channel.
* This value is relevant only for an audio Rx COPP.
* For the voice processor block and Tx audio block, this field
* is set to zero and is ignored.
*/
} __packed;
/* ADM device open command payload of the
* #ADM_CMD_DEVICE_OPEN_V6 command.
*/
struct adm_cmd_device_open_v6 {
struct apr_hdr hdr;
u16 flags;
/* Reserved for future use. Clients must set this field
* to zero.
*/
u16 mode_of_operation;
/* Specifies whether the COPP must be opened on the Tx or Rx
* path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
* supported values and interpretation.
* Supported values:
* - 0x1 -- Rx path COPP
* - 0x2 -- Tx path live COPP
* - 0x3 -- Tx path nonlive COPP
* Live connections cause sample discarding in the Tx device
* matrix if the destination output ports do not pull them
* fast enough. Nonlive connections queue the samples
* indefinitely.
*/
u16 endpoint_id_1;
/* Logical and physical endpoint ID of the audio path.
* If the ID is a voice processor Tx block, it receives near
* samples. Supported values: Any pseudoport, AFE Rx port,
* or AFE Tx port For a list of valid IDs, refer to
* @xhyperref{Q4,[Q4]}.
* Q4 = Hexagon Multimedia: AFE Interface Specification
*/
u16 endpoint_id_2;
/* Logical and physical endpoint ID 2 for a voice processor
* Tx block.
* This is not applicable to audio COPP.
* Supported values:
* - AFE Rx port
* - 0xFFFF -- Endpoint 2 is unavailable and the voice
* processor Tx
* block ignores this endpoint
* When the voice processor Tx block is created on the audio
* record path,
* it can receive far-end samples from an AFE Rx port if the
* voice call
* is active. The ID of the AFE port is provided in this
* field.
* For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
*/
u32 topology_id;
/* Audio COPP topology ID; 32-bit GUID. */
u16 dev_num_channel;
/* Number of channels the audio COPP sends to/receives from
* the endpoint.
* Supported values: 1 to 8.
* The value is ignored for the voice processor Tx block,
* where channel
* configuration is derived from the topology ID.
*/
u16 bit_width;
/* Bit width (in bits) that the audio COPP sends to/receives
* from the
* endpoint. The value is ignored for the voice processing
* Tx block,
* where the PCM width is 16 bits.
*/
u32 sample_rate;
/* Sampling rate at which the audio COPP/voice processor
* Tx block
* interfaces with the endpoint.
* Supported values for voice processor Tx: 8000, 16000,
* 48000 Hz
* Supported values for audio COPP: >0 and <=192 kHz
*/
u8 dev_channel_mapping[8];
/* Array of channel mapping of buffers that the audio COPP
* sends to the endpoint. Channel[i] mapping describes channel
* I inside the buffer, where 0 < i < dev_num_channel.
* This value is relevant only for an audio Rx COPP.
* For the voice processor block and Tx audio block, this field
* is set to zero and is ignored.
*/
u16 dev_num_channel_eid2;
/* Number of channels the voice processor block sends
* to/receives from the endpoint2.
* Supported values: 1 to 8.
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u16 bit_width_eid2;
/* Bit width (in bits) that the voice processor sends
* to/receives from the endpoint2.
* Supported values: 16 and 24.
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u32 sample_rate_eid2;
/* Sampling rate at which the voice processor Tx block
* interfaces with the endpoint2.
* Supported values for Tx voice processor: >0 and <=384 kHz
* The value is ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
u8 dev_channel_mapping_eid2[8];
/* Array of channel mapping of buffers that the voice processor
* sends to the endpoint. Channel[i] mapping describes channel
* I inside the buffer, where 0 < i < dev_num_channel.
* This value is relevant only for the Tx voice processor.
* The values are ignored for audio COPP or if endpoint_id_2 is
* set to 0xFFFF.
*/
} __packed;
/*
* This command allows the client to close a COPP and disconnect
* the device session.
*/
#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
/* Sets one or more parameters to a COPP. */
#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328
/* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command.
* If the data_payload_addr_lsw and data_payload_addr_msw element
* are NULL, a series of adm_param_datastructures immediately
* follows, whose total size is data_payload_size bytes.
*/
struct adm_cmd_set_pp_params_v5 {
struct apr_hdr hdr;
u32 payload_addr_lsw;
/* LSW of parameter data payload address. */
u32 payload_addr_msw;
/* MSW of parameter data payload address. */
u32 mem_map_handle;
/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS
* command
*
* If mem_map_handle is zero implies the message is in
* the payload
*/
u32 payload_size;
/* Size in bytes of the variable payload accompanying this
* message or
* in shared memory. This is used for parsing the parameter
* payload.
*/
} __packed;
/* Payload format for COPP parameter data.
* Immediately following this structure are param_size bytes
* of parameter
* data.
*/
struct adm_param_data_v5 {
u32 module_id;
/* Unique ID of the module. */
u32 param_id;
/* Unique ID of the parameter. */
u16 param_size;
/* Data size of the param_id/module_id combination.
* This value is a
* multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for future enhancements.
* This field must be set to zero.
*/
} __packed;
/* set customized mixing on matrix mixer */
#define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 0x00010344
struct adm_cmd_set_pspd_mtmx_strtr_params_v5 {
struct apr_hdr hdr;
/* LSW of parameter data payload address.*/
u32 payload_addr_lsw;
/* MSW of parameter data payload address.*/
u32 payload_addr_msw;
/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */
/* command. If mem_map_handle is zero implies the message is in */
/* the payload */
u32 mem_map_handle;
/* Size in bytes of the variable payload accompanying this */
/* message or in shared memory. This is used for parsing the */
/* parameter payload. */
u32 payload_size;
u16 direction;
u16 sessionid;
u16 deviceid;
u16 reserved;
} __packed;
/* Defined specifically for in-band use, includes params */
struct adm_cmd_set_pp_params_inband_v5 {
struct apr_hdr hdr;
/* LSW of parameter data payload address.*/
u32 payload_addr_lsw;
/* MSW of parameter data payload address.*/
u32 payload_addr_msw;
/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */
/* command. If mem_map_handle is zero implies the message is in */
/* the payload */
u32 mem_map_handle;
/* Size in bytes of the variable payload accompanying this */
/* message or in shared memory. This is used for parsing the */
/* parameter payload. */
u32 payload_size;
/* Parameters passed for in band payload */
struct adm_param_data_v5 params;
} __packed;
/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
*/
#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message,
* which returns the
* status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
*/
struct adm_cmd_rsp_device_open_v5 {
u32 status;
/* Status message (error code).*/
u16 copp_id;
/* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/
u16 reserved;
/* Reserved. This field must be set to zero.*/
} __packed;
/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command. */
#define ADM_CMDRSP_DEVICE_OPEN_V6 0x00010357
/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message,
* which returns the
* status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command
* is the exact same as ADM_CMDRSP_DEVICE_OPEN_V5.
*/
/* This command allows a query of one COPP parameter. */
#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A
/* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command. */
struct adm_cmd_get_pp_params_v5 {
struct apr_hdr hdr;
u32 data_payload_addr_lsw;
/* LSW of parameter data payload address.*/
u32 data_payload_addr_msw;
/* MSW of parameter data payload address.*/
/* If the mem_map_handle is non zero,
* on ACK, the ParamData payloads begin at
* the address specified (out-of-band).
*/
u32 mem_map_handle;
/* Memory map handle returned
* by ADM_CMD_SHARED_MEM_MAP_REGIONS command.
* If the mem_map_handle is 0, it implies that
* the ACK's payload will contain the ParamData (in-band).
*/
u32 module_id;
/* Unique ID of the module. */
u32 param_id;
/* Unique ID of the parameter. */
u16 param_max_size;
/* Maximum data size of the parameter
*ID/module ID combination. This
* field is a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for future enhancements.
* This field must be set to zero.
*/
} __packed;
/* Returns parameter values
* in response to an #ADM_CMD_GET_PP_PARAMS_V5 command.
*/
#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B
/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message,
* which returns parameter values in response
* to an #ADM_CMD_GET_PP_PARAMS_V5 command.
* Immediately following this
* structure is the adm_param_data_v5
* structure containing the pre/postprocessing
* parameter data. For an in-band
* scenario, the variable payload depends
* on the size of the parameter.
*/
struct adm_cmd_rsp_get_pp_params_v5 {
u32 status;
/* Status message (error code).*/
} __packed;
/* Structure for holding soft stepping volume parameters. */
/*
* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
* parameters used by the Volume Control module.
*/
struct audproc_softvolume_params {
u32 period;
u32 step;
u32 rampingcurve;
} __packed;
/*
* ID of the Media Format Converter (MFC) module.
* This module supports the following parameter IDs:
* #AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT
* #AUDPROC_CHMIXER_PARAM_ID_COEFF
*/
#define AUDPROC_MODULE_ID_MFC 0x00010912
/* ID of the Output Media Format parameters used by AUDPROC_MODULE_ID_MFC.
*
*/
#define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913
struct audproc_mfc_output_media_fmt {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
uint32_t sampling_rate;
uint16_t bits_per_sample;
uint16_t num_channels;
uint16_t channel_type[8];
} __packed;
struct audproc_volume_ctrl_master_gain {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
/* Linear gain in Q13 format. */
uint16_t master_gain;
/* Clients must set this field to zero. */
uint16_t reserved;
} __packed;
struct audproc_soft_step_volume_params {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
/*
* Period in milliseconds.
* Supported values: 0 to 15000
*/
uint32_t period;
/*
* Step in microseconds.
* Supported values: 0 to 15000000
*/
uint32_t step;
/*
* Ramping curve type.
* Supported values:
* - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LINEAR
* - #AUDPROC_PARAM_SVC_RAMPINGCURVE_EXP
* - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LOG
*/
uint32_t ramping_curve;
} __packed;
struct audproc_enable_param_t {
struct adm_cmd_set_pp_params_inband_v5 pp_params;
/*
* Specifies whether the Audio processing module is enabled.
* This parameter is generic/common parameter to configure or
* determine the state of any audio processing module.
* @values 0 : Disable 1: Enable
*/
uint32_t enable;
};
/*
* Allows a client to control the gains on various session-to-COPP paths.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C
/* Indicates that the target gain in the
* current adm_session_copp_gain_v5
* structure is to be applied to all
* the session-to-COPP paths that exist for
* the specified session.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
/* Indicates that the target gain is
* to be immediately applied to the
* specified session-to-COPP path,
* without a ramping fashion.
*/
#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000
/* Enumeration for a linear ramping curve.*/
#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000
/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
* Immediately following this structure are num_gains of the
* adm_session_copp_gain_v5structure.
*/
struct adm_cmd_matrix_ramp_gains_v5 {
u32 matrix_id;
/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
* Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
* macros to set this field.
*/
u16 num_gains;
/* Number of gains being applied. */
u16 reserved_for_align;
/* Reserved. This field must be set to zero.*/
} __packed;
/* Session-to-COPP path gain structure, used by the
* #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
* This structure specifies the target
* gain (per channel) that must be applied
* to a particular session-to-COPP path in
* the audio matrix. The structure can
* also be used to apply the gain globally
* to all session-to-COPP paths that
* exist for the given session.
* The aDSP uses device channel mapping to
* determine which channel gains to
* use from this command. For example,
* if the device is configured as stereo,
* the aDSP uses only target_gain_ch_1 and
* target_gain_ch_2, and it ignores
* the others.
*/
struct adm_session_copp_gain_v5 {
u16 session_id;
/* Handle of the ASM session.
* Supported values: 1 to 8.
*/
u16 copp_id;
/* Handle of the COPP. Gain will be applied on the Session ID
* COPP ID path.
*/
u16 ramp_duration;
/* Duration (in milliseconds) of the ramp over
* which target gains are
* to be applied. Use
* #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE
* to indicate that gain must be applied immediately.
*/
u16 step_duration;
/* Duration (in milliseconds) of each step in the ramp.
* This parameter is ignored if ramp_duration is equal to
* #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE.
* Supported value: 1
*/
u16 ramp_curve;
/* Type of ramping curve.
* Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR
*/
u16 reserved_for_align;
/* Reserved. This field must be set to zero. */
u16 target_gain_ch_1;
/* Target linear gain for channel 1 in Q13 format; */
u16 target_gain_ch_2;
/* Target linear gain for channel 2 in Q13 format; */
u16 target_gain_ch_3;
/* Target linear gain for channel 3 in Q13 format; */
u16 target_gain_ch_4;
/* Target linear gain for channel 4 in Q13 format; */
u16 target_gain_ch_5;
/* Target linear gain for channel 5 in Q13 format; */
u16 target_gain_ch_6;
/* Target linear gain for channel 6 in Q13 format; */
u16 target_gain_ch_7;
/* Target linear gain for channel 7 in Q13 format; */
u16 target_gain_ch_8;
/* Target linear gain for channel 8 in Q13 format; */
} __packed;
/* Allows to set mute/unmute on various session-to-COPP paths.
* For every session-to-COPP path (stream-device interconnection),
* mute/unmute can be set individually on the output channels.
*/
#define ADM_CMD_MATRIX_MUTE_V5 0x0001032D
/* Indicates that mute/unmute in the
* current adm_session_copp_mute_v5structure
* is to be applied to all the session-to-COPP
* paths that exist for the specified session.
*/
#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
/* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/
struct adm_cmd_matrix_mute_v5 {
u32 matrix_id;
/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
* Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
* macros to set this field.
*/
u16 session_id;
/* Handle of the ASM session.
* Supported values: 1 to 8.
*/
u16 copp_id;
/* Handle of the COPP.
* Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS
* to indicate that mute/unmute must be applied to
* all the COPPs connected to session_id.
* Supported values:
* - 0xFFFF -- Apply mute/unmute to all connected COPPs
* - Other values -- Valid COPP ID
*/
u8 mute_flag_ch_1;
/* Mute flag for channel 1 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_2;
/* Mute flag for channel 2 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_3;
/* Mute flag for channel 3 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_4;
/* Mute flag for channel 4 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_5;
/* Mute flag for channel 5 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_6;
/* Mute flag for channel 6 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_7;
/* Mute flag for channel 7 is set to unmute (0) or mute (1). */
u8 mute_flag_ch_8;
/* Mute flag for channel 8 is set to unmute (0) or mute (1). */
u16 ramp_duration;
/* Period (in milliseconds) over which the soft mute/unmute will be
* applied.
* Supported values: 0 (Default) to 0xFFFF
* The default of 0 means mute/unmute will be applied immediately.
*/
u16 reserved_for_align;
/* Clients must set this field to zero.*/
} __packed;
#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8)
struct asm_aac_stereo_mix_coeff_selection_param_v2 {
struct apr_hdr hdr;
u32 param_id;
u32 param_size;
u32 aac_stereo_mix_coeff_flag;
} __packed;
/* Allows a client to connect the desired stream to
* the desired AFE port through the stream router
*
* This command allows the client to connect specified session to
* specified AFE port. This is used for compressed streams only
* opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
* #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command.
*
* @prerequisites
* Session ID and AFE Port ID must be valid.
* #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
* #ASM_STREAM_CMD_OPEN_READ_COMPRESSED
* must have been called on this session.
*/
#define ADM_CMD_CONNECT_AFE_PORT_V5 0x0001032E
#define ADM_CMD_DISCONNECT_AFE_PORT_V5 0x0001032F
/* Enumeration for the Rx stream router ID.*/
#define ADM_STRTR_ID_RX 0
/* Enumeration for the Tx stream router ID.*/
#define ADM_STRTR_IDX 1
/* Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/
struct adm_cmd_connect_afe_port_v5 {
struct apr_hdr hdr;
u8 mode;
/* ID of the stream router (RX/TX). Use the
* ADM_STRTR_ID_RX or ADM_STRTR_IDX macros
* to set this field.
*/
u8 session_id;
/* Session ID of the stream to connect */
u16 afe_port_id;
/* Port ID of the AFE port to connect to.*/
u32 num_channels;
/* Number of device channels
* Supported values: 2(Audio Sample Packet),
* 8 (HBR Audio Stream Sample Packet)
*/
u32 sampling_rate;
/* Device sampling rate
* Supported values: Any
*/
} __packed;
/* adsp_adm_api.h */
/* Port ID. Update afe_get_port_index
* when a new port is added here.
*/
#define PRIMARY_I2S_RX 0
#define PRIMARY_I2S_TX 1
#define SECONDARY_I2S_RX 4
#define SECONDARY_I2S_TX 5
#define MI2S_RX 6
#define MI2S_TX 7
#define HDMI_RX 8
#define RSVD_2 9
#define RSVD_3 10
#define DIGI_MIC_TX 11
#define VOICE2_PLAYBACK_TX 0x8002
#define VOICE_RECORD_RX 0x8003
#define VOICE_RECORD_TX 0x8004
#define VOICE_PLAYBACK_TX 0x8005
/* Slimbus Multi channel port id pool */
#define SLIMBUS_0_RX 0x4000
#define SLIMBUS_0_TX 0x4001
#define SLIMBUS_1_RX 0x4002
#define SLIMBUS_1_TX 0x4003
#define SLIMBUS_2_RX 0x4004
#define SLIMBUS_2_TX 0x4005
#define SLIMBUS_3_RX 0x4006
#define SLIMBUS_3_TX 0x4007
#define SLIMBUS_4_RX 0x4008
#define SLIMBUS_4_TX 0x4009
#define SLIMBUS_5_RX 0x400a
#define SLIMBUS_5_TX 0x400b
#define SLIMBUS_6_RX 0x400c
#define SLIMBUS_6_TX 0x400d
#define SLIMBUS_7_RX 0x400e
#define SLIMBUS_7_TX 0x400f
#define SLIMBUS_8_RX 0x4010
#define SLIMBUS_8_TX 0x4011
#define SLIMBUS_PORT_LAST SLIMBUS_8_TX
#define INT_BT_SCO_RX 0x3000
#define INT_BT_SCO_TX 0x3001
#define INT_BT_A2DP_RX 0x3002
#define INT_FM_RX 0x3004
#define INT_FM_TX 0x3005
#define RT_PROXY_PORT_001_RX 0x2000
#define RT_PROXY_PORT_001_TX 0x2001
#define DISPLAY_PORT_RX 0x6020
#define AFE_PORT_INVALID 0xFFFF
#define SLIMBUS_INVALID AFE_PORT_INVALID
#define AFE_PORT_CMD_START 0x000100ca
#define AFE_EVENT_RTPORT_START 0
#define AFE_EVENT_RTPORT_STOP 1
#define AFE_EVENT_RTPORT_LOW_WM 2
#define AFE_EVENT_RTPORT_HI_WM 3
#define ADSP_AFE_VERSION 0x00200000
/* Size of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE 0xF
/* Size of the range of port IDs for internal BT-FM ports. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE 0x6
/* Size of the range of port IDs for SLIMbus<sup>&reg;
* </sup> multichannel
* ports.
*/
#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE 0xA
/* Size of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE 0x2
/* Size of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE 0x5
/* Start of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START 0x1000
/* End of the range of port IDs for the audio interface. */
#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \
(AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\
AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1)
/* Start of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_START 0x2000
/* End of the range of port IDs for real-time proxy ports. */
#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \
(AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\
AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1)
/* Start of the range of port IDs for internal BT-FM devices. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START 0x3000
/* End of the range of port IDs for internal BT-FM devices. */
#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \
(AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\
AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1)
/* Start of the range of port IDs for SLIMbus devices. */
#define AFE_PORT_ID_SLIMBUS_RANGE_START 0x4000
/* End of the range of port IDs for SLIMbus devices. */
#define AFE_PORT_ID_SLIMBUS_RANGE_END \
(AFE_PORT_ID_SLIMBUS_RANGE_START +\
AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1)
/* Start of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_START 0x8001
/* End of the range of port IDs for pseudoports. */
#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \
(AFE_PORT_ID_PSEUDOPORT_RANGE_START +\
AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1)
/* Start of the range of port IDs for TDM devices. */
#define AFE_PORT_ID_TDM_PORT_RANGE_START 0x9000
/* End of the range of port IDs for TDM devices. */
#define AFE_PORT_ID_TDM_PORT_RANGE_END \
(AFE_PORT_ID_TDM_PORT_RANGE_START+0x40-1)
/* Size of the range of port IDs for TDM ports. */
#define AFE_PORT_ID_TDM_PORT_RANGE_SIZE \
(AFE_PORT_ID_TDM_PORT_RANGE_END - \
AFE_PORT_ID_TDM_PORT_RANGE_START+1)
#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004
#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005
#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
#define AUDIO_PORT_ID_I2S_RX 0x1008
#define AFE_PORT_ID_DIGITAL_MIC_TX 0x1009
#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A
#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C
#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
#define AFE_PORT_ID_SECONDARY_MI2S_RX_SD1 0x1010
#define AFE_PORT_ID_TERTIARY_PCM_RX 0x1012
#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013
#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014
#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015
#define AFE_PORT_ID_QUINARY_MI2S_RX 0x1016
#define AFE_PORT_ID_QUINARY_MI2S_TX 0x1017
/* ID of the senary MI2S Rx port. */
#define AFE_PORT_ID_SENARY_MI2S_RX 0x1018
/* ID of the senary MI2S Tx port. */
#define AFE_PORT_ID_SENARY_MI2S_TX 0x1019
/* ID of the Internal 0 MI2S Rx port */
#define AFE_PORT_ID_INT0_MI2S_RX 0x102E
/* ID of the Internal 0 MI2S Tx port */
#define AFE_PORT_ID_INT0_MI2S_TX 0x102F
/* ID of the Internal 1 MI2S Rx port */
#define AFE_PORT_ID_INT1_MI2S_RX 0x1030
/* ID of the Internal 1 MI2S Tx port */
#define AFE_PORT_ID_INT1_MI2S_TX 0x1031
/* ID of the Internal 2 MI2S Rx port */
#define AFE_PORT_ID_INT2_MI2S_RX 0x1032
/* ID of the Internal 2 MI2S Tx port */
#define AFE_PORT_ID_INT2_MI2S_TX 0x1033
/* ID of the Internal 3 MI2S Rx port */
#define AFE_PORT_ID_INT3_MI2S_RX 0x1034
/* ID of the Internal 3 MI2S Tx port */
#define AFE_PORT_ID_INT3_MI2S_TX 0x1035
/* ID of the Internal 4 MI2S Rx port */
#define AFE_PORT_ID_INT4_MI2S_RX 0x1036
/* ID of the Internal 4 MI2S Tx port */
#define AFE_PORT_ID_INT4_MI2S_TX 0x1037
/* ID of the Internal 5 MI2S Rx port */
#define AFE_PORT_ID_INT5_MI2S_RX 0x1038
/* ID of the Internal 5 MI2S Tx port */
#define AFE_PORT_ID_INT5_MI2S_TX 0x1039
/* ID of the Internal 6 MI2S Rx port */
#define AFE_PORT_ID_INT6_MI2S_RX 0x103A
/* ID of the Internal 6 MI2S Tx port */
#define AFE_PORT_ID_INT6_MI2S_TX 0x103B
#define AFE_PORT_ID_SPDIF_RX 0x5000
#define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000
#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001
#define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000
#define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001
#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002
#define AFE_PORT_ID_INTERNAL_FM_RX 0x3004
#define AFE_PORT_ID_INTERNAL_FM_TX 0x3005
/* SLIMbus Rx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
/* SLIMbus Tx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
/* SLIMbus Rx port on channel 1. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
/* SLIMbus Tx port on channel 1. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
/* SLIMbus Rx port on channel 2. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
/* SLIMbus Tx port on channel 2. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
/* SLIMbus Rx port on channel 3. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
/* SLIMbus Tx port on channel 3. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
/* SLIMbus Rx port on channel 4. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
/* SLIMbus Tx port on channel 4. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
/* SLIMbus Rx port on channel 5. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX 0x400a
/* SLIMbus Tx port on channel 5. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX 0x400b
/* SLIMbus Rx port on channel 6. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
/* SLIMbus Tx port on channel 6. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
/* SLIMbus Rx port on channel 7. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_RX 0x400e
/* SLIMbus Tx port on channel 7. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_TX 0x400f
/* SLIMbus Rx port on channel 8. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_RX 0x4010
/* SLIMbus Tx port on channel 8. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_TX 0x4011
/* AFE Rx port for audio over Display port */
#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020
/*USB AFE port */
#define AFE_PORT_ID_USB_RX 0x7000
#define AFE_PORT_ID_USB_TX 0x7001
/* Generic pseudoport 1. */
#define AFE_PORT_ID_PSEUDOPORT_01 0x8001
/* Generic pseudoport 2. */
#define AFE_PORT_ID_PSEUDOPORT_02 0x8002
/* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx}
* Primary Aux PCM Tx port ID.
*/
#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
/* Pseudoport that corresponds to the voice Rx path.
* For recording, the voice Rx path samples are written to this
* port and consumed by the audio path.
*/
#define AFE_PORT_ID_VOICE_RECORD_RX 0x8003
/* Pseudoport that corresponds to the voice Tx path.
* For recording, the voice Tx path samples are written to this
* port and consumed by the audio path.
*/
#define AFE_PORT_ID_VOICE_RECORD_TX 0x8004
/* Pseudoport that corresponds to in-call voice delivery samples.
* During in-call audio delivery, the audio path delivers samples
* to this port from where the voice path delivers them on the
* Rx path.
*/
#define AFE_PORT_ID_VOICE2_PLAYBACK_TX 0x8002
#define AFE_PORT_ID_VOICE_PLAYBACK_TX 0x8005
#define AFE_PORT_ID_PRIMARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x00)
#define AFE_PORT_ID_PRIMARY_TDM_RX_1 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x02)
#define AFE_PORT_ID_PRIMARY_TDM_RX_2 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x04)
#define AFE_PORT_ID_PRIMARY_TDM_RX_3 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x06)
#define AFE_PORT_ID_PRIMARY_TDM_RX_4 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x08)
#define AFE_PORT_ID_PRIMARY_TDM_RX_5 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_PRIMARY_TDM_RX_6 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_PRIMARY_TDM_RX_7 \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_PRIMARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x01)
#define AFE_PORT_ID_PRIMARY_TDM_TX_1 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x02)
#define AFE_PORT_ID_PRIMARY_TDM_TX_2 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x04)
#define AFE_PORT_ID_PRIMARY_TDM_TX_3 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x06)
#define AFE_PORT_ID_PRIMARY_TDM_TX_4 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x08)
#define AFE_PORT_ID_PRIMARY_TDM_TX_5 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_PRIMARY_TDM_TX_6 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_PRIMARY_TDM_TX_7 \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x0E)
#define AFE_PORT_ID_SECONDARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x10)
#define AFE_PORT_ID_SECONDARY_TDM_RX_1 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x02)
#define AFE_PORT_ID_SECONDARY_TDM_RX_2 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x04)
#define AFE_PORT_ID_SECONDARY_TDM_RX_3 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x06)
#define AFE_PORT_ID_SECONDARY_TDM_RX_4 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x08)
#define AFE_PORT_ID_SECONDARY_TDM_RX_5 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_SECONDARY_TDM_RX_6 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_SECONDARY_TDM_RX_7 \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_SECONDARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x11)
#define AFE_PORT_ID_SECONDARY_TDM_TX_1 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x02)
#define AFE_PORT_ID_SECONDARY_TDM_TX_2 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x04)
#define AFE_PORT_ID_SECONDARY_TDM_TX_3 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x06)
#define AFE_PORT_ID_SECONDARY_TDM_TX_4 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x08)
#define AFE_PORT_ID_SECONDARY_TDM_TX_5 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_SECONDARY_TDM_TX_6 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_SECONDARY_TDM_TX_7 \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x0E)
#define AFE_PORT_ID_TERTIARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x20)
#define AFE_PORT_ID_TERTIARY_TDM_RX_1 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x02)
#define AFE_PORT_ID_TERTIARY_TDM_RX_2 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x04)
#define AFE_PORT_ID_TERTIARY_TDM_RX_3 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x06)
#define AFE_PORT_ID_TERTIARY_TDM_RX_4 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x08)
#define AFE_PORT_ID_TERTIARY_TDM_RX_5 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_TERTIARY_TDM_RX_6 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_TERTIARY_TDM_RX_7 \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_TERTIARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x21)
#define AFE_PORT_ID_TERTIARY_TDM_TX_1 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x02)
#define AFE_PORT_ID_TERTIARY_TDM_TX_2 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x04)
#define AFE_PORT_ID_TERTIARY_TDM_TX_3 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x06)
#define AFE_PORT_ID_TERTIARY_TDM_TX_4 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x08)
#define AFE_PORT_ID_TERTIARY_TDM_TX_5 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_TERTIARY_TDM_TX_6 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_TERTIARY_TDM_TX_7 \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x0E)
#define AFE_PORT_ID_QUATERNARY_TDM_RX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x30)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_1 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x02)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_2 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x04)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_3 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x06)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_4 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x08)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_5 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0A)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_6 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0C)
#define AFE_PORT_ID_QUATERNARY_TDM_RX_7 \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0E)
#define AFE_PORT_ID_QUATERNARY_TDM_TX \
(AFE_PORT_ID_TDM_PORT_RANGE_START + 0x31)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_1 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x02)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_2 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x04)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_3 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x06)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_4 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x08)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_5 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0A)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_6 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0C)
#define AFE_PORT_ID_QUATERNARY_TDM_TX_7 \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0E)
#define AFE_PORT_ID_INVALID 0xFFFF
#define AAC_ENC_MODE_AAC_LC 0x02
#define AAC_ENC_MODE_AAC_P 0x05
#define AAC_ENC_MODE_EAAC_P 0x1D
#define AFE_PSEUDOPORT_CMD_START 0x000100cf
struct afe_pseudoport_start_command {
struct apr_hdr hdr;
u16 port_id; /* Pseudo Port 1 = 0x8000 */
/* Pseudo Port 2 = 0x8001 */
/* Pseudo Port 3 = 0x8002 */
u16 timing; /* FTRT = 0 , AVTimer = 1, */
} __packed;
#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
struct afe_pseudoport_stop_command {
struct apr_hdr hdr;
u16 port_id; /* Pseudo Port 1 = 0x8000 */
/* Pseudo Port 2 = 0x8001 */
/* Pseudo Port 3 = 0x8002 */
u16 reserved;
} __packed;
#define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202
#define AFE_PARAM_ID_ENABLE 0x00010203
/* Payload of the #AFE_PARAM_ID_ENABLE
* parameter, which enables or
* disables any module.
* The fixed size of this structure is four bytes.
*/
struct afe_mod_enable_param {
u16 enable;
/* Enables (1) or disables (0) the module. */
u16 reserved;
/* This field must be set to zero. */
} __packed;
/* ID of the configuration parameter used by the
* #AFE_MODULE_SIDETONE_IIR_FILTER module.
*/
#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204
#define MAX_SIDETONE_IIR_DATA_SIZE 224
#define MAX_NO_IIR_FILTER_STAGE 10
struct afe_sidetone_iir_filter_config_params {
u16 num_biquad_stages;
/* Number of stages.
* Supported values: Minimum of 5 and maximum of 10
*/
u16 pregain;
/* Pregain for the compensating filter response.
* Supported values: Any number in Q13 format
*/
uint8_t iir_config[MAX_SIDETONE_IIR_DATA_SIZE];
} __packed;
#define AFE_MODULE_LOOPBACK 0x00010205
#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206
/* 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 which can be
* Tx or Rx port. In addition, we can configure the type of
* routing mode to handle different use cases.
*/
#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
enum afe_loopback_routing_mode {
LB_MODE_DEFAULT = 1,
/* Regular loopback from source to destination port */
LB_MODE_SIDETONE,
/* Sidetone feed from Tx source to Rx destination port */
LB_MODE_EC_REF_VOICE_AUDIO,
/* Echo canceller reference, voice + audio + DTMF */
LB_MODE_EC_REF_VOICE
/* Echo canceller reference, voice alone */
} __packed;
/* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG ,
* which enables/disables one AFE loopback.
*/
struct afe_loopback_cfg_v1 {
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;
struct afe_loopback_sidetone_gain {
u16 rx_port_id;
u16 gain;
} __packed;
struct loopback_cfg_data {
u32 loopback_cfg_minor_version;
/* Minor version used for tracking the version of the RMC module
* configuration interface.
* Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
*/
u16 dst_port_id;
/* Destination Port Id. */
u16 routing_mode;
/* Specifies data path type from src to dest port.
* Supported values:
* #LB_MODE_DEFAULT
* #LB_MODE_SIDETONE
* #LB_MODE_EC_REF_VOICE_AUDIO
* #LB_MODE_EC_REF_VOICE_A
* #LB_MODE_EC_REF_VOICE
*/
u16 enable;
/* Specifies whether to enable (1) or
* disable (0) an AFE loopback.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.
*/
} __packed;
struct afe_st_loopback_cfg_v1 {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 gain_pdata;
struct afe_loopback_sidetone_gain gain_data;
struct afe_port_param_data_v2 cfg_pdata;
struct loopback_cfg_data cfg_data;
} __packed;
struct afe_loopback_iir_cfg_v2 {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 st_iir_enable_pdata;
struct afe_mod_enable_param st_iir_mode_enable_data;
struct afe_port_param_data_v2 st_iir_filter_config_pdata;
struct afe_sidetone_iir_filter_config_params st_iir_filter_config_data;
} __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 following parameter ids.
* - #AFE_PARAM_ID_I2S_CONFIG
* - #AFE_PARAM_ID_PCM_CONFIG
* - #AFE_PARAM_ID_DIGI_MIC_CONFIG
* - #AFE_PARAM_ID_HDMI_CONFIG
* - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
* - #AFE_PARAM_ID_SLIMBUS_CONFIG
* - #AFE_PARAM_ID_RT_PROXY_CONFIG
*/
#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
#define AFE_PORT_SAMPLE_RATE_8K 8000
#define AFE_PORT_SAMPLE_RATE_16K 16000
#define AFE_PORT_SAMPLE_RATE_48K 48000
#define AFE_PORT_SAMPLE_RATE_96K 96000
#define AFE_PORT_SAMPLE_RATE_176P4K 176400
#define AFE_PORT_SAMPLE_RATE_192K 192000
#define AFE_PORT_SAMPLE_RATE_352P8K 352800
#define AFE_LINEAR_PCM_DATA 0x0
#define AFE_NON_LINEAR_DATA 0x1
#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2
#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3
#define AFE_GENERIC_COMPRESSED 0x8
/* This param id is used to configure I2S interface */
#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
#define AFE_API_VERSION_I2S_CONFIG 0x1
/* Enumeration for setting the I2S configuration
* channel_mode parameter to
* serial data wire number 1-3 (SD3).
*/
#define AFE_PORT_I2S_SD0 0x1
#define AFE_PORT_I2S_SD1 0x2
#define AFE_PORT_I2S_SD2 0x3
#define AFE_PORT_I2S_SD3 0x4
#define AFE_PORT_I2S_QUAD01 0x5
#define AFE_PORT_I2S_QUAD23 0x6
#define AFE_PORT_I2S_6CHS 0x7
#define AFE_PORT_I2S_8CHS 0x8
#define AFE_PORT_I2S_MONO 0x0
#define AFE_PORT_I2S_STEREO 0x1
#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
/* Payload of the #AFE_PARAM_ID_I2S_CONFIG
* command's (I2S configuration
* parameter).
*/
struct afe_param_id_i2s_cfg {
u32 i2s_cfg_minor_version;
/* Minor version used for tracking the version of the I2S
* configuration interface.
* Supported values: #AFE_API_VERSION_I2S_CONFIG
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 channel_mode;
/* I2S lines and multichannel operation.
* Supported values:
* - #AFE_PORT_I2S_SD0
* - #AFE_PORT_I2S_SD1
* - #AFE_PORT_I2S_SD2
* - #AFE_PORT_I2S_SD3
* - #AFE_PORT_I2S_QUAD01
* - #AFE_PORT_I2S_QUAD23
* - #AFE_PORT_I2S_6CHS
* - #AFE_PORT_I2S_8CHS
*/
u16 mono_stereo;
/* Specifies mono or stereo. This applies only when
* a single I2S line is used.
* Supported values:
* - #AFE_PORT_I2S_MONO
* - #AFE_PORT_I2S_STEREO
*/
u16 ws_src;
/* Word select source: internal or external.
* Supported values:
* - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL
* - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
u16 data_format;
/* data format
* Supported values:
* - #LINEAR_PCM_DATA
* - #NON_LINEAR_DATA
* - #LINEAR_PCM_DATA_PACKED_IN_60958
* - #NON_LINEAR_DATA_PACKED_IN_60958
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
/*
* This param id is used to configure PCM interface
*/
#define AFE_API_VERSION_SPDIF_CONFIG 0x1
#define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1
#define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1
#define AFE_CH_STATUS_A 1
#define AFE_CH_STATUS_B 2
#define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244
#define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245
#define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246
#define AFE_PORT_CLK_ROOT_LPAPLL 0x3
#define AFE_PORT_CLK_ROOT_LPAQ6PLL 0x4
struct afe_param_id_spdif_cfg {
/* Minor version used for tracking the version of the SPDIF
* configuration interface.
* Supported values: #AFE_API_VERSION_SPDIF_CONFIG
*/
u32 spdif_cfg_minor_version;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_22_05K
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_44_1K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_176_4K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
u32 sample_rate;
/* data format
* Supported values:
* - #AFE_LINEAR_PCM_DATA
* - #AFE_NON_LINEAR_DATA
*/
u16 data_format;
/* Number of channels supported by the port
* - PCM - 1, Compressed Case - 2
*/
u16 num_channels;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 bit_width;
/* This field must be set to zero. */
u16 reserved;
} __packed;
struct afe_param_id_spdif_ch_status_cfg {
u32 ch_status_cfg_minor_version;
/* Minor version used for tracking the version of channel
* status configuration. Current supported version is 1
*/
u32 status_type;
/* Indicate if the channel status is for channel A or B
* Supported values:
* - #AFE_CH_STATUS_A
* - #AFE_CH_STATUS_B
*/
u8 status_bits[24];
/* Channel status - 192 bits for channel
* Byte ordering as defined by IEC60958-3
*/
u8 status_mask[24];
/* Channel status with mask bits 1 will be applied.
* Byte ordering as defined by IEC60958-3
*/
} __packed;
struct afe_param_id_spdif_clk_cfg {
u32 clk_cfg_minor_version;
/* Minor version used for tracking the version of SPDIF
* interface clock configuration. Current supported version
* is 1
*/
u32 clk_value;
/* Specifies the clock frequency in Hz to set
* Supported values:
* 0 - Disable the clock
* 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2
* (channels A and B)
*/
u32 clk_root;
/* Specifies SPDIF root clk source
* Supported Values:
* - #AFE_PORT_CLK_ROOT_LPAPLL
* - #AFE_PORT_CLK_ROOT_LPAQ6PLL
*/
} __packed;
struct afe_spdif_clk_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_spdif_clk_cfg clk_cfg;
} __packed;
struct afe_spdif_chstatus_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_spdif_ch_status_cfg ch_status;
} __packed;
struct afe_spdif_port_config {
struct afe_param_id_spdif_cfg cfg;
struct afe_param_id_spdif_ch_status_cfg ch_status;
} __packed;
#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E
#define AFE_API_VERSION_PCM_CONFIG 0x1
/* Enumeration for the auxiliary PCM synchronization signal
* provided by an external source.
*/
#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0
/* Enumeration for the auxiliary PCM synchronization signal
* provided by an internal source.
*/
#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1
/* Enumeration for the PCM configuration aux_mode parameter,
* which configures the auxiliary PCM interface to use
* short synchronization.
*/
#define AFE_PORT_PCM_AUX_MODE_PCM 0x0
/*
* Enumeration for the PCM configuration aux_mode parameter,
* which configures the auxiliary PCM interface to use long
* synchronization.
*/
#define AFE_PORT_PCM_AUX_MODE_AUX 0x1
/*
* Enumeration for setting the PCM configuration frame to 8.
*/
#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0
/*
* Enumeration for setting the PCM configuration frame to 16.
*/
#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1
/* Enumeration for setting the PCM configuration frame to 32.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2
/* Enumeration for setting the PCM configuration frame to 64.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3
/* Enumeration for setting the PCM configuration frame to 128.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4
/* Enumeration for setting the PCM configuration frame to 256.*/
#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5
/* Enumeration for setting the PCM configuration
* quantype parameter to A-law with no padding.
*/
#define AFE_PORT_PCM_ALAW_NOPADDING 0x0
/* Enumeration for setting the PCM configuration quantype
* parameter to mu-law with no padding.
*/
#define AFE_PORT_PCM_MULAW_NOPADDING 0x1
/* Enumeration for setting the PCM configuration quantype
* parameter to linear with no padding.
*/
#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2
/* Enumeration for setting the PCM configuration quantype
* parameter to A-law with padding.
*/
#define AFE_PORT_PCM_ALAW_PADDING 0x3
/* Enumeration for setting the PCM configuration quantype
* parameter to mu-law with padding.
*/
#define AFE_PORT_PCM_MULAW_PADDING 0x4
/* Enumeration for setting the PCM configuration quantype
* parameter to linear with padding.
*/
#define AFE_PORT_PCM_LINEAR_PADDING 0x5
/* Enumeration for disabling the PCM configuration
* ctrl_data_out_enable parameter.
* The PCM block is the only master.
*/
#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0
/*
* Enumeration for enabling the PCM configuration
* ctrl_data_out_enable parameter. The PCM block shares
* the signal with other masters.
*/
#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1
/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
* (PCM configuration parameter).
*/
struct afe_param_id_pcm_cfg {
u32 pcm_cfg_minor_version;
/* Minor version used for tracking the version of the AUX PCM
* configuration interface.
* Supported values: #AFE_API_VERSION_PCM_CONFIG
*/
u16 aux_mode;
/* PCM synchronization setting.
* Supported values:
* - #AFE_PORT_PCM_AUX_MODE_PCM
* - #AFE_PORT_PCM_AUX_MODE_AUX
*/
u16 sync_src;
/* Synchronization source.
* Supported values:
* - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL
* - #AFE_PORT_PCM_SYNC_SRC_INTERNAL
*/
u16 frame_setting;
/* Number of bits per frame.
* Supported values:
* - #AFE_PORT_PCM_BITS_PER_FRAME_8
* - #AFE_PORT_PCM_BITS_PER_FRAME_16
* - #AFE_PORT_PCM_BITS_PER_FRAME_32
* - #AFE_PORT_PCM_BITS_PER_FRAME_64
* - #AFE_PORT_PCM_BITS_PER_FRAME_128
* - #AFE_PORT_PCM_BITS_PER_FRAME_256
*/
u16 quantype;
/* PCM quantization type.
* Supported values:
* - #AFE_PORT_PCM_ALAW_NOPADDING
* - #AFE_PORT_PCM_MULAW_NOPADDING
* - #AFE_PORT_PCM_LINEAR_NOPADDING
* - #AFE_PORT_PCM_ALAW_PADDING
* - #AFE_PORT_PCM_MULAW_PADDING
* - #AFE_PORT_PCM_LINEAR_PADDING
*/
u16 ctrl_data_out_enable;
/* Specifies whether the PCM block shares the data-out
* signal to the drive with other masters.
* Supported values:
* - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE
* - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE
*/
u16 reserved;
/* This field must be set to zero. */
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to 4
*/
u16 slot_number_mapping[4];
/* Specifies the slot number for the each channel in
* multi channel scenario.
* Supported values: 1 to 32
*/
} __packed;
/*
* This param id is used to configure DIGI MIC interface
*/
#define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to left 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1
/*Enumeration for setting the digital mic configuration
* channel_mode parameter to right 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to left 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to right 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to stereo 0.
*/
#define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to stereo 1.
*/
#define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6
/* Enumeration for setting the digital mic configuration
* channel_mode parameter to quad.
*/
#define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7
/* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's
* (DIGI MIC configuration
* parameter).
*/
struct afe_param_id_digi_mic_cfg {
u32 digi_mic_cfg_minor_version;
/* Minor version used for tracking the version of the DIGI Mic
* configuration interface.
* Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 channel_mode;
/* Digital mic and multichannel operation.
* Supported values:
* - #AFE_PORT_DIGI_MIC_MODE_LEFT0
* - #AFE_PORT_DIGI_MIC_MODE_RIGHT0
* - #AFE_PORT_DIGI_MIC_MODE_LEFT1
* - #AFE_PORT_DIGI_MIC_MODE_RIGHT1
* - #AFE_PORT_DIGI_MIC_MODE_STEREO0
* - #AFE_PORT_DIGI_MIC_MODE_STEREO1
* - #AFE_PORT_DIGI_MIC_MODE_QUAD
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
*/
} __packed;
/* This param id is used to configure HDMI interface */
#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_HDMI_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command,
* which configures a multichannel HDMI audio interface.
*/
struct afe_param_id_hdmi_multi_chan_audio_cfg {
u32 hdmi_cfg_minor_version;
/* Minor version used for tracking the version of the HDMI
* configuration interface.
* Supported values: #AFE_API_VERSION_HDMI_CONFIG
*/
u16 datatype;
/* data type
* Supported values:
* - #LINEAR_PCM_DATA
* - #NON_LINEAR_DATA
* - #LINEAR_PCM_DATA_PACKED_IN_60958
* - #NON_LINEAR_DATA_PACKED_IN_60958
*/
u16 channel_allocation;
/* HDMI channel allocation information for programming an HDMI
* frame. The default is 0 (Stereo).
*
* This information is defined in the HDMI standard, CEA 861-D
* (refer to @xhyperref{S1,[S1]}). The number of channels is also
* inferred from this parameter.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - 22050, 44100, 176400 for compressed streams
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 reserved;
/* This field must be set to zero. */
} __packed;
/* This param id is used to configure BT or FM(RIVA) interface */
#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
* command's BT voice/BT audio/FM configuration parameter.
*/
struct afe_param_id_internal_bt_fm_cfg {
u32 bt_fm_cfg_minor_version;
/* Minor version used for tracking the version of the BT and FM
* configuration interface.
* Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to 2
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO)
* - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO)
* - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP)
*/
} __packed;
/* This param id is used to configure SLIMBUS interface using
* shared channel approach.
*/
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
/* Enumeration for setting SLIMbus device ID 1. */
#define AFE_SLIMBUS_DEVICE_1 0x0
/* Enumeration for setting SLIMbus device ID 2. */
#define AFE_SLIMBUS_DEVICE_2 0x1
/* Enumeration for setting the SLIMbus data formats. */
#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0
/* Enumeration for setting the maximum number of streams per
* device.
*/
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus
* port configuration parameter.
*/
struct afe_param_id_slimbus_cfg {
u32 sb_cfg_minor_version;
/* Minor version used for tracking the version of the SLIMBUS
* configuration interface.
* Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
*/
u16 slimbus_dev_id;
/* SLIMbus hardware device ID, which is required to handle
* multiple SLIMbus hardware blocks.
* Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 data_format;
/* Data format supported by the SLIMbus hardware. The default is
* 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
* hardware does not perform any format conversions before the data
* transfer.
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
/* Mapping of shared channel IDs (128 to 255) to which the
* master port is to be connected.
* Shared_channel_mapping[i] represents the shared channel assigned
* for audio channel i in multichannel audio data.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_192K
*/
} __packed;
/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS to configure
* USB audio device parameter. It should be used with
* AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5
/* ID of the parameter used to set the endianness value for the
* USB audio device. It should be used with
* AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA
/* Minor version used for tracking USB audio configuration */
#define AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG 0x1
/* Payload of the AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_usb_audio_dev_params {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Token of actual end USB aduio device */
u32 dev_token;
} __packed;
struct afe_param_id_usb_audio_dev_lpcm_fmt {
/* Minor version used for tracking USB audio device parameter.
* Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Endianness of actual end USB audio device */
u32 endian;
} __packed;
/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_CONFIG to configure
* USB audio interface. It should be used with AFE_MODULE_AUDIO_DEV_INTERFACE
*/
#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4
/* Payload of the AFE_PARAM_ID_USB_AUDIO_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_usb_audio_cfg {
/* Minor version used for tracking USB audio device configuration.
* Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
*/
u32 cfg_minor_version;
/* Sampling rate of the port.
* Supported values:
* - AFE_PORT_SAMPLE_RATE_8K
* - AFE_PORT_SAMPLE_RATE_11025
* - AFE_PORT_SAMPLE_RATE_12K
* - AFE_PORT_SAMPLE_RATE_16K
* - AFE_PORT_SAMPLE_RATE_22050
* - AFE_PORT_SAMPLE_RATE_24K
* - AFE_PORT_SAMPLE_RATE_32K
* - AFE_PORT_SAMPLE_RATE_44P1K
* - AFE_PORT_SAMPLE_RATE_48K
* - AFE_PORT_SAMPLE_RATE_96K
* - AFE_PORT_SAMPLE_RATE_192K
*/
u32 sample_rate;
/* Bit width of the sample.
* Supported values: 16, 24
*/
u16 bit_width;
/* Number of channels.
* Supported values: 1 and 2
*/
u16 num_channels;
/* Data format supported by the USB. The supported value is
* 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM).
*/
u16 data_format;
/* this field must be 0 */
u16 reserved;
/* device token of actual end USB aduio device */
u32 dev_token;
/* endianness of this interface */
u32 endian;
} __packed;
struct afe_usb_audio_dev_param_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
union {
struct afe_param_id_usb_audio_dev_params usb_dev;
struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt;
};
} __packed;
/* This param id is used to configure Real Time Proxy interface. */
#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213
/* This version information is used to handle the new
* additions to the config interface in future in backward
* compatible manner.
*/
#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1
/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG
* command (real-time proxy port configuration parameter).
*/
struct afe_param_id_rt_proxy_port_cfg {
u32 rt_proxy_cfg_minor_version;
/* Minor version used for tracking the version of rt-proxy
* config interface.
*/
u16 bit_width;
/* Bit width of the sample.
* Supported values: 16
*/
u16 interleaved;
/* Specifies whether the data exchanged between the AFE
* interface and real-time port is interleaved.
* Supported values: - 0 -- Non-interleaved (samples from each
* channel are contiguous in the buffer) - 1 -- Interleaved
* (corresponding samples from each input channel are interleaved
* within the buffer)
*/
u16 frame_size;
/* Size of the frames that are used for PCM exchanges with this
* port.
* Supported values: > 0, in bytes
* For example, 5 ms buffers of 16 bits and 16 kHz stereo samples
* is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320
* bytes.
*/
u16 jitter_allowance;
/* Configures the amount of jitter that the port will allow.
* Supported values: > 0
* For example, if +/-10 ms of jitter is anticipated in the timing
* of sending frames to the port, and the configuration is 16 kHz
* mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2
* bytes/sample = 320.
*/
u16 low_water_mark;
/* Low watermark in bytes (including all channels).
* Supported values:
* - 0 -- Do not send any low watermark events
* - > 0 -- Low watermark for triggering an event
* If the number of bytes in an internal circular buffer is lower
* than this low_water_mark parameter, a LOW_WATER_MARK event is
* sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
* event).
* Use of watermark events is optional for debugging purposes.
*/
u16 high_water_mark;
/* High watermark in bytes (including all channels).
* Supported values:
* - 0 -- Do not send any high watermark events
* - > 0 -- High watermark for triggering an event
* If the number of bytes in an internal circular buffer exceeds
* TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event
* is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
* event).
* The use of watermark events is optional and for debugging
* purposes.
*/
u32 sample_rate;
/* Sampling rate of the port.
* Supported values:
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_48K
*/
u16 num_channels;
/* Number of channels.
* Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
u16 reserved;
/* For 32 bit alignment. */
} __packed;
/* This param id is used to configure the Pseudoport interface */
#define AFE_PARAM_ID_PSEUDO_PORT_CONFIG 0x00010219
/* Version information used to handle future additions to the configuration
* interface (for backward compatibility).
*/
#define AFE_API_VERSION_PSEUDO_PORT_CONFIG 0x1
/* Enumeration for setting the timing_mode parameter to faster than real
* time.
*/
#define AFE_PSEUDOPORT_TIMING_MODE_FTRT 0x0
/* Enumeration for setting the timing_mode parameter to real time using
* timers.
*/
#define AFE_PSEUDOPORT_TIMING_MODE_TIMER 0x1
/* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_pseudo_port_cfg {
u32 pseud_port_cfg_minor_version;
/*
* Minor version used for tracking the version of the pseudoport
* configuration interface.
*/
u16 bit_width;
/* Bit width of the sample at values 16, 24 */
u16 num_channels;
/* Number of channels at values 1 to 8 */
u16 data_format;
/* Non-linear data format supported by the pseudoport (for future use).
* At values #AFE_LINEAR_PCM_DATA
*/
u16 timing_mode;
/* Indicates whether the pseudoport synchronizes to the clock or
* operates faster than real time.
* at values
* - #AFE_PSEUDOPORT_TIMING_MODE_FTRT
* - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend
*/
u32 sample_rate;
/* Sample rate at which the pseudoport will run.
* at values
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_96K
* - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend
*/
} __packed;
#define AFE_PARAM_ID_TDM_CONFIG 0x0001029D
#define AFE_API_VERSION_TDM_CONFIG 1
#define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE 0
#define AFE_PORT_TDM_LONG_SYNC_MODE 1
#define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE 2
#define AFE_PORT_TDM_SYNC_SRC_EXTERNAL 0
#define AFE_PORT_TDM_SYNC_SRC_INTERNAL 1
#define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE 0
#define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE 1
#define AFE_PORT_TDM_SYNC_NORMAL 0
#define AFE_PORT_TDM_SYNC_INVERT 1
#define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE 0
#define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE 1
#define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE 2
/* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_tdm_cfg {
u32 tdm_cfg_minor_version;
/* < Minor version used to track TDM configuration.
* @values #AFE_API_VERSION_TDM_CONFIG
*/
u32 num_channels;
/* < Number of enabled slots for TDM frame.
* @values 1 to 8
*/
u32 sample_rate;
/* < Sampling rate of the port.
* @values
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_24K
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_48K
* - #AFE_PORT_SAMPLE_RATE_176P4K
* - #AFE_PORT_SAMPLE_RATE_352P8K @tablebulletend
*/
u32 bit_width;
/* < Bit width of the sample.
* @values 16, 24
*/
u16 data_format;
/* < Data format: linear ,compressed, generic compresssed
* @values
* - #AFE_LINEAR_PCM_DATA
* - #AFE_NON_LINEAR_DATA
* - #AFE_GENERIC_COMPRESSED
*/
u16 sync_mode;
/* < TDM synchronization setting.
* @values (short, long, slot) sync mode
* - #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE
* - #AFE_PORT_TDM_LONG_SYNC_MODE
* - #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend
*/
u16 sync_src;
/* < Synchronization source.
* @values
* - #AFE_PORT_TDM_SYNC_SRC_EXTERNAL
* - #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend
*/
u16 nslots_per_frame;
/* < Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
* @values 1 - 32
*/
u16 ctrl_data_out_enable;
/* < Specifies whether the TDM block shares the data-out signal to the
* drive with other masters.
* @values
* - #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE
* - #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend
*/
u16 ctrl_invert_sync_pulse;
/* < Specifies whether to invert the sync or not.
* @values
* - #AFE_PORT_TDM_SYNC_NORMAL
* - #AFE_PORT_TDM_SYNC_INVERT @tablebulletend
*/
u16 ctrl_sync_data_delay;
/* < Specifies the number of bit clock to delay data with respect to
* sync edge.
* @values
* - #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE
* - #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE
* - #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend
*/
u16 slot_width;
/* < Slot width of the slot in a TDM frame. (slot_width >= bit_width)
* have to be satisfied.
* @values 16, 24, 32
*/
u32 slot_mask;
/* < Position of active slots. When that bit is set,
* that paricular slot is active.
* Number of active slots can be inferred by number of
* bits set in the mask. Only 8 individual bits can be enabled.
* Bits 0..31 corresponding to slot 0..31
* @values 1 to 2^32 - 1
*/
} __packed;
/* ID of Time Divsion Multiplexing (TDM) module,
* which is used for configuring the AFE TDM.
*
* This module supports following parameter IDs:
* - #AFE_PORT_TDM_SLOT_CONFIG
*
* To configure the TDM interface, the client must use the
* #AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the
* respective parameter IDs as listed above.
*/
#define AFE_MODULE_TDM 0x0001028A
/* ID of the parameter used by #AFE_MODULE_TDM to configure
* the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
*/
#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297
/* Version information used to handle future additions to slot mapping
* configuration (for backward compatibility).
*/
#define AFE_API_VERSION_SLOT_MAPPING_CONFIG 0x1
/* Data align type */
#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB 0
#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB 1
#define AFE_SLOT_MAPPING_OFFSET_INVALID 0xFFFF
/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG
* command's TDM configuration parameter.
*/
struct afe_param_id_slot_mapping_cfg {
u32 minor_version;
/* < Minor version used for tracking TDM slot configuration.
* @values #AFE_API_VERSION_TDM_SLOT_CONFIG
*/
u16 num_channel;
/* < number of channel of the audio sample.
* @values 1, 2, 4, 6, 8 @tablebulletend
*/
u16 bitwidth;
/* < Slot bit width for each channel
* @values 16, 24, 32
*/
u32 data_align_type;
/* < indicate how data packed from slot_offset for 32 slot bit width
* in case of sample bit width is 24.
* @values
* #AFE_SLOT_MAPPING_DATA_ALIGN_MSB
* #AFE_SLOT_MAPPING_DATA_ALIGN_LSB
*/
u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT];
/* < Array of the slot mapping start offset in bytes for this frame.
* The bytes is counted from 0. The 0 is mapped to the 1st byte
* in or out of the digital serial data line this sub-frame belong to.
* slot_offset[] setting is per-channel based.
* The max num of channel supported is 8.
* The valid offset value must always be continuly placed in from
* index 0.
* Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays.
* If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24,
* "data_align_type" is used to indicate how 24 bit sample data in
* aligning with 32 bit slot width per-channel.
* @values, in byte
*/
} __packed;
/* ID of the parameter used by #AFE_MODULE_TDM to configure
* the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
*/
#define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG 0x00010298
/* Version information used to handle future additions to custom TDM header
* configuration (for backward compatibility).
*/
#define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG 0x1
#define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID 0x0
#define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT 0x1
#define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST 0x2
#define AFE_CUSTOM_TDM_HEADER_MAX_CNT 0x8
/* Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID */
struct afe_param_id_custom_tdm_header_cfg {
u32 minor_version;
/* < Minor version used for tracking custom TDM header configuration.
* @values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG
*/
u16 start_offset;
/* < the slot mapping start offset in bytes from this sub-frame
* The bytes is counted from 0. The 0 is mapped to the 1st byte in or
* out of the digital serial data line this sub-frame belong to.
* @values, in byte,
* supported values are 0, 4, 8
*/
u16 header_width;
/* < the header width per-frame followed.
* 2 bytes for MOST/TDM case
* @values, in byte
* supported value is 2
*/
u16 header_type;
/* < Indicate what kind of custom TDM header it is.
* @values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0
* #AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1 (for AAN channel per MOST)
* #AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2
* (for entertainment channel, which will overwrite
* AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST)
*/
u16 num_frame_repeat;
/* < num of header followed.
* @values, supported value is 8
*/
u16 header[AFE_CUSTOM_TDM_HEADER_MAX_CNT];
/* < SAD header for MOST/TDM case is followed as payload as below.
* The size of followed SAD header in bytes is num_of_frame_repeat *
* header_width_per_frame, which is 2 * 8 = 16 bytes here.
* the supported payload format is in uint16_t as below
* uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00
* uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218
* uint16_t header2; DTCP Info Info[6] - unused -> 0x0
* uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0
* uint16_t header4; Reserved Info[0] - Num of bytes following -> 0x7
* uint16_t header5; Reserved Info[1] - Media Type -> 0x0
* uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0
* uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low Byte -> 0x0
*/
} __packed;
struct afe_slot_mapping_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_slot_mapping_cfg slot_mapping;
} __packed;
struct afe_custom_tdm_header_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_custom_tdm_header_cfg custom_tdm_header;
} __packed;
struct afe_tdm_port_config {
struct afe_param_id_tdm_cfg tdm;
struct afe_param_id_slot_mapping_cfg slot_mapping;
struct afe_param_id_custom_tdm_header_cfg custom_tdm_header;
} __packed;
#define AFE_PARAM_ID_DEVICE_HW_DELAY 0x00010243
#define AFE_API_VERSION_DEVICE_HW_DELAY 0x1
struct afe_param_id_device_hw_delay_cfg {
uint32_t device_hw_delay_minor_version;
uint32_t delay_in_us;
} __packed;
#define AFE_PARAM_ID_SET_TOPOLOGY 0x0001025A
#define AFE_API_VERSION_TOPOLOGY_V1 0x1
struct afe_param_id_set_topology_cfg {
/*
* Minor version used for tracking afe topology id configuration.
* @values #AFE_API_VERSION_TOPOLOGY_V1
*/
u32 minor_version;
/*
* Id of the topology for the afe session.
* @values Any valid AFE topology ID
*/
u32 topology_id;
} __packed;
/*
* Generic encoder module ID.
* This module supports the following parameter IDs:
* #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time)
* #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time)
* #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time)
* #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time)
* Opcode - AVS_MODULE_ID_ENCODER
* AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID.
*/
#define AFE_MODULE_ID_ENCODER 0x00013229
/* Macro for defining the packetizer ID: COP. */
#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A
/*
* Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter cannot be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E
/*
* Encoder config block parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter may be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C
/*
* Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module.
* This parameter cannot be set runtime.
*/
#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID 0x0001322B
/*
* Data format to send compressed data
* is transmitted/received over Slimbus lines.
*/
#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED 0x3
/*
* ID for AFE port module. This will be used to define port properties.
* This module supports following parameter IDs:
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
* To configure the port property, the client must use the
* #AFE_PORT_CMD_SET_PARAM_V2 command,
* and fill the module ID with the respective parameter IDs as listed above.
* @apr_hdr_fields
* Opcode -- AFE_MODULE_PORT
*/
#define AFE_MODULE_PORT 0x000102a6
/*
* ID of the parameter used by #AFE_MODULE_PORT to set the port media type.
* parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command.
*/
#define AFE_PARAM_ID_PORT_MEDIA_TYPE 0x000102a7
/*
* Macros for defining the "data_format" field in the
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
*/
#define AFE_PORT_DATA_FORMAT_PCM 0x0
#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED 0x1
/*
* Macro for defining the "minor_version" field in the
* #AFE_PARAM_ID_PORT_MEDIA_TYPE
*/
#define AFE_API_VERSION_PORT_MEDIA_TYPE 0x1
#define ASM_MEDIA_FMT_NONE 0x0
/*
* Media format ID for SBC encode configuration.
* @par SBC encode configuration (asm_sbc_enc_cfg_t)
* @table{weak__asm__sbc__enc__cfg__t}
*/
#define ASM_MEDIA_FMT_SBC 0x00010BF2
/* SBC channel Mono mode.*/
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO 1
/* SBC channel Stereo mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO 2
/* SBC channel Dual Mono mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO 8
/* SBC channel Joint Stereo mode. */
#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9
/* SBC bit allocation method = loudness. */
#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0
/* SBC bit allocation method = SNR. */
#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1
/*
* Payload of the SBC encoder configuration parameters in the
* #ASM_MEDIA_FMT_SBC media format.
*/
struct asm_sbc_enc_cfg_t {
/*
* Number of subbands.
* @values 4, 8
*/
uint32_t num_subbands;
/*
* Size of the encoded block in samples.
* @values 4, 8, 12, 16
*/
uint32_t blk_len;
/*
* Mode used to allocate bits between channels.
* @values
* 0 (Native mode)
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO
* #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO
* Native mode indicates that encoding must be performed with the number
* of channels at the input.
* If postprocessing outputs one-channel data, Mono mode is used. If
* postprocessing outputs two-channel data, Stereo mode is used.
* The number of channels must not change during encoding.
*/
uint32_t channel_mode;
/*
* Encoder bit allocation method.
* @values
* #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS
* #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend
*/
uint32_t alloc_method;
/*
* Number of encoded bits per second.
* @values
* Mono channel -- Maximum of 320 kbps
* Stereo channel -- Maximum of 512 kbps @tablebulletend
*/
uint32_t bit_rate;
/*
* Number of samples per second.
* @values 0 (Native mode), 16000, 32000, 44100, 48000&nbsp;Hz
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
uint32_t sample_rate;
};
#define ASM_MEDIA_FMT_AAC_AOT_LC 2
#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
#define ASM_MEDIA_FMT_AAC_AOT_PS 29
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
struct asm_aac_enc_cfg_v2_t {
/* Encoding rate in bits per second.*/
uint32_t bit_rate;
/*
* Encoding mode.
* Supported values:
* #ASM_MEDIA_FMT_AAC_AOT_LC
* #ASM_MEDIA_FMT_AAC_AOT_SBR
* #ASM_MEDIA_FMT_AAC_AOT_PS
*/
uint32_t enc_mode;
/*
* AAC format flag.
* Supported values:
* #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
uint16_t aac_fmt_flag;
/*
* Number of channels to encode.
* Supported values:
* 0 - Native mode
* 1 - Mono
* 2 - Stereo
* Other values are not supported.
* @note1hang The eAAC+ encoder mode supports only stereo.
* Native mode indicates that encoding must be performed with the
* number of channels at the input.
* The number of channels must not change during encoding.
*/
uint16_t channel_cfg;
/*
* Number of samples per second.
* Supported values: - 0 -- Native mode - For other values,
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
uint32_t sample_rate;
} __packed;
/* FMT ID for apt-X Classic */
#define ASM_MEDIA_FMT_APTX 0x000131ff
/* FMT ID for apt-X HD */
#define ASM_MEDIA_FMT_APTX_HD 0x00013200
#define PCM_CHANNEL_L 1
#define PCM_CHANNEL_R 2
#define PCM_CHANNEL_C 3
struct asm_custom_enc_cfg_aptx_t {
uint32_t sample_rate;
/* Mono or stereo */
uint16_t num_channels;
uint16_t reserved;
/* num_ch == 1, then PCM_CHANNEL_C,
* num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
*/
uint8_t channel_mapping[8];
uint32_t custom_size;
} __packed;
struct afe_enc_fmt_id_param_t {
/*
* Supported values:
* #ASM_MEDIA_FMT_SBC
* #ASM_MEDIA_FMT_AAC_V2
* Any OpenDSP supported values
*/
uint32_t fmt_id;
} __packed;
struct afe_port_media_type_t {
/*
* Minor version
* @values #AFE_API_VERSION_PORT_MEDIA_TYPE.
*/
uint32_t minor_version;
/*
* Sampling rate of the port.
* @values
* #AFE_PORT_SAMPLE_RATE_8K
* #AFE_PORT_SAMPLE_RATE_11_025K
* #AFE_PORT_SAMPLE_RATE_12K
* #AFE_PORT_SAMPLE_RATE_16K
* #AFE_PORT_SAMPLE_RATE_22_05K
* #AFE_PORT_SAMPLE_RATE_24K
* #AFE_PORT_SAMPLE_RATE_32K
* #AFE_PORT_SAMPLE_RATE_44_1K
* #AFE_PORT_SAMPLE_RATE_48K
* #AFE_PORT_SAMPLE_RATE_88_2K
* #AFE_PORT_SAMPLE_RATE_96K
* #AFE_PORT_SAMPLE_RATE_176_4K
* #AFE_PORT_SAMPLE_RATE_192K
* #AFE_PORT_SAMPLE_RATE_352_8K
* #AFE_PORT_SAMPLE_RATE_384K
*/
uint32_t sample_rate;
/*
* Bit width of the sample.
* @values 16, 24
*/
uint16_t bit_width;
/*
* Number of channels.
* @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
*/
uint16_t num_channels;
/*
* Data format supported by this port.
* If the port media type and device media type are different,
* it signifies a encoding/decoding use case
* @values
* #AFE_PORT_DATA_FORMAT_PCM
* #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED
*/
uint16_t data_format;
/*This field must be set to zero.*/
uint16_t reserved;
} __packed;
union afe_enc_config_data {
struct asm_sbc_enc_cfg_t sbc_config;
struct asm_aac_enc_cfg_v2_t aac_config;
struct asm_custom_enc_cfg_aptx_t aptx_config;
};
struct afe_enc_config {
u32 format;
union afe_enc_config_data data;
};
struct afe_enc_cfg_blk_param_t {
uint32_t enc_cfg_blk_size;
/*
*Size of the encoder configuration block that follows this member
*/
union afe_enc_config_data enc_blk_config;
};
/*
* Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
*/
struct avs_enc_packetizer_id_param_t {
/*
* Supported values:
* #AVS_MODULE_ID_PACKETIZER_COP
* Any OpenDSP supported values
*/
uint32_t enc_packetizer_id;
};
union afe_port_config {
struct afe_param_id_pcm_cfg pcm;
struct afe_param_id_i2s_cfg i2s;
struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
struct afe_param_id_slimbus_cfg slim_sch;
struct afe_param_id_rt_proxy_port_cfg rtproxy;
struct afe_param_id_internal_bt_fm_cfg int_bt_fm;
struct afe_param_id_pseudo_port_cfg pseudo_port;
struct afe_param_id_device_hw_delay_cfg hw_delay;
struct afe_param_id_spdif_cfg spdif;
struct afe_param_id_set_topology_cfg topology;
struct afe_param_id_tdm_cfg tdm;
struct afe_param_id_usb_audio_cfg usb_audio;
struct afe_enc_fmt_id_param_t enc_fmt;
struct afe_port_media_type_t media_type;
struct afe_enc_cfg_blk_param_t enc_blk_param;
struct avs_enc_packetizer_id_param_t enc_pkt_id_param;
} __packed;
struct afe_audioif_config_command_no_payload {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
} __packed;
struct afe_audioif_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
union afe_port_config port;
} __packed;
#define AFE_PORT_CMD_DEVICE_START 0x000100E5
/* Payload of the #AFE_PORT_CMD_DEVICE_START.*/
struct afe_port_cmd_device_start {
struct apr_hdr hdr;
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. An even
* number represents the Rx direction, and an odd number represents
* the Tx direction.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.*/
} __packed;
#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
/* Payload of the #AFE_PORT_CMD_DEVICE_STOP. */
struct afe_port_cmd_device_stop {
struct apr_hdr hdr;
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. An even
* number represents the Rx direction, and an odd number represents
* the Tx direction.
*/
u16 reserved;
/* Reserved for 32-bit alignment. This field must be set to 0.*/
} __packed;
#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA
/* Memory map regions command payload used by the
* #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS .
* This structure allows clients to map multiple shared memory
* regions in a single command. Following this structure are
* num_regions of afe_service_shared_map_region_payload.
*/
struct afe_service_cmd_shared_mem_map_regions {
struct apr_hdr hdr;
u16 mem_pool_id;
/* Type of memory on which this memory region is mapped.
* Supported values:
* - #ADSP_MEMORY_MAP_EBI_POOL
* - #ADSP_MEMORY_MAP_SMI_POOL
* - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL
* - Other values are reserved
*
* The memory pool ID implicitly defines the characteristics of the
* memory. Characteristics may include alignment type, permissions,
* etc.
*
* ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory
* ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory
* ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte
* addressable, and 4 KB aligned.
*/
u16 num_regions;
/* Number of regions to map.
* Supported values:
* - Any value greater than zero
*/
u32 property_flag;
/* Configures one common property for all the regions in the
* payload.
*
* Supported values: - 0x00000000 to 0x00000001
*
* b0 - bit 0 indicates physical or virtual mapping 0 Shared memory
* address provided in afe_service_shared_map_region_payloadis a
* physical address. The shared memory needs to be mapped( hardware
* TLB entry) and a software entry needs to be added for internal
* book keeping.
*
* 1 Shared memory address provided in
* afe_service_shared_map_region_payloadis a virtual address. The
* shared memory must not be mapped (since hardware TLB entry is
* already available) but a software entry needs to be added for
* internal book keeping. This can be useful if two services with in
* ADSP is communicating via APR. They can now directly communicate
* via the Virtual address instead of Physical address. The virtual
* regions must be contiguous. num_regions must be 1 in this case.
*
* b31-b1 - reserved bits. must be set to zero
*/
} __packed;
/* Map region payload used by the
* afe_service_shared_map_region_payloadstructure.
*/
struct afe_service_shared_map_region_payload {
u32 shm_addr_lsw;
/* least significant word of starting address in the memory
* region to map. It must be contiguous memory, and it must be 4 KB
* aligned.
* Supported values: - Any 32 bit value
*/
u32 shm_addr_msw;
/* most significant word of startng address in the memory region
* to map. For 32 bit shared memory address, this field must be set
* to zero. For 36 bit shared memory address, bit31 to bit 4 must be
* set to zero
*
* Supported values: - For 32 bit shared memory address, this field
* must be set to zero. - For 36 bit shared memory address, bit31 to
* bit 4 must be set to zero - For 64 bit shared memory address, any
* 32 bit value
*/
u32 mem_size_bytes;
/* Number of bytes in the region. The aDSP will always map the
* regions as virtual contiguous memory, but the memory size must be
* in multiples of 4 KB to avoid gaps in the virtually contiguous
* mapped memory.
*
* Supported values: - multiples of 4KB
*/
} __packed;
#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB
struct afe_service_cmdrsp_shared_mem_map_regions {
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory attributes is
* returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
* successful. In the case of failure , a generic APR error response
* is returned to the client.
*
* Supported Values: - Any 32 bit value
*/
} __packed;
#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC
/* Memory unmap regions command payload used by the
* #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS
*
* This structure allows clients to unmap multiple shared memory
* regions in a single command.
*/
struct afe_service_cmd_shared_mem_unmap_regions {
struct apr_hdr hdr;
u32 mem_map_handle;
/* memory map handle returned by
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands
*
* Supported Values:
* - Any 32 bit value
*/
} __packed;
#define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0
/* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command,
* which queries for one post/preprocessing parameter of a
* stream.
*/
struct afe_port_cmd_get_param_v2 {
u16 port_id;
/* Port interface and direction (Rx or Tx) to start. */
u16 payload_size;
/* Maximum data size of the parameter ID/module ID combination.
* This is a multiple of four bytes
* Supported values: > 0
*/
u32 payload_address_lsw;
/* LSW of 64 bit Payload address. Address should be 32-byte,
* 4kbyte aligned and must be contig memory.
*/
u32 payload_address_msw;
/* MSW of 64 bit Payload address. In case of 32-bit shared
* memory address, this field must be set to zero. In case of 36-bit
* shared memory address, bit-4 to bit-31 must be set to zero.
* Address should be 32-byte, 4kbyte aligned and must be contiguous
* memory.
*/
u32 mem_map_handle;
/* Memory map handle returned by
* AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
* Supported Values: - NULL -- Message. The parameter data is
* in-band. - Non-NULL -- The parameter data is Out-band.Pointer to
* - the physical address in shared memory of the payload data.
* For detailed payload content, see the afe_port_param_data_v2
* structure
*/
u32 module_id;
/* ID of the module to be queried.
* Supported values: Valid module ID
*/
u32 param_id;
/* ID of the parameter to be queried.
* Supported values: Valid parameter ID
*/
} __packed;
#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which
* responds to an #AFE_PORT_CMD_GET_PARAM_V2 command.
*
* Immediately following this structure is the parameters structure
* (afe_port_param_data) containing the response(acknowledgment)
* parameter payload. This payload is included for an in-band
* scenario. For an address/shared memory-based set parameter, this
* payload is not needed.
*/
struct afe_port_cmdrsp_get_param_v2 {
u32 status;
} __packed;
#define AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG 0x0001028C
#define AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG 0x1
/* Payload of the AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG parameter used by
* AFE_MODULE_AUDIO_DEV_INTERFACE.
*/
struct afe_param_id_lpass_core_shared_clk_cfg {
u32 lpass_core_shared_clk_cfg_minor_version;
/*
* Minor version used for lpass core shared clock configuration
* Supported value: AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG
*/
u32 enable;
/*
* Specifies whether the lpass core shared clock is
* enabled (1) or disabled (0).
*/
} __packed;
struct afe_lpass_core_shared_clk_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_lpass_core_shared_clk_cfg clk_cfg;
} __packed;
/* adsp_afe_service_commands.h */
#define ADSP_MEMORY_MAP_EBI_POOL 0
#define ADSP_MEMORY_MAP_SMI_POOL 1
#define ADSP_MEMORY_MAP_IMEM_POOL 2
#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
/* Definition of virtual memory flag */
#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1
/* Definition of physical memory flag */
#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0
#define NULL_POPP_TOPOLOGY 0x00010C68
#define NULL_COPP_TOPOLOGY 0x00010312
#define DEFAULT_COPP_TOPOLOGY 0x00010314
#define DEFAULT_POPP_TOPOLOGY 0x00010BE4
#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY 0x0001076B
#define COMPRESSED_PASSTHROUGH_NONE_TOPOLOGY 0x00010774
#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75
#define VPM_TX_DM_RFECNS_COPP_TOPOLOGY 0x00010F86
#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_DTS_HPX 0x10015002
#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_AUDIOSPHERE 0x10028000
/* Memory map regions command payload used by the
* #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS
* commands.
*
* This structure allows clients to map multiple shared memory
* regions in a single command. Following this structure are
* num_regions of avs_shared_map_region_payload.
*/
struct avs_cmd_shared_mem_map_regions {
struct apr_hdr hdr;
u16 mem_pool_id;
/* Type of memory on which this memory region is mapped.
*
* Supported values: - #ADSP_MEMORY_MAP_EBI_POOL -
* #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL
* (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values
* are reserved
*
* The memory ID implicitly defines the characteristics of the
* memory. Characteristics may include alignment type, permissions,
* etc.
*
* SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned.
*/
u16 num_regions;
/* Number of regions to map.*/
u32 property_flag;
/* Configures one common property for all the regions in the
* payload. No two regions in the same memory map regions cmd can
* have differnt property. Supported values: - 0x00000000 to
* 0x00000001
*
* b0 - bit 0 indicates physical or virtual mapping 0 shared memory
* address provided in avs_shared_map_regions_payload is physical
* address. The shared memory needs to be mapped( hardware TLB
* entry)
*
* and a software entry needs to be added for internal book keeping.
*
* 1 Shared memory address provided in MayPayload[usRegions] is
* virtual address. The shared memory must not be mapped (since
* hardware TLB entry is already available) but a software entry
* needs to be added for internal book keeping. This can be useful
* if two services with in ADSP is communicating via APR. They can
* now directly communicate via the Virtual address instead of
* Physical address. The virtual regions must be contiguous.
*
* b31-b1 - reserved bits. must be set to zero
*/
} __packed;
struct avs_shared_map_region_payload {
u32 shm_addr_lsw;
/* least significant word of shared memory address of the memory
* region to map. It must be contiguous memory, and it must be 4 KB
* aligned.
*/
u32 shm_addr_msw;
/* most significant word of shared memory address of the memory
* region to map. For 32 bit shared memory address, this field must
* tbe set to zero. For 36 bit shared memory address, bit31 to bit 4
* must be set to zero
*/
u32 mem_size_bytes;
/* Number of bytes in the region.
*
* The aDSP will always map the regions as virtual contiguous
* memory, but the memory size must be in multiples of 4 KB to avoid
* gaps in the virtually contiguous mapped memory.
*/
} __packed;
struct avs_cmd_shared_mem_unmap_regions {
struct apr_hdr hdr;
u32 mem_map_handle;
/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS
* , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands
*/
} __packed;
/* Memory map command response payload used by the
* #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS
* ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS
*/
struct avs_cmdrsp_shared_mem_map_regions {
u32 mem_map_handle;
/* A memory map handle encapsulating shared memory attributes is
* returned
*/
} __packed;
/*adsp_audio_memmap_api.h*/
/* ASM related data structures */
struct asm_wma_cfg {
u16 format_tag;
u16 ch_cfg;
u32 sample_rate;
u32 avg_bytes_per_sec;
u16 block_align;
u16 valid_bits_per_sample;
u32 ch_mask;
u16 encode_opt;
u16 adv_encode_opt;
u32 adv_encode_opt2;
u32 drc_peak_ref;
u32 drc_peak_target;
u32 drc_ave_ref;
u32 drc_ave_target;
} __packed;
struct asm_wmapro_cfg {
u16 format_tag;
u16 ch_cfg;
u32 sample_rate;
u32 avg_bytes_per_sec;
u16 block_align;
u16 valid_bits_per_sample;
u32 ch_mask;
u16 encode_opt;
u16 adv_encode_opt;
u32 adv_encode_opt2;
u32 drc_peak_ref;
u32 drc_peak_target;
u32 drc_ave_ref;
u32 drc_ave_target;
} __packed;
struct asm_aac_cfg {
u16 format;
u16 aot;
u16 ep_config;
u16 section_data_resilience;
u16 scalefactor_data_resilience;
u16 spectral_data_resilience;
u16 ch_cfg;
u16 reserved;
u32 sample_rate;
} __packed;
struct asm_amrwbplus_cfg {
u32 size_bytes;
u32 version;
u32 num_channels;
u32 amr_band_mode;
u32 amr_dtx_mode;
u32 amr_frame_fmt;
u32 amr_lsf_idx;
} __packed;
struct asm_flac_cfg {
u32 sample_rate;
u32 ext_sample_rate;
u32 min_frame_size;
u32 max_frame_size;
u16 stream_info_present;
u16 min_blk_size;
u16 max_blk_size;
u16 ch_cfg;
u16 sample_size;
u16 md5_sum;
};
struct asm_alac_cfg {
u32 frame_length;
u8 compatible_version;
u8 bit_depth;
u8 pb;
u8 mb;
u8 kb;
u8 num_channels;
u16 max_run;
u32 max_frame_bytes;
u32 avg_bit_rate;
u32 sample_rate;
u32 channel_layout_tag;
};
struct asm_g711_dec_cfg {
u32 sample_rate;
};
struct asm_vorbis_cfg {
u32 bit_stream_fmt;
};
struct asm_ape_cfg {
u16 compatible_version;
u16 compression_level;
u32 format_flags;
u32 blocks_per_frame;
u32 final_frame_blocks;
u32 total_frames;
u16 bits_per_sample;
u16 num_channels;
u32 sample_rate;
u32 seek_table_present;
};
struct asm_dsd_cfg {
u16 num_version;
u16 is_bitwise_big_endian;
u16 dsd_channel_block_size;
u16 num_channels;
u8 channel_mapping[8];
u32 dsd_data_rate;
};
struct asm_softpause_params {
u32 enable;
u32 period;
u32 step;
u32 rampingcurve;
} __packed;
struct asm_softvolume_params {
u32 period;
u32 step;
u32 rampingcurve;
} __packed;
#define ASM_END_POINT_DEVICE_MATRIX 0
#define PCM_CHANNEL_NULL 0
/* Front left channel. */
#define PCM_CHANNEL_FL 1
/* Front right channel. */
#define PCM_CHANNEL_FR 2
/* Front center channel. */
#define PCM_CHANNEL_FC 3
/* Left surround channel.*/
#define PCM_CHANNEL_LS 4
/* Right surround channel.*/
#define PCM_CHANNEL_RS 5
/* Low frequency effect channel. */
#define PCM_CHANNEL_LFE 6
/* Center surround channel; Rear center channel. */
#define PCM_CHANNEL_CS 7
/* Left back channel; Rear left channel. */
#define PCM_CHANNEL_LB 8
/* Right back channel; Rear right channel. */
#define PCM_CHANNEL_RB 9
/* Top surround channel. */
#define PCM_CHANNELS 10
/* Center vertical height channel.*/
#define PCM_CHANNEL_CVH 11
/* Mono surround channel.*/
#define PCM_CHANNEL_MS 12
/* Front left of center. */
#define PCM_CHANNEL_FLC 13
/* Front right of center. */
#define PCM_CHANNEL_FRC 14
/* Rear left of center. */
#define PCM_CHANNEL_RLC 15
/* Rear right of center. */
#define PCM_CHANNEL_RRC 16
#define PCM_FORMAT_MAX_NUM_CHANNEL 8
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C
#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
#define ASM_MEDIA_FMT_GENERIC_COMPRESSED 0x00013212
#define ASM_MAX_EQ_BANDS 12
#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
struct asm_data_cmd_media_fmt_update_v2 {
u32 fmt_blk_size;
/* Media format block size in bytes.*/
} __packed;
struct asm_generic_compressed_fmt_blk_t {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
/*
* Channel mapping array of bitstream output.
* Channel[i] mapping describes channel i inside the buffer, where
* i < num_channels. All valid used channels must be
* present at the beginning of the array.
*/
uint8_t channel_mapping[8];
/*
* Number of channels of the incoming bitstream.
* Supported values: 1,2,3,4,5,6,7,8
*/
uint16_t num_channels;
/*
* Nominal bits per sample value of the incoming bitstream.
* Supported values: 16, 32
*/
uint16_t bits_per_sample;
/*
* Nominal sampling rate of the incoming bitstream.
* Supported values: 8000, 11025, 16000, 22050, 24000, 32000,
* 44100, 48000, 88200, 96000, 176400, 192000,
* 352800, 384000
*/
uint32_t sampling_rate;
} __packed;
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_multi_channel_pcm_fmt_blk_v3 {
uint16_t num_channels;
/*
* Number of channels
* Supported values: 1 to 8
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel
* Supported values: 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second
* Supported values: 2000 to 48000, 96000,192000 Hz
*/
uint16_t is_signed;
/* Flag that indicates that PCM samples are signed (1) */
uint16_t sample_word_size;
/*
* Size in bits of the word that holds a sample of a channel.
* Supported values: 12,24,32
*/
uint8_t channel_mapping[8];
/*
* Each element, i, in the array describes channel i inside the buffer where
* 0 <= i < num_channels. Unused channels are set to 0.
*/
} __packed;
struct asm_multi_channel_pcm_fmt_blk_v4 {
uint16_t num_channels;
/*
* Number of channels
* Supported values: 1 to 8
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel
* Supported values: 16, 24, 32
*/
uint32_t sample_rate;
/*
* Number of samples per second
* Supported values: 2000 to 48000, 96000,192000 Hz
*/
uint16_t is_signed;
/* Flag that indicates that PCM samples are signed (1) */
uint16_t sample_word_size;
/*
* Size in bits of the word that holds a sample of a channel.
* Supported values: 12,24,32
*/
uint8_t channel_mapping[8];
/*
* Each element, i, in the array describes channel i inside the buffer where
* 0 <= i < num_channels. Unused channels are set to 0.
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
* 31 - for 24b unpacked or 32bit
*/
} __packed;
/*
* Payload of the multichannel PCM configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
*/
struct asm_multi_channel_pcm_fmt_blk_param_v3 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
struct asm_multi_channel_pcm_fmt_blk_v3 param;
} __packed;
/*
* Payload of the multichannel PCM configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
*/
struct asm_multi_channel_pcm_fmt_blk_param_v4 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
struct asm_multi_channel_pcm_fmt_blk_v4 param;
} __packed;
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
/* ID of the parameter. */
u32 param_size;
/* Data size of this parameter, in bytes. The size is a multiple
* of 4 bytes.
*/
} __packed;
struct asm_enc_cfg_blk_param_v2 {
u32 frames_per_buf;
/* Number of encoded frames to pack into each buffer.
*
* @note1hang This is only guidance information for the aDSP. The
* number of encoded frames put into each buffer (specified by the
* client) is less than or equal to this number.
*/
u32 enc_cfg_blk_size;
/* Size in bytes of the encoder configuration block that follows
* this member.
*/
} __packed;
/* @brief Dolby Digital Plus end point configuration structure
*/
struct asm_dec_ddp_endp_param_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
int endp_param_value;
} __packed;
/*
* Payload of the multichannel PCM encoder configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
*/
struct asm_multi_channel_pcm_enc_cfg_v4 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint16_t num_channels;
/*
* Number of PCM channels.
* @values
* - 0 -- Native mode
* - 1 -- 8 channels
* Native mode indicates that encoding must be performed with the number
* of channels at the input.
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel.
* @values 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second.
* @values 0, 8000 to 48000 Hz
* A value of 0 indicates the native sampling rate. Encoding is
* performed at the input sampling rate.
*/
uint16_t is_signed;
/*
* Flag that indicates the PCM samples are signed (1). Currently, only
* signed PCM samples are supported.
*/
uint16_t sample_word_size;
/*
* The size in bits of the word that holds a sample of a channel.
* @values 16, 24, 32
* 16-bit samples are always placed in 16-bit words:
* sample_word_size = 1.
* 24-bit samples can be placed in 32-bit words or in consecutive
* 24-bit words.
* - If sample_word_size = 32, 24-bit samples are placed in the
* most significant 24 bits of a 32-bit word.
* - If sample_word_size = 24, 24-bit samples are placed in
* 24-bit words. @tablebulletend
*/
uint8_t channel_mapping[8];
/*
* Channel mapping array expected at the encoder output.
* Channel[i] mapping describes channel i inside the buffer, where
* 0 @le i < num_channels. All valid used channels must be present at
* the beginning of the array.
* If Native mode is set for the channels, this field is ignored.
* @values See Section @xref{dox:PcmChannelDefs}
*/
uint16_t endianness;
/*
* Flag to indicate the endianness of the pcm sample
* Supported values: 0 - Little endian (all other formats)
* 1 - Big endian (AIFF)
*/
uint16_t mode;
/*
* Mode to provide additional info about the pcm input data.
* Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
* Q31 for unpacked 24b or 32b)
* 15 - for 16 bit
* 23 - for 24b packed or 8.24 format
* 31 - for 24b unpacked or 32bit
*/
} __packed;
/*
* Payload of the multichannel PCM encoder configuration parameters in
* the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
*/
struct asm_multi_channel_pcm_enc_cfg_v3 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint16_t num_channels;
/*
* Number of PCM channels.
* @values
* - 0 -- Native mode
* - 1 -- 8 channels
* Native mode indicates that encoding must be performed with the number
* of channels at the input.
*/
uint16_t bits_per_sample;
/*
* Number of bits per sample per channel.
* @values 16, 24
*/
uint32_t sample_rate;
/*
* Number of samples per second.
* @values 0, 8000 to 48000 Hz
* A value of 0 indicates the native sampling rate. Encoding is
* performed at the input sampling rate.
*/
uint16_t is_signed;
/*
* Flag that indicates the PCM samples are signed (1). Currently, only
* signed PCM samples are supported.
*/
uint16_t sample_word_size;
/*
* The size in bits of the word that holds a sample of a channel.
* @values 16, 24, 32
* 16-bit samples are always placed in 16-bit words:
* sample_word_size = 1.
* 24-bit samples can be placed in 32-bit words or in consecutive
* 24-bit words.
* - If sample_word_size = 32, 24-bit samples are placed in the
* most significant 24 bits of a 32-bit word.
* - If sample_word_size = 24, 24-bit samples are placed in
* 24-bit words. @tablebulletend
*/
uint8_t channel_mapping[8];
/*
* Channel mapping array expected at the encoder output.
* Channel[i] mapping describes channel i inside the buffer, where
* 0 @le i < num_channels. All valid used channels must be present at
* the beginning of the array.
* If Native mode is set for the channels, this field is ignored.
* @values See Section @xref{dox:PcmChannelDefs}
*/
};
/* @brief Multichannel PCM encoder configuration structure used
* in the #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
*/
struct asm_multi_channel_pcm_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
uint16_t num_channels;
/*< Number of PCM channels.
*
* Supported values: - 0 -- Native mode - 1 -- 8 Native mode
* indicates that encoding must be performed with the number of
* channels at the input.
*/
uint16_t bits_per_sample;
/*< Number of bits per sample per channel.
* Supported values: 16, 24
*/
uint32_t sample_rate;
/*< Number of samples per second (in Hertz).
*
* Supported values: 0, 8000 to 48000 A value of 0 indicates the
* native sampling rate. Encoding is performed at the input sampling
* rate.
*/
uint16_t is_signed;
/*< Specifies whether the samples are signed (1). Currently,
* only signed samples are supported.
*/
uint16_t reserved;
/*< reserved field for 32 bit alignment. must be set to zero.*/
uint8_t channel_mapping[8];
} __packed;
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6
/* @xreflabel
* {hdr:AsmMediaFmtDolbyAac} Media format ID for the
* Dolby AAC decoder. This format ID is be used if the client wants
* to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC
* contents.
*/
#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86
/* Enumeration for the audio data transport stream AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
/* Enumeration for low overhead audio stream AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS 1
/* Enumeration for the audio data interchange format
* AAC format.
*/
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF 2
/* Enumeration for the raw AAC format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
/* Enumeration for the AAC LATM format. */
#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM 4
#define ASM_MEDIA_FMT_AAC_AOT_LC 2
#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
#define ASM_MEDIA_FMT_AAC_AOT_PS 29
#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22
struct asm_aac_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 aac_fmt_flag;
/* Bitstream format option.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
u16 audio_objype;
/* Audio Object Type (AOT) present in the AAC stream.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_AOT_LC
* - #ASM_MEDIA_FMT_AAC_AOT_SBR
* - #ASM_MEDIA_FMT_AAC_AOT_BSAC
* - #ASM_MEDIA_FMT_AAC_AOT_PS
* - Otherwise -- Not supported
*/
u16 channel_config;
/* Number of channels present in the AAC stream.
* Supported values:
* - 1 -- Mono
* - 2 -- Stereo
* - 6 -- 5.1 content
*/
u16 total_size_of_PCE_bits;
/* greater or equal to zero. * -In case of RAW formats and
* channel config = 0 (PCE), client can send * the bit stream
* containing PCE immediately following this structure * (in-band).
* -This number does not include bits included for 32 bit alignment.
* -If zero, then the PCE info is assumed to be available in the
* audio -bit stream & not in-band.
*/
u32 sample_rate;
/* Number of samples per second (in Hertz).
*
* Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000,
* 44100, 48000
*
* This field must be equal to the sample rate of the AAC-LC
* decoder's output. - For MP4 or 3GP containers, this is indicated
* by the samplingFrequencyIndex field in the AudioSpecificConfig
* element. - For ADTS format, this is indicated by the
* samplingFrequencyIndex in the ADTS fixed header. - For ADIF
* format, this is indicated by the samplingFrequencyIndex in the
* program_config_element present in the ADIF header.
*/
} __packed;
struct asm_aac_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 bit_rate;
/* Encoding rate in bits per second. */
u32 enc_mode;
/* Encoding mode.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_AOT_LC
* - #ASM_MEDIA_FMT_AAC_AOT_SBR
* - #ASM_MEDIA_FMT_AAC_AOT_PS
*/
u16 aac_fmt_flag;
/* AAC format flag.
* Supported values:
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
*/
u16 channel_cfg;
/* Number of channels to encode.
* Supported values:
* - 0 -- Native mode
* - 1 -- Mono
* - 2 -- Stereo
* - Other values are not supported.
* @note1hang The eAAC+ encoder mode supports only stereo.
* Native mode indicates that encoding must be performed with the
* number of channels at the input.
* The number of channels must not change during encoding.
*/
u32 sample_rate;
/* Number of samples per second.
* Supported values: - 0 -- Native mode - For other values,
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
} __packed;
#define ASM_MEDIA_FMT_G711_ALAW_FS 0x00010BF7
#define ASM_MEDIA_FMT_G711_MLAW_FS 0x00010C2E
struct asm_g711_enc_cfg_v2 {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 sample_rate;
/*
* Number of samples per second.
* Supported values: 8000, 16000 Hz
*/
} __packed;
struct asm_vorbis_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 bit_stream_fmt;
/* Bit stream format.
* Supported values:
* - 0 -- Raw bitstream
* - 1 -- Transcoded bitstream
*
* Transcoded bitstream containing the size of the frame as the first
* word in each frame.
*/
} __packed;
struct asm_flac_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 is_stream_info_present;
/* Specifies whether stream information is present in the FLAC format
* block.
*
* Supported values:
* - 0 -- Stream information is not present in this message
* - 1 -- Stream information is present in this message
*
* When set to 1, the FLAC bitstream was successfully parsed by the
* client, and other fields in the FLAC format block can be read by the
* decoder to get metadata stream information.
*/
u16 num_channels;
/* Number of channels for decoding.
* Supported values: 1 to 2
*/
u16 min_blk_size;
/* Minimum block size (in samples) used in the stream. It must be less
* than or equal to max_blk_size.
*/
u16 max_blk_size;
/* Maximum block size (in samples) used in the stream. If the
* minimum block size equals the maximum block size, a fixed block
* size stream is implied.
*/
u16 md5_sum[8];
/* MD5 signature array of the unencoded audio data. This allows the
* decoder to determine if an error exists in the audio data, even when
* the error does not result in an invalid bitstream.
*/
u32 sample_rate;
/* Number of samples per second.
* Supported values: 8000 to 48000 Hz
*/
u32 min_frame_size;
/* Minimum frame size used in the stream.
* Supported values:
* - > 0 bytes
* - 0 -- The value is unknown
*/
u32 max_frame_size;
/* Maximum frame size used in the stream.
* Supported values:
* -- > 0 bytes
* -- 0 . The value is unknown
*/
u16 sample_size;
/* Bits per sample.Supported values: 8, 16 */
u16 reserved;
/* Clients must set this field to zero
*/
} __packed;
struct asm_alac_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 frame_length;
u8 compatible_version;
u8 bit_depth;
u8 pb;
u8 mb;
u8 kb;
u8 num_channels;
u16 max_run;
u32 max_frame_bytes;
u32 avg_bit_rate;
u32 sample_rate;
u32 channel_layout_tag;
} __packed;
struct asm_g711_dec_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 sample_rate;
} __packed;
struct asm_ape_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 compatible_version;
u16 compression_level;
u32 format_flags;
u32 blocks_per_frame;
u32 final_frame_blocks;
u32 total_frames;
u16 bits_per_sample;
u16 num_channels;
u32 sample_rate;
u32 seek_table_present;
} __packed;
struct asm_dsd_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 num_version;
u16 is_bitwise_big_endian;
u16 dsd_channel_block_size;
u16 num_channels;
u8 channel_mapping[8];
u32 dsd_data_rate;
} __packed;
#define ASM_MEDIA_FMT_AMRNB_FS 0x00010BEB
/* Enumeration for 4.75 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR475 0
/* Enumeration for 5.15 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR515 1
/* Enumeration for 5.90 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR59 2
/* Enumeration for 6.70 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR67 3
/* Enumeration for 7.40 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR74 4
/* Enumeration for 7.95 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR795 5
/* Enumeration for 10.20 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR102 6
/* Enumeration for 12.20 kbps AMR-NB Encoding mode. */
#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR122 7
/* Enumeration for AMR-NB Discontinuous Transmission mode off. */
#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF 0
/* Enumeration for AMR-NB DTX mode VAD1. */
#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 1
/* Enumeration for AMR-NB DTX mode VAD2. */
#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD2 2
/* Enumeration for AMR-NB DTX mode auto. */
#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_AUTO 3
struct asm_amrnb_enc_cfg {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u16 enc_mode;
/* AMR-NB encoding rate.
* Supported values:
* Use the ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_*
* macros
*/
u16 dtx_mode;
/* Specifies whether DTX mode is disabled or enabled.
* Supported values:
* - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
* - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
*/
} __packed;
#define ASM_MEDIA_FMT_AMRWB_FS 0x00010BEC
/* Enumeration for 6.6 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR66 0
/* Enumeration for 8.85 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR885 1
/* Enumeration for 12.65 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1265 2
/* Enumeration for 14.25 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1425 3
/* Enumeration for 15.85 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1585 4
/* Enumeration for 18.25 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1825 5
/* Enumeration for 19.85 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1985 6
/* Enumeration for 23.05 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2305 7
/* Enumeration for 23.85 kbps AMR-WB Encoding mode. */
#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2385 8
struct asm_amrwb_enc_cfg {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u16 enc_mode;
/* AMR-WB encoding rate.
* Suupported values:
* Use the ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_*
* macros
*/
u16 dtx_mode;
/* Specifies whether DTX mode is disabled or enabled.
* Supported values:
* - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
* - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
*/
} __packed;
#define ASM_MEDIA_FMT_V13K_FS 0x00010BED
/* Enumeration for 14.4 kbps V13K Encoding mode. */
#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 0
/* Enumeration for 12.2 kbps V13K Encoding mode. */
#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220 1
/* Enumeration for 11.2 kbps V13K Encoding mode. */
#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120 2
/* Enumeration for 9.0 kbps V13K Encoding mode. */
#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90 3
/* Enumeration for 7.2 kbps V13K eEncoding mode. */
#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720 4
/* Enumeration for 1/8 vocoder rate.*/
#define ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE 1
/* Enumeration for 1/4 vocoder rate. */
#define ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE 2
/* Enumeration for 1/2 vocoder rate. */
#define ASM_MEDIA_FMT_VOC_HALF_RATE 3
/* Enumeration for full vocoder rate. */
#define ASM_MEDIA_FMT_VOC_FULL_RATE 4
struct asm_v13k_enc_cfg {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u16 max_rate;
/* Maximum allowed encoder frame rate.
* Supported values:
* - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
* - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
* - #ASM_MEDIA_FMT_VOC_HALF_RATE
* - #ASM_MEDIA_FMT_VOC_FULL_RATE
*/
u16 min_rate;
/* Minimum allowed encoder frame rate.
* Supported values:
* - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
* - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
* - #ASM_MEDIA_FMT_VOC_HALF_RATE
* - #ASM_MEDIA_FMT_VOC_FULL_RATE
*/
u16 reduced_rate_cmd;
/* Reduced rate command, used to change
* the average bitrate of the V13K
* vocoder.
* Supported values:
* - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 (Default)
* - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220
* - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120
* - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90
* - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720
*/
u16 rate_mod_cmd;
/* Rate modulation command. Default = 0.
*- If bit 0=1, rate control is enabled.
*- If bit 1=1, the maximum number of consecutive full rate
* frames is limited with numbers supplied in
* bits 2 to 10.
*- If bit 1=0, the minimum number of non-full rate frames
* in between two full rate frames is forced to
* the number supplied in bits 2 to 10. In both cases, if necessary,
* half rate is used to substitute full rate. - Bits 15 to 10 are
* reserved and must all be set to zero.
*/
} __packed;
#define ASM_MEDIA_FMT_EVRC_FS 0x00010BEE
/* EVRC encoder configuration structure used in the
* #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
*/
struct asm_evrc_enc_cfg {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u16 max_rate;
/* Maximum allowed encoder frame rate.
* Supported values:
* - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
* - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
* - #ASM_MEDIA_FMT_VOC_HALF_RATE
* - #ASM_MEDIA_FMT_VOC_FULL_RATE
*/
u16 min_rate;
/* Minimum allowed encoder frame rate.
* Supported values:
* - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
* - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
* - #ASM_MEDIA_FMT_VOC_HALF_RATE
* - #ASM_MEDIA_FMT_VOC_FULL_RATE
*/
u16 rate_mod_cmd;
/* Rate modulation command. Default: 0.
* - If bit 0=1, rate control is enabled.
* - If bit 1=1, the maximum number of consecutive full rate frames
* is limited with numbers supplied in bits 2 to 10.
*
* - If bit 1=0, the minimum number of non-full rate frames in
* between two full rate frames is forced to the number supplied in
* bits 2 to 10. In both cases, if necessary, half rate is used to
* substitute full rate.
*
* - Bits 15 to 10 are reserved and must all be set to zero.
*/
u16 reserved;
/* Reserved. Clients must set this field to zero. */
} __packed;
#define ASM_MEDIA_FMT_WMA_V10PRO_V2 0x00010DA7
struct asm_wmaprov10_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 fmtag;
/* WMA format type.
* Supported values:
* - 0x162 -- WMA 9 Pro
* - 0x163 -- WMA 9 Pro Lossless
* - 0x166 -- WMA 10 Pro
* - 0x167 -- WMA 10 Pro Lossless
*/
u16 num_channels;
/* Number of channels encoded in the input stream.
* Supported values: 1 to 8
*/
u32 sample_rate;
/* Number of samples per second (in Hertz).
* Supported values: 11025, 16000, 22050, 32000, 44100, 48000,
* 88200, 96000
*/
u32 avg_bytes_per_sec;
/* Bitrate expressed as the average bytes per second.
* Supported values: 2000 to 96000
*/
u16 blk_align;
/* Size of the bitstream packet size in bytes. WMA Pro files
* have a payload of one block per bitstream packet.
* Supported values: @le 13376
*/
u16 bits_per_sample;
/* Number of bits per sample in the encoded WMA stream.
* Supported values: 16, 24
*/
u32 channel_mask;
/* Bit-packed double word (32-bits) that indicates the
* recommended speaker positions for each source channel.
*/
u16 enc_options;
/* Bit-packed word with values that indicate whether certain
* features of the bitstream are used.
* Supported values: - 0x0001 -- ENCOPT3_PURE_LOSSLESS - 0x0006 --
* ENCOPT3_FRM_SIZE_MOD - 0x0038 -- ENCOPT3_SUBFRM_DIV - 0x0040 --
* ENCOPT3_WRITE_FRAMESIZE_IN_HDR - 0x0080 --
* ENCOPT3_GENERATE_DRC_PARAMS - 0x0100 -- ENCOPT3_RTMBITS
*/
u16 usAdvancedEncodeOpt;
/* Advanced encoding option. */
u32 advanced_enc_options2;
/* Advanced encoding option 2. */
} __packed;
#define ASM_MEDIA_FMT_WMA_V9_V2 0x00010DA8
struct asm_wmastdv9_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u16 fmtag;
/* WMA format tag.
* Supported values: 0x161 (WMA 9 standard)
*/
u16 num_channels;
/* Number of channels in the stream.
* Supported values: 1, 2
*/
u32 sample_rate;
/* Number of samples per second (in Hertz).
* Supported values: 48000
*/
u32 avg_bytes_per_sec;
/* Bitrate expressed as the average bytes per second. */
u16 blk_align;
/* Block align. All WMA files with a maximum packet size of
* 13376 are supported.
*/
u16 bits_per_sample;
/* Number of bits per sample in the output.
* Supported values: 16
*/
u32 channel_mask;
/* Channel mask.
* Supported values:
* - 3 -- Stereo (front left/front right)
* - 4 -- Mono (center)
*/
u16 enc_options;
/* Options used during encoding. */
u16 reserved;
} __packed;
#define ASM_MEDIA_FMT_WMA_V8 0x00010D91
struct asm_wmastdv8_enc_cfg {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 bit_rate;
/* Encoding rate in bits per second. */
u32 sample_rate;
/* Number of samples per second.
*
* Supported values:
* - 0 -- Native mode
* - Other Supported values are 22050, 32000, 44100, and 48000.
*
* Native mode indicates that encoding must be performed with the
* sampling rate at the input.
* The sampling rate must not change during encoding.
*/
u16 channel_cfg;
/* Number of channels to encode.
* Supported values:
* - 0 -- Native mode
* - 1 -- Mono
* - 2 -- Stereo
* - Other values are not supported.
*
* Native mode indicates that encoding must be performed with the
* number of channels at the input.
* The number of channels must not change during encoding.
*/
u16 reserved;
/* Reserved. Clients must set this field to zero.*/
} __packed;
#define ASM_MEDIA_FMT_AMR_WB_PLUS_V2 0x00010DA9
struct asm_amrwbplus_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 amr_frame_fmt;
/* AMR frame format.
* Supported values:
* - 6 -- Transport Interface Format (TIF)
* - Any other value -- File storage format (FSF)
*
* TIF stream contains 2-byte header for each frame within the
* superframe. FSF stream contains one 2-byte header per superframe.
*/
} __packed;
#define ASM_MEDIA_FMT_AC3 0x00010DEE
#define ASM_MEDIA_FMT_EAC3 0x00010DEF
#define ASM_MEDIA_FMT_DTS 0x00010D88
#define ASM_MEDIA_FMT_MP2 0x00010DE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
#define ASM_MEDIA_FMT_ALAC 0x00012F31
#define ASM_MEDIA_FMT_VORBIS 0x00010C15
#define ASM_MEDIA_FMT_APE 0x00012F32
#define ASM_MEDIA_FMT_DSD 0x00012F3E
/* 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 doesn't reset unless the system is reset.
*/
u32 timestamp_msw;
/* Upper 32 bits of the 64-bit session time in microseconds of
* the first sample in the buffer.
*/
u32 flags;
/* Bitfield of flags. Bit 30 indicates whether frame metadata is
* present. If frame metadata is present, num_frames consecutive
* instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start
* at the buffer address.
* Supported values for bit 31:
* - 1 -- Timestamp is valid.
* - 0 -- Timestamp is invalid.
* - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and
* #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit.
*
* Supported values for bit 30:
* - 1 -- Frame metadata is present.
* - 0 -- Frame metadata is absent.
* - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and
* #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit.
*
* All other bits are reserved; the aDSP sets them to 0.
*/
u32 num_frames;
/* Number of encoded frames in the buffer. */
u32 seq_id;
/* Optional buffer sequence ID. */
} __packed;
struct asm_data_read_buf_metadata_v2 {
u32 offset;
/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to
* the frame associated with this metadata.
* Supported values: > 0
*/
u32 frm_size;
/* Size of the encoded frame in bytes.
* Supported values: > 0
*/
u32 num_encoded_pcm_samples;
/* Number of encoded PCM samples (per channel) in the frame
* associated with this metadata.
* Supported values: > 0
*/
u32 timestamp_lsw;
/* Lower 32 bits of the 64-bit session time in microseconds of the
* first sample for this frame.
* If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1
* then the 64 bit timestamp is absolute capture time otherwise it
* is relative session time. The absolute timestamp doesn't reset
* unless the system is reset.
*/
u32 timestamp_msw;
/* Lower 32 bits of the 64-bit session time in microseconds of the
* first sample for this frame.
*/
u32 flags;
/* Frame flags.
* Supported values for bit 31:
* - 1 -- Time stamp is valid
* - 0 -- Time stamp is not valid
* - All other bits are reserved; the aDSP sets them to 0.
*/
} __packed;
/* Notifies the client of a change in the data sampling rate or
* Channel mode. This event is raised by the decoder service. The
* event is enabled through the mode flags of
* #ASM_STREAM_CMD_OPEN_WRITE_V2 or
* #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
* in the output sampling frequency or the number/positioning of
* output channels, or if it is the first frame decoded.The new
* sampling frequency or the new channel configuration is
* communicated back to the client asynchronously.
*/
#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event.
* This event is raised when the following conditions are both true:
* - The event is enabled through the mode_flags of
* #ASM_STREAM_CMD_OPEN_WRITE_V2 or
* #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
* in either the output sampling frequency or the number/positioning
* of output channels, or if it is the first frame decoded.
* This event is not raised (even if enabled) if the decoder is
* MIDI, because
*/
struct asm_data_event_sr_cm_change_notify {
u32 sample_rate;
/* New sampling rate (in Hertz) after detecting a change in the
* bitstream.
* Supported values: 2000 to 48000
*/
u16 num_channels;
/* New number of channels after detecting a change in the
* bitstream.
* Supported values: 1 to 8
*/
u16 reserved;
/* Reserved for future use. This field must be set to 0.*/
u8 channel_mapping[8];
} __packed;
/* Notifies the client of a data sampling rate or channel mode
* change. This event is raised by the encoder service.
* This event is raised when :
* - Native mode encoding was requested in the encoder
* configuration (i.e., the channel number was 0), the sample rate
* was 0, or both were 0.
*
* - The input data frame at the encoder is the first one, or the
* sampling rate/channel mode is different from the previous input
* data frame.
*
*/
#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE
struct asm_data_event_enc_sr_cm_change_notify {
u32 sample_rate;
/* New sampling rate (in Hertz) after detecting a change in the
* input data.
* Supported values: 2000 to 48000
*/
u16 num_channels;
/* New number of channels after detecting a change in the input
* data. Supported values: 1 to 8
*/
u16 bits_per_sample;
/* New bits per sample after detecting a change in the input
* data.
* Supported values: 16, 24
*/
u8 channel_mapping[8];
} __packed;
#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87
/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command,
* which is used to indicate the IEC 60958 frame rate of a given
* packetized audio stream.
*/
struct asm_data_cmd_iec_60958_frame_rate {
u32 frame_rate;
/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream.
* Supported values: Any valid frame rate
*/
} __packed;
/* adsp_asm_data_commands.h*/
/* Definition of the stream ID bitmask.*/
#define ASM_BIT_MASK_STREAM_ID (0x000000FFUL)
/* Definition of the stream ID shift value.*/
#define ASM_SHIFT_STREAM_ID 0
/* Definition of the session ID bitmask.*/
#define ASM_BIT_MASK_SESSION_ID (0x0000FF00UL)
/* Definition of the session ID shift value.*/
#define ASM_SHIFT_SESSION_ID 8
/* Definition of the service ID bitmask.*/
#define ASM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
/* Definition of the service ID shift value.*/
#define ASM_SHIFT_SERVICE_ID 16
/* Definition of the domain ID bitmask.*/
#define ASM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
/* Definition of the domain ID shift value.*/
#define ASM_SHIFT_DOMAIN_ID 24
#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
/* adsp_asm_service_commands.h */
#define ASM_MAX_SESSION_ID (15)
/* Maximum number of sessions.*/
#define ASM_MAX_NUM_SESSIONS ASM_MAX_SESSION_ID
/* Maximum number of streams per session.*/
#define ASM_MAX_STREAMS_PER_SESSION (8)
#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE 0
#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1
#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2
#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY 3
#define ASM_BIT_MASK_RUN_STARTIME (0x00000003UL)
/* Bit shift value used to specify the start time for the
* ASM_SESSION_CMD_RUN_V2 command.
*/
#define ASM_SHIFT_RUN_STARTIME 0
struct asm_session_cmd_run_v2 {
struct apr_hdr hdr;
u32 flags;
/* Specifies whether to run immediately or at a specific
* rendering time or with a specified delay. Run with delay is
* useful for delaying in case of ASM loopback opened through
* ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME
* and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag.
*
*
*Bits 0 and 1 can take one of four possible values:
*
*- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE
*- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME
*- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME
*- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY
*
*All other bits are reserved; clients must set them to zero.
*/
u32 time_lsw;
/* Lower 32 bits of the time in microseconds used to align the
* session origin time. When bits 0-1 of flags is
* ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of
* the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
* maximum value of the 64 bit delay is 150 ms.
*/
u32 time_msw;
/* Upper 32 bits of the time in microseconds used to align the
* session origin time. When bits 0-1 of flags is
* ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of
* the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
* maximum value of the 64 bit delay is 150 ms.
*/
} __packed;
#define ASM_SESSION_CMD_PAUSE 0x00010BD3
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D
#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
struct asm_session_cmd_rgstr_rx_underflow {
struct apr_hdr hdr;
u16 enable_flag;
/* Specifies whether a client is to receive events when an Rx
* session underflows.
* Supported values:
* - 0 -- Do not send underflow events
* - 1 -- Send underflow events
*/
u16 reserved;
/* Reserved. This field must be set to zero.*/
} __packed;
#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6
struct asm_session_cmd_regx_overflow {
struct apr_hdr hdr;
u16 enable_flag;
/* Specifies whether a client is to receive events when a Tx
* session overflows.
* Supported values:
* - 0 -- Do not send overflow events
* - 1 -- Send overflow events
*/
u16 reserved;
/* Reserved. This field must be set to zero.*/
} __packed;
#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17
#define ASM_SESSION_EVENTX_OVERFLOW 0x00010C18
#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E
struct asm_session_cmdrsp_get_sessiontime_v3 {
u32 status;
/* Status message (error code).
* Supported values: Refer to @xhyperref{Q3,[Q3]}
*/
u32 sessiontime_lsw;
/* Lower 32 bits of the current session time in microseconds.*/
u32 sessiontime_msw;
/* Upper 32 bits of the current session time in microseconds.*/
u32 absolutetime_lsw;
/* Lower 32 bits in micro seconds of the absolute time at which
* the * sample corresponding to the above session time gets
* rendered * to hardware. This absolute time may be slightly in the
* future or past.
*/
u32 absolutetime_msw;
/* Upper 32 bits in micro seconds of the absolute time at which
* the * sample corresponding to the above session time gets
* rendered to * hardware. This absolute time may be slightly in the
* future or past.
*/
} __packed;
#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2 0x00010D9F
struct asm_session_cmd_adjust_session_clock_v2 {
struct apr_hdr hdr;
u32 adjustime_lsw;
/* Lower 32 bits of the signed 64-bit quantity that specifies the
* adjustment time in microseconds to the session clock.
*
* Positive values indicate advancement of the session clock.
* Negative values indicate delay of the session clock.
*/
u32 adjustime_msw;
/* Upper 32 bits of the signed 64-bit quantity that specifies
* the adjustment time in microseconds to the session clock.
* Positive values indicate advancement of the session clock.
* Negative values indicate delay of the session clock.
*/
} __packed;
#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 0x00010DA0
struct asm_session_cmdrsp_adjust_session_clock_v2 {
u32 status;
/* Status message (error code).
* Supported values: Refer to @xhyperref{Q3,[Q3]}
* An error means the session clock is not adjusted. In this case,
* the next two fields are irrelevant.
*/
u32 actual_adjustime_lsw;
/* Lower 32 bits of the signed 64-bit quantity that specifies
* the actual adjustment in microseconds performed by the aDSP.
* A positive value indicates advancement of the session clock. A
* negative value indicates delay of the session clock.
*/
u32 actual_adjustime_msw;
/* Upper 32 bits of the signed 64-bit quantity that specifies
* the actual adjustment in microseconds performed by the aDSP.
* A positive value indicates advancement of the session clock. A
* negative value indicates delay of the session clock.
*/
u32 cmd_latency_lsw;
/* Lower 32 bits of the unsigned 64-bit quantity that specifies
* the amount of time in microseconds taken to perform the session
* clock adjustment.
*/
u32 cmd_latency_msw;
/* Upper 32 bits of the unsigned 64-bit quantity that specifies
* the amount of time in microseconds taken to perform the session
* clock adjustment.
*/
} __packed;
#define ASM_SESSION_CMD_GET_PATH_DELAY_V2 0x00010DAF
#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0
struct asm_session_cmdrsp_get_path_delay_v2 {
u32 status;
/* Status message (error code). Whether this get delay operation
* is successful or not. Delay value is valid only if status is
* success.
* Supported values: Refer to @xhyperref{Q5,[Q5]}
*/
u32 audio_delay_lsw;
/* Upper 32 bits of the aDSP delay in microseconds. */
u32 audio_delay_msw;
/* Lower 32 bits of the aDSP delay in microseconds. */
} __packed;
/* adsp_asm_session_command.h*/
#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
#define ASM_LOW_LATENCY_STREAM_SESSION 0x10000000
#define ASM_ULTRA_LOW_LATENCY_STREAM_SESSION 0x20000000
#define ASM_ULL_POST_PROCESSING_STREAM_SESSION 0x40000000
#define ASM_LEGACY_STREAM_SESSION 0
struct asm_stream_cmd_open_write_v3 {
struct apr_hdr hdr;
uint32_t mode_flags;
/* Mode flags that configure the stream to notify the client
* whenever it detects an SR/CM change at the input to its POPP.
* Supported values for bits 0 to 1:
* - Reserved; clients must set them to zero.
* Supported values for bit 2:
* - 0 -- SR/CM change notification event is disabled.
* - 1 -- SR/CM change notification event is enabled.
* - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
* #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit.
*
* Supported values for bit 31:
* - 0 -- Stream to be opened in on-Gapless mode.
* - 1 -- Stream to be opened in Gapless mode. In Gapless mode,
* successive streams must be opened with same session ID but
* different stream IDs.
*
* - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and
* #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit.
*
*
* @note1hang MIDI and DTMF streams cannot be opened in Gapless mode.
*/
uint16_t sink_endpointype;
/*< Sink point type.
* Supported values:
* - 0 -- Device matrix
* - Other values are reserved.
*
* The device matrix is the gateway to the hardware ports.
*/
uint16_t bits_per_sample;
/*< Number of bits per sample processed by ASM modules.
* Supported values: 16 and 24 bits per sample
*/
uint32_t postprocopo_id;
/*< Specifies the topology (order of processing) of
* postprocessing algorithms. <i>None</i> means no postprocessing.
* Supported values:
* - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
* - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
* - #ASM_STREAM_POSTPROCOPO_ID_NONE
*
* This field can also be enabled through SetParams flags.
*/
uint32_t dec_fmt_id;
/*< Configuration ID of the decoder media format.
*
* Supported values:
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_ADPCM
* - #ASM_MEDIA_FMT_MP3
* - #ASM_MEDIA_FMT_AAC_V2
* - #ASM_MEDIA_FMT_DOLBY_AAC
* - #ASM_MEDIA_FMT_AMRNB_FS
* - #ASM_MEDIA_FMT_AMRWB_FS
* - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
* - #ASM_MEDIA_FMT_V13K_FS
* - #ASM_MEDIA_FMT_EVRC_FS
* - #ASM_MEDIA_FMT_EVRCB_FS
* - #ASM_MEDIA_FMT_EVRCWB_FS
* - #ASM_MEDIA_FMT_SBC
* - #ASM_MEDIA_FMT_WMA_V10PRO_V2
* - #ASM_MEDIA_FMT_WMA_V9_V2
* - #ASM_MEDIA_FMT_AC3
* - #ASM_MEDIA_FMT_EAC3
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_FR_FS
* - #ASM_MEDIA_FMT_VORBIS
* - #ASM_MEDIA_FMT_FLAC
* - #ASM_MEDIA_FMT_ALAC
* - #ASM_MEDIA_FMT_APE
* - #ASM_MEDIA_FMT_EXAMPLE
*/
} __packed;
#define ASM_STREAM_CMD_OPEN_PULL_MODE_WRITE 0x00010DD9
/* Bitmask for the stream_perf_mode subfield. */
#define ASM_BIT_MASK_STREAM_PERF_FLAG_PULL_MODE_WRITE 0xE0000000UL
/* Bitmask for the stream_perf_mode subfield. */
#define ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE 29
#define ASM_STREAM_CMD_OPEN_PUSH_MODE_READ 0x00010DDA
#define ASM_BIT_MASK_STREAM_PERF_FLAG_PUSH_MODE_READ 0xE0000000UL
#define ASM_SHIFT_STREAM_PERF_FLAG_PUSH_MODE_READ 29
#define ASM_DATA_EVENT_WATERMARK 0x00010DDB
struct asm_shared_position_buffer {
volatile uint32_t frame_counter;
/* Counter used to handle interprocessor synchronization issues.
* When frame_counter is 0: read_index, wall_clock_us_lsw, and
* wall_clock_us_msw are invalid.
* Supported values: >= 0.
*/
volatile uint32_t index;
/* Index in bytes from where the aDSP is reading/writing.
* Supported values: 0 to circular buffer size - 1
*/
volatile uint32_t wall_clock_us_lsw;
/* Lower 32 bits of the 64-bit wall clock time in microseconds when the
* read index was updated.
* Supported values: >= 0
*/
volatile uint32_t wall_clock_us_msw;
/* Upper 32 bits of the 64 bit wall clock time in microseconds when the
* read index was updated
* Supported values: >= 0
*/
} __packed;
struct asm_shared_watermark_level {
uint32_t watermark_level_bytes;
} __packed;
struct asm_stream_cmd_open_shared_io {
struct apr_hdr hdr;
uint32_t mode_flags;
uint16_t endpoint_type;
uint16_t topo_bits_per_sample;
uint32_t topo_id;
uint32_t fmt_id;
uint32_t shared_pos_buf_phy_addr_lsw;
uint32_t shared_pos_buf_phy_addr_msw;
uint16_t shared_pos_buf_mem_pool_id;
uint16_t shared_pos_buf_num_regions;
uint32_t shared_pos_buf_property_flag;
uint32_t shared_circ_buf_start_phy_addr_lsw;
uint32_t shared_circ_buf_start_phy_addr_msw;
uint32_t shared_circ_buf_size;
uint16_t shared_circ_buf_mem_pool_id;
uint16_t shared_circ_buf_num_regions;
uint32_t shared_circ_buf_property_flag;
uint32_t num_watermark_levels;
struct asm_multi_channel_pcm_fmt_blk_v3 fmt;
struct avs_shared_map_region_payload map_region_pos_buf;
struct avs_shared_map_region_payload map_region_circ_buf;
struct asm_shared_watermark_level watermark[0];
} __packed;
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
/* Definition of the timestamp type flag bitmask */
#define ASM_BIT_MASKIMESTAMPYPE_FLAG (0x00000020UL)
/* Definition of the timestamp type flag shift value. */
#define ASM_SHIFTIMESTAMPYPE_FLAG 5
/* Relative timestamp is identified by this value.*/
#define ASM_RELATIVEIMESTAMP 0
/* Absolute timestamp is identified by this value.*/
#define ASM_ABSOLUTEIMESTAMP 1
/* Bit value for Low Latency Tx stream subfield */
#define ASM_LOW_LATENCY_TX_STREAM_SESSION 1
/* Bit shift for the stream_perf_mode subfield. */
#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
struct asm_stream_cmd_open_read_v3 {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags that indicate whether meta information per encoded
* frame is to be provided.
* Supported values for bit 4:
*
* - 0 -- Return data buffer contains all encoded frames only; it
* does not contain frame metadata.
*
* - 1 -- Return data buffer contains an array of metadata and
* encoded frames.
*
* - Use #ASM_BIT_MASK_META_INFO_FLAG as the bitmask and
* #ASM_SHIFT_META_INFO_FLAG as the shift value for this bit.
*
*
* Supported values for bit 5:
*
* - ASM_RELATIVEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will have
* - relative time-stamp.
* - ASM_ABSOLUTEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will
* - have absolute time-stamp.
*
* - Use #ASM_BIT_MASKIMESTAMPYPE_FLAG as the bitmask and
* #ASM_SHIFTIMESTAMPYPE_FLAG as the shift value for this bit.
*
* All other bits are reserved; clients must set them to zero.
*/
u32 src_endpointype;
/* Specifies the endpoint providing the input samples.
* Supported values:
* - 0 -- Device matrix
* - All other values are reserved; clients must set them to zero.
* Otherwise, an error is returned.
* The device matrix is the gateway from the tunneled Tx ports.
*/
u32 preprocopo_id;
/* Specifies the topology (order of processing) of preprocessing
* algorithms. <i>None</i> means no preprocessing.
* Supported values:
* - #ASM_STREAM_PREPROCOPO_ID_DEFAULT
* - #ASM_STREAM_PREPROCOPO_ID_NONE
*
* This field can also be enabled through SetParams flags.
*/
u32 enc_cfg_id;
/* Media configuration ID for encoded output.
* Supported values:
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_AAC_V2
* - #ASM_MEDIA_FMT_AMRNB_FS
* - #ASM_MEDIA_FMT_AMRWB_FS
* - #ASM_MEDIA_FMT_V13K_FS
* - #ASM_MEDIA_FMT_EVRC_FS
* - #ASM_MEDIA_FMT_EVRCB_FS
* - #ASM_MEDIA_FMT_EVRCWB_FS
* - #ASM_MEDIA_FMT_SBC
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_EXAMPLE
* - #ASM_MEDIA_FMT_WMA_V8
*/
u16 bits_per_sample;
/* Number of bits per sample processed by ASM modules.
* Supported values: 16 and 24 bits per sample
*/
u16 reserved;
/* Reserved for future use. This field must be set to zero.*/
} __packed;
#define ASM_POPP_OUTPUT_SR_NATIVE_RATE 0
/* Enumeration for the maximum sampling rate at the POPP output.*/
#define ASM_POPP_OUTPUT_SR_MAX_RATE 48000
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
struct asm_stream_cmd_open_readwrite_v2 {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags.
* Supported values for bit 2:
* - 0 -- SR/CM change notification event is disabled.
* - 1 -- SR/CM change notification event is enabled. Use
* #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
* #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or
* getting this flag.
*
* Supported values for bit 4:
* - 0 -- Return read data buffer contains all encoded frames only; it
* does not contain frame metadata.
* - 1 -- Return read data buffer contains an array of metadata and
* encoded frames.
*
* All other bits are reserved; clients must set them to zero.
*/
u32 postprocopo_id;
/* Specifies the topology (order of processing) of postprocessing
* algorithms. <i>None</i> means no postprocessing.
*
* Supported values:
* - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
* - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
* - #ASM_STREAM_POSTPROCOPO_ID_NONE
*/
u32 dec_fmt_id;
/* Specifies the media type of the input data. PCM indicates that
* no decoding must be performed, e.g., this is an NT encoder
* session.
* Supported values:
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_ADPCM
* - #ASM_MEDIA_FMT_MP3
* - #ASM_MEDIA_FMT_AAC_V2
* - #ASM_MEDIA_FMT_DOLBY_AAC
* - #ASM_MEDIA_FMT_AMRNB_FS
* - #ASM_MEDIA_FMT_AMRWB_FS
* - #ASM_MEDIA_FMT_V13K_FS
* - #ASM_MEDIA_FMT_EVRC_FS
* - #ASM_MEDIA_FMT_EVRCB_FS
* - #ASM_MEDIA_FMT_EVRCWB_FS
* - #ASM_MEDIA_FMT_SBC
* - #ASM_MEDIA_FMT_WMA_V10PRO_V2
* - #ASM_MEDIA_FMT_WMA_V9_V2
* - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
* - #ASM_MEDIA_FMT_AC3
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_EXAMPLE
*/
u32 enc_cfg_id;
/* Specifies the media type for the output of the stream. PCM
* indicates that no encoding must be performed, e.g., this is an NT
* decoder session.
* Supported values:
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_AAC_V2
* - #ASM_MEDIA_FMT_AMRNB_FS
* - #ASM_MEDIA_FMT_AMRWB_FS
* - #ASM_MEDIA_FMT_V13K_FS
* - #ASM_MEDIA_FMT_EVRC_FS
* - #ASM_MEDIA_FMT_EVRCB_FS
* - #ASM_MEDIA_FMT_EVRCWB_FS
* - #ASM_MEDIA_FMT_SBC
* - #ASM_MEDIA_FMT_G711_ALAW_FS
* - #ASM_MEDIA_FMT_G711_MLAW_FS
* - #ASM_MEDIA_FMT_G729A_FS
* - #ASM_MEDIA_FMT_EXAMPLE
* - #ASM_MEDIA_FMT_WMA_V8
*/
u16 bits_per_sample;
/* Number of bits per sample processed by ASM modules.
* Supported values: 16 and 24 bits per sample
*/
u16 reserved;
/* Reserved for future use. This field must be set to zero.*/
} __packed;
#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E
struct asm_stream_cmd_open_loopback_v2 {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags.
* Bit 0-31: reserved; client should set these bits to 0
*/
u16 src_endpointype;
/* Endpoint type. 0 = Tx Matrix */
u16 sink_endpointype;
/* Endpoint type. 0 = Rx Matrix */
u32 postprocopo_id;
/* Postprocessor topology ID. Specifies the topology of
* postprocessing algorithms.
*/
u16 bits_per_sample;
/* The number of bits per sample processed by ASM modules
* Supported values: 16 and 24 bits per sample
*/
u16 reserved;
/* Reserved for future use. This field must be set to zero. */
} __packed;
#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA
/* Bitmask for the stream's Performance mode. */
#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \
(0x70000000UL)
/* Bit shift for the stream's Performance mode. */
#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28
/* Bitmask for the decoder converter enable flag. */
#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL)
/* Shift value for the decoder converter enable flag. */
#define ASM_SHIFT_DECODER_CONVERTER_FLAG 3
/* Converter mode is None (Default). */
#define ASM_CONVERTER_MODE_NONE 0
/* Converter mode is DDP-to-DD. */
#define ASM_DDP_DD_CONVERTER_MODE 1
/* Identifies a special converter mode where source and sink formats
* are the same but postprocessing must applied. Therefore, Decode
* @rarrow Re-encode is necessary.
*/
#define ASM_POST_PROCESS_CONVERTER_MODE 2
struct asm_stream_cmd_open_transcode_loopback_t {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode Flags specifies the performance mode in which this stream
* is to be opened.
* Supported values{for bits 30 to 28}(stream_perf_mode flag)
*
* #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward
* compatibility to the original behavior
* of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK
*
* #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using
* shortened buffers in low latency POPP
* - Recommendation: Do not enable high latency algorithms. They might
* negate the benefits of opening a low latency stream, and they
* might also suffer quality degradation from unexpected jitter.
* - This Low Latency mode is supported only for PCM In and PCM Out
* loopbacks. An error is returned if Low Latency mode is opened for
* other transcode loopback modes.
* - To configure this subfield, use
* ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and
* ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK.
*
* Supported values{for bits 6 to 3} (decoder-converter compatibility)
* #ASM_CONVERTER_MODE_NONE (0x0) -- Default
* #ASM_DDP_DD_CONVERTER_MODE (0x1)
* #ASM_POST_PROCESS_CONVERTER_MODE (0x2)
* 0x3-0xF -- Reserved for future use
* - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and
* ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit
* All other bits are reserved; clients must set them to 0.
*/
u32 src_format_id;
/* Specifies the media format of the input audio stream.
*
* Supported values
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
* - #ASM_MEDIA_FMT_DTS
* - #ASM_MEDIA_FMT_EAC3_DEC
* - #ASM_MEDIA_FMT_EAC3
* - #ASM_MEDIA_FMT_AC3_DEC
* - #ASM_MEDIA_FMT_AC3
*/
u32 sink_format_id;
/* Specifies the media format of the output stream.
*
* Supported values
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
* - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
* - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode)
* - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode)
*/
u32 audproc_topo_id;
/* Postprocessing topology ID, which specifies the topology (order of
* processing) of postprocessing algorithms.
*
* Supported values
* - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT
* - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER
* - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL
* - #ASM_STREAM_POSTPROC_TOPO_ID_NONE
* Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES.
* This field is ignored for the Converter mode, in which no
* postprocessing is performed.
*/
u16 src_endpoint_type;
/* Specifies the source endpoint that provides the input samples.
*
* Supported values
* - 0 -- Tx device matrix or stream router (gateway to the hardware
* ports)
* - All other values are reserved
* Clients must set this field to 0. Otherwise, an error is returned.
*/
u16 sink_endpoint_type;
/* Specifies the sink endpoint type.
*
* Supported values
* - 0 -- Rx device matrix or stream router (gateway to the hardware
* ports)
* - All other values are reserved
* Clients must set this field to 0. Otherwise, an error is returned.
*/
u16 bits_per_sample;
/* Number of bits per sample processed by the ASM modules.
* Supported values 16, 24
*/
u16 reserved;
/* This field must be set to 0.
*/
} __packed;
#define ASM_STREAM_CMD_CLOSE 0x00010BCD
#define ASM_STREAM_CMD_FLUSH 0x00010BCE
#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1
struct asm_stream_cmd_set_pp_params_v2 {
u32 data_payload_addr_lsw;
/* LSW of parameter data payload address. Supported values: any. */
u32 data_payload_addr_msw;
/* MSW of Parameter data payload address. Supported values: any.
* - Must be set to zero for in-band data.
* - In the case of 32 bit Shared memory address, msw field must be
* - set to zero.
* - In the case of 36 bit shared memory address, bit 31 to bit 4 of
* msw
*
* - must be set to zero.
*/
u32 mem_map_handle;
/* Supported Values: Any.
* memory map handle returned by DSP through
* ASM_CMD_SHARED_MEM_MAP_REGIONS
* command.
* if mmhandle is NULL, the ParamData payloads are within the
* message payload (in-band).
* If mmhandle is non-NULL, the ParamData payloads begin at the
* address specified in the address msw and lsw (out-of-band).
*/
u32 data_payload_size;
/* Size in bytes of the variable payload accompanying the
* message, or in shared memory. This field is used for parsing the
* parameter payload.
*/
} __packed;
struct asm_stream_param_data_v2 {
u32 module_id;
/* Unique module ID. */
u32 param_id;
/* Unique parameter ID. */
u16 param_size;
/* Data size of the param_id/module_id combination. This is
* a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for future enhancements. This field must be set to
* zero.
*/
} __packed;
#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2
struct asm_stream_cmd_get_pp_params_v2 {
u32 data_payload_addr_lsw;
/* LSW of the parameter data payload address. */
u32 data_payload_addr_msw;
/* MSW of the parameter data payload address.
* - Size of the shared memory, if specified, shall be large enough
* to contain the whole ParamData payload, including Module ID,
* Param ID, Param Size, and Param Values
* - Must be set to zero for in-band data
* - In the case of 32 bit Shared memory address, msw field must be
* set to zero.
* - In the case of 36 bit shared memory address, bit 31 to bit 4 of
* msw must be set to zero.
*/
u32 mem_map_handle;
/* Supported Values: Any.
* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS
* command.
* if mmhandle is NULL, the ParamData payloads in the ACK are within the
* message payload (in-band).
* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the
* address specified in the address msw and lsw.
* (out-of-band).
*/
u32 module_id;
/* Unique module ID. */
u32 param_id;
/* Unique parameter ID. */
u16 param_max_size;
/* Maximum data size of the module_id/param_id combination. This
* is a multiple of 4 bytes.
*/
u16 reserved;
/* Reserved for backward compatibility. Clients must set this
* field to zero.
*/
} __packed;
#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
#define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13
struct asm_bitrate_param {
u32 bitrate;
/* Maximum supported bitrate. Only the AAC encoder is supported.*/
} __packed;
#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
#define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63
/* Flag to turn off both SBR and PS processing, if they are
* present in the bitstream.
*/
#define ASM_AAC_SBR_OFF_PS_OFF (2)
/* Flag to turn on SBR but turn off PS processing,if they are
* present in the bitstream.
*/
#define ASM_AAC_SBR_ON_PS_OFF (1)
/* Flag to turn on both SBR and PS processing, if they are
* present in the bitstream (default behavior).
*/
#define ASM_AAC_SBR_ON_PS_ON (0)
/* Structure for an AAC SBR PS processing flag. */
/* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_aac_sbr_ps_flag_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 sbr_ps_flag;
/* Control parameter to enable or disable SBR/PS processing in
* the AAC bitstream. Use the following macros to set this field:
* - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS
* processing, if they are present in the bitstream.
* - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS
* processing, if they are present in the bitstream.
* - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing,
* if they are present in the bitstream (default behavior).
* - All other values are invalid.
* Changes are applied to the next decoded frame.
*/
} __packed;
#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64
/* First single channel element in a dual mono bitstream.*/
#define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1)
/* Second single channel element in a dual mono bitstream.*/
#define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2)
/* Structure for AAC decoder dual mono channel mapping. */
struct asm_aac_dual_mono_mapping_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
u16 left_channel_sce;
u16 right_channel_sce;
} __packed;
#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4
struct asm_stream_cmdrsp_get_pp_params_v2 {
u32 status;
} __packed;
#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73
/* Enumeration for both vocals in a karaoke stream.*/
#define AC3_KARAOKE_MODE_NO_VOCAL (0)
/* Enumeration for only the left vocal in a karaoke stream.*/
#define AC3_KARAOKE_MODE_LEFT_VOCAL (1)
/* Enumeration for only the right vocal in a karaoke stream.*/
#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2)
/* Enumeration for both vocal channels in a karaoke stream.*/
#define AC3_KARAOKE_MODE_BOTH_VOCAL (3)
#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74
/* Enumeration for the Custom Analog mode.*/
#define AC3_DRC_MODE_CUSTOM_ANALOG (0)
/* Enumeration for the Custom Digital mode.*/
#define AC3_DRC_MODE_CUSTOM_DIGITAL (1)
/* Enumeration for the Line Out mode (light compression).*/
#define AC3_DRC_MODE_LINE_OUT (2)
/* Enumeration for the RF remodulation mode (heavy compression).*/
#define AC3_DRC_MODE_RF_REMOD (3)
#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75
/* Enumeration for playing dual mono in stereo mode.*/
#define AC3_DUAL_MONO_MODE_STEREO (0)
/* Enumeration for playing left mono.*/
#define AC3_DUAL_MONO_MODE_LEFT_MONO (1)
/* Enumeration for playing right mono.*/
#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2)
/* Enumeration for mixing both dual mono channels and playing them.*/
#define AC3_DUAL_MONO_MODE_MIXED_MONO (3)
#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76
/* Enumeration for using the Downmix mode indicated in the bitstream. */
#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0)
/* Enumeration for Surround Compatible mode (preserves the
* surround information).
*/
#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1)
/* Enumeration for Mono Compatible mode (if the output is to be
* further downmixed to mono).
*/
#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2)
/* ID of the AC3 PCM scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78
/* ID of the AC3 DRC boost scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79
/* ID of the AC3 DRC cut scale factor parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A
/* Structure for AC3 Generic Parameter. */
/* Payload of the AC3 parameters in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_ac3_generic_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct asm_enc_cfg_blk_param_v2 encblk;
u32 generic_parameter;
/* AC3 generic parameter. Select from one of the following
* possible values.
*
* For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are:
* - AC3_KARAOKE_MODE_NO_VOCAL
* - AC3_KARAOKE_MODE_LEFT_VOCAL
* - AC3_KARAOKE_MODE_RIGHT_VOCAL
* - AC3_KARAOKE_MODE_BOTH_VOCAL
*
* For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are:
* - AC3_DRC_MODE_CUSTOM_ANALOG
* - AC3_DRC_MODE_CUSTOM_DIGITAL
* - AC3_DRC_MODE_LINE_OUT
* - AC3_DRC_MODE_RF_REMOD
*
* For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are:
* - AC3_DUAL_MONO_MODE_STEREO
* - AC3_DUAL_MONO_MODE_LEFT_MONO
* - AC3_DUAL_MONO_MODE_RIGHT_MONO
* - AC3_DUAL_MONO_MODE_MIXED_MONO
*
* For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are:
* - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT
* - AC3_STEREO_DOWNMIX_MODE_LT_RT
* - AC3_STEREO_DOWNMIX_MODE_LO_RO
*
* For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*
* For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*
* For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are
* 0 to 1 in Q31 format.
*/
} __packed;
/* Enumeration for Raw mode (no downmixing), which specifies
* that all channels in the bitstream are to be played out as is
* without any downmixing. (Default)
*/
#define WMAPRO_CHANNEL_MASK_RAW (-1)
/* Enumeration for setting the channel mask to 0. The 7.1 mode
* (Home Theater) is assigned.
*/
#define WMAPRO_CHANNEL_MASK_ZERO 0x0000
/* Speaker layout mask for one channel (Home Theater, mono).
* - Speaker front center
*/
#define WMAPRO_CHANNEL_MASK_1_C 0x0004
/* Speaker layout mask for two channels (Home Theater, stereo).
* - Speaker front left
* - Speaker front right
*/
#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003
/* Speaker layout mask for three channels (Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
*/
#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007
/* Speaker layout mask for two channels (stereo).
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030
/* Speaker layout mask for four channels.
* - Speaker front left
* - Speaker front right
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033
/* Speaker layout mask for four channels (Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107
/* Speaker layout mask for five channels.
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037
/* Speaker layout mask for five channels (5 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607
/* Speaker layout mask for six channels (5.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back left
* - Speaker back right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F
/* Speaker layout mask for six channels (5.1 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F
/* Speaker layout mask for six channels (5.1 mode, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137
/* Speaker layout mask for six channels (5.1 mode, Home Theater,
* no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707
/* Speaker layout mask for seven channels (6.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back left
* - Speaker back right
* - Speaker back center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F
/* Speaker layout mask for seven channels (6.1 mode, Home
* Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker low frequency
* - Speaker back center
* - Speaker side left
* - Speaker side right
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F
/* Speaker layout mask for seven channels (6.1 mode, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7
/* Speaker layout mask for seven channels (6.1 mode, Home
* Theater, no LFE).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637
/* Speaker layout mask for eight channels (7.1 mode).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker back left
* - Speaker back right
* - Speaker low frequency
* - Speaker front left of center
* - Speaker front right of center
*/
#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \
0x00FF
/* Speaker layout mask for eight channels (7.1 mode, Home Theater).
* - Speaker front left
* - Speaker front right
* - Speaker front center
* - Speaker side left
* - Speaker side right
* - Speaker low frequency
* - Speaker front left of center
* - Speaker front right of center
*
*/
#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \
0x063F
#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82
/* Maximum number of decoder output channels. */
#define MAX_CHAN_MAP_CHANNELS 16
/* Structure for decoder output channel mapping. */
/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the
* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
*/
struct asm_dec_out_chan_map_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
u32 num_channels;
/* Number of decoder output channels.
* Supported values: 0 to #MAX_CHAN_MAP_CHANNELS
*
* A value of 0 indicates native channel mapping, which is valid
* only for NT mode. This means the output of the decoder is to be
* preserved as is.
*/
u8 channel_mapping[MAX_CHAN_MAP_CHANNELS];
} __packed;
#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84
/* Bitmask for the IEC 61937 enable flag.*/
#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL)
/* Shift value for the IEC 61937 enable flag.*/
#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0
/* Bitmask for the IEC 60958 enable flag.*/
#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL)
/* Shift value for the IEC 60958 enable flag.*/
#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1
/* Payload format for open write compressed command */
/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
* command, which opens a stream for a given session ID and stream ID
* to be rendered in the compressed format.
*/
struct asm_stream_cmd_open_write_compressed {
struct apr_hdr hdr;
u32 flags;
/* Mode flags that configure the stream for a specific format.
* Supported values:
* - Bit 0 -- IEC 61937 compatibility
* - 0 -- Stream is not in IEC 61937 format
* - 1 -- Stream is in IEC 61937 format
* - Bit 1 -- IEC 60958 compatibility
* - 0 -- Stream is not in IEC 60958 format
* - 1 -- Stream is in IEC 60958 format
* - Bits 2 to 31 -- 0 (Reserved)
*
* For the same stream, bit 0 cannot be set to 0 and bit 1 cannot
* be set to 1. A compressed stream connot have IEC 60958
* packetization applied without IEC 61937 packetization.
* @note1hang Currently, IEC 60958 packetized input streams are not
* supported.
*/
u32 fmt_id;
/* Specifies the media type of the HDMI stream to be opened.
* Supported values:
* - #ASM_MEDIA_FMT_AC3
* - #ASM_MEDIA_FMT_EAC3
* - #ASM_MEDIA_FMT_DTS
* - #ASM_MEDIA_FMT_ATRAC
* - #ASM_MEDIA_FMT_MAT
*
* @note1hang This field must be set to a valid media type even if
* IEC 61937 packetization is not performed by the aDSP.
*/
} __packed;
/* Indicates the number of samples per channel to be removed from the
* beginning of the stream.
*/
#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67
/* Indicates the number of samples per channel to be removed from
* the end of the stream.
*/
#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
struct asm_data_cmd_remove_silence {
struct apr_hdr hdr;
u32 num_samples_to_remove;
/* < Number of samples per channel to be removed.
* @values 0 to (2@sscr{32}-1)
*/
} __packed;
#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95
struct asm_stream_cmd_open_read_compressed {
struct apr_hdr hdr;
u32 mode_flags;
/* Mode flags that indicate whether meta information per encoded
* frame is to be provided.
* Supported values for bit 4:
* - 0 -- Return data buffer contains all encoded frames only; it does
* not contain frame metadata.
* - 1 -- Return data buffer contains an array of metadata and encoded
* frames.
* - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and
* #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit.
* All other bits are reserved; clients must set them to zero.
*/
u32 frames_per_buf;
/* Indicates the number of frames that need to be returned per
* read buffer
* Supported values: should be greater than 0
*/
} __packed;
/* adsp_asm_stream_commands.h*/
/* adsp_asm_api.h (no changes)*/
#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \
0x00010BE4
#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \
0x00010D83
#define ASM_STREAM_POSTPROCOPO_ID_NONE \
0x00010C68
#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \
0x00010D8B
#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \
ASM_STREAM_POSTPROCOPO_ID_DEFAULT
#define ASM_STREAM_PREPROCOPO_ID_NONE \
ASM_STREAM_POSTPROCOPO_ID_NONE
#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \
0x00010312
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \
0x00010313
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \
0x00010314
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\
0x00010704
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\
0x0001070D
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\
0x0001070E
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\
0x0001070F
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRC_V3 \
0x11000000
#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \
0x0001031B
#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP 0x00010315
#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316
#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO 0x00010715
#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP 0x00010BE3
#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317
#define AUDPROC_MODULE_ID_AIG 0x00010716
#define AUDPROC_PARAM_ID_AIG_ENABLE 0x00010717
#define AUDPROC_PARAM_ID_AIG_CONFIG 0x00010718
struct Audio_AigParam {
uint16_t mode;
/*< Mode word for enabling AIG/SIG mode .
* Byte offset: 0
*/
int16_t staticGainL16Q12;
/*< Static input gain when aigMode is set to 1.
* Byte offset: 2
*/
int16_t initialGainDBL16Q7;
/*<Initial value that the adaptive gain update starts from dB
* Q7 Byte offset: 4
*/
int16_t idealRMSDBL16Q7;
/*<Average RMS level that AIG attempts to achieve Q8.7
* Byte offset: 6
*/
int32_t noiseGateL32;
/*Threshold below which signal is considered as noise and AIG
* Byte offset: 8
*/
int32_t minGainL32Q15;
/*Minimum gain that can be provided by AIG Q16.15
* Byte offset: 12
*/
int32_t maxGainL32Q15;
/*Maximum gain that can be provided by AIG Q16.15
* Byte offset: 16
*/
uint32_t gainAtRtUL32Q31;
/*Attack/release time for AIG update Q1.31
* Byte offset: 20
*/
uint32_t longGainAtRtUL32Q31;
/*Long attack/release time while updating gain for
* noise/silence Q1.31 Byte offset: 24
*/
uint32_t rmsTavUL32Q32;
/* RMS smoothing time constant used for long-term RMS estimate
* Q0.32 Byte offset: 28
*/
uint32_t gainUpdateStartTimMsUL32Q0;
/* The waiting time before which AIG starts to apply adaptive
* gain update Q32.0 Byte offset: 32
*/
} __packed;
#define ADM_MODULE_ID_EANS 0x00010C4A
#define ADM_PARAM_ID_EANS_ENABLE 0x00010C4B
#define ADM_PARAM_ID_EANS_PARAMS 0x00010C4C
struct adm_eans_enable {
uint32_t enable_flag;
/*< Specifies whether EANS is disabled (0) or enabled
* (nonzero).
* This is supported only for sampling rates of 8, 12, 16, 24, 32,
* and 48 kHz. It is not supported for sampling rates of 11.025,
* 22.05, or 44.1 kHz.
*/
} __packed;
struct adm_eans_params {
int16_t eans_mode;
/*< Mode word for enabling/disabling submodules.
* Byte offset: 0
*/
int16_t eans_input_gain;
/*< Q2.13 input gain to the EANS module.
* Byte offset: 2
*/
int16_t eans_output_gain;
/*< Q2.13 output gain to the EANS module.
* Byte offset: 4
*/
int16_t eansarget_ns;
/*< Target noise suppression level in dB.
* Byte offset: 6
*/
int16_t eans_s_alpha;
/*< Q3.12 over-subtraction factor for stationary noise
* suppression.
* Byte offset: 8
*/
int16_t eans_n_alpha;
/* < Q3.12 over-subtraction factor for nonstationary noise
* suppression.
* Byte offset: 10
*/
int16_t eans_n_alphamax;
/*< Q3.12 maximum over-subtraction factor for nonstationary
* noise suppression.
* Byte offset: 12
*/
int16_t eans_e_alpha;
/*< Q15 scaling factor for excess noise suppression.
* Byte offset: 14
*/
int16_t eans_ns_snrmax;
/*< Upper boundary in dB for SNR estimation.
* Byte offset: 16
*/
int16_t eans_sns_block;
/*< Quarter block size for stationary noise suppression.
* Byte offset: 18
*/
int16_t eans_ns_i;
/*< Initialization block size for noise suppression.
* Byte offset: 20
*/
int16_t eans_np_scale;
/*< Power scale factor for nonstationary noise update.
* Byte offset: 22
*/
int16_t eans_n_lambda;
/*< Smoothing factor for higher level nonstationary noise
* update.
* Byte offset: 24
*/
int16_t eans_n_lambdaf;
/*< Medium averaging factor for noise update.
* Byte offset: 26
*/
int16_t eans_gs_bias;
/*< Bias factor in dB for gain calculation.
* Byte offset: 28
*/
int16_t eans_gs_max;
/*< SNR lower boundary in dB for aggressive gain calculation.
* Byte offset: 30
*/
int16_t eans_s_alpha_hb;
/*< Q3.12 over-subtraction factor for high-band stationary
* noise suppression.
* Byte offset: 32
*/
int16_t eans_n_alphamax_hb;
/*< Q3.12 maximum over-subtraction factor for high-band
* nonstationary noise suppression.
* Byte offset: 34
*/
int16_t eans_e_alpha_hb;
/*< Q15 scaling factor for high-band excess noise suppression.
* Byte offset: 36
*/
int16_t eans_n_lambda0;
/*< Smoothing factor for nonstationary noise update during
* speech activity.
* Byte offset: 38
*/
int16_t thresh;
/*< Threshold for generating a binary VAD decision.
* Byte offset: 40
*/
int16_t pwr_scale;
/*< Indirect lower boundary of the noise level estimate.
* Byte offset: 42
*/
int16_t hangover_max;
/*< Avoids mid-speech clipping and reliably detects weak speech
* bursts at the end of speech activity.
* Byte offset: 44
*/
int16_t alpha_snr;
/*< Controls responsiveness of the VAD.
* Byte offset: 46
*/
int16_t snr_diff_max;
/*< Maximum SNR difference. Decreasing this parameter value may
* help in making correct decisions during abrupt changes; however,
* decreasing too much may increase false alarms during long
* pauses/silences.
* Byte offset: 48
*/
int16_t snr_diff_min;
/*< Minimum SNR difference. Decreasing this parameter value may
* help in making correct decisions during abrupt changes; however,
* decreasing too much may increase false alarms during long
* pauses/silences.
* Byte offset: 50
*/
int16_t init_length;
/*< Defines the number of frames for which a noise level
* estimate is set to a fixed value.
* Byte offset: 52
*/
int16_t max_val;
/*< Defines the upper limit of the noise level.
* Byte offset: 54
*/
int16_t init_bound;
/*< Defines the initial bounding value for the noise level
* estimate. This is used during the initial segment defined by the
* init_length parameter.
* Byte offset: 56
*/
int16_t reset_bound;
/*< Reset boundary for noise tracking.
* Byte offset: 58
*/
int16_t avar_scale;
/*< Defines the bias factor in noise estimation.
* Byte offset: 60
*/
int16_t sub_nc;
/*< Defines the window length for noise estimation.
* Byte offset: 62
*/
int16_t spow_min;
/*< Defines the minimum signal power required to update the
* boundaries for the noise floor estimate.
* Byte offset: 64
*/
int16_t eans_gs_fast;
/*< Fast smoothing factor for postprocessor gain.
* Byte offset: 66
*/
int16_t eans_gs_med;
/*< Medium smoothing factor for postprocessor gain.
* Byte offset: 68
*/
int16_t eans_gs_slow;
/*< Slow smoothing factor for postprocessor gain.
* Byte offset: 70
*/
int16_t eans_swb_salpha;
/*< Q3.12 super wideband aggressiveness factor for stationary
* noise suppression.
* Byte offset: 72
*/
int16_t eans_swb_nalpha;
/*< Q3.12 super wideband aggressiveness factor for
* nonstationary noise suppression.
* Byte offset: 74
*/
} __packed;
#define ADM_MODULE_IDX_MIC_GAIN_CTRL 0x00010C35
/* @addtogroup audio_pp_param_ids
* ID of the Tx mic gain control parameter used by the
* #ADM_MODULE_IDX_MIC_GAIN_CTRL module.
* @messagepayload
* @structure{admx_mic_gain}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex}
*/
#define ADM_PARAM_IDX_MIC_GAIN 0x00010C36
/* Structure for a Tx mic gain parameter for the mic gain
* control module.
*/
/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the
* Tx Mic Gain Control module.
*/
struct admx_mic_gain {
uint16_t tx_mic_gain;
/*< Linear gain in Q13 format. */
uint16_t reserved;
/*< Clients must set this field to zero. */
} __packed;
struct adm_set_mic_gain_params {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
struct admx_mic_gain mic_gain_data;
} __packed;
/* end_addtogroup audio_pp_param_ids */
/* @ingroup audio_pp_module_ids
* ID of the Rx Codec Gain Control module.
*
* This module supports the following parameter ID:
* - #ADM_PARAM_ID_RX_CODEC_GAIN
*/
#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL 0x00010C37
/* @addtogroup audio_pp_param_ids
* ID of the Rx codec gain control parameter used by the
* #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module.
*
* @messagepayload
* @structure{adm_rx_codec_gain}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex}
*/
#define ADM_PARAM_ID_RX_CODEC_GAIN 0x00010C38
/* Structure for the Rx common codec gain control module. */
/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter
* in the Rx Codec Gain Control module.
*/
struct adm_rx_codec_gain {
uint16_t rx_codec_gain;
/* Linear gain in Q13 format. */
uint16_t reserved;
/* Clients must set this field to zero.*/
} __packed;
/* end_addtogroup audio_pp_param_ids */
/* @ingroup audio_pp_module_ids
* ID of the HPF Tuning Filter module on the Tx path.
* This module supports the following parameter IDs:
* - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
* - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN
* - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
*/
#define ADM_MODULE_ID_HPF_IIRX_FILTER 0x00010C3D
/* @addtogroup audio_pp_param_ids */
/* ID of the Tx HPF IIR filter enable parameter used by the
* #ADM_MODULE_ID_HPF_IIRX_FILTER module.
* @parspace Message payload
* @structure{adm_hpfx_iir_filter_enable_cfg}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex}
*/
#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG 0x00010C3E
/* ID of the Tx HPF IIR filter pregain parameter used by the
* #ADM_MODULE_ID_HPF_IIRX_FILTER module.
* @parspace Message payload
* @structure{adm_hpfx_iir_filter_pre_gain}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex}
*/
#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN 0x00010C3F
/* ID of the Tx HPF IIR filter configuration parameters used by the
* #ADM_MODULE_ID_HPF_IIRX_FILTER module.
* @parspace Message payload
* @structure{adm_hpfx_iir_filter_cfg_params}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA
* RAMS.tex}
*/
#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS 0x00010C40
/* Structure for enabling a configuration parameter for
* the HPF IIR tuning filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
* parameter in the Tx path HPF Tuning Filter module.
*/
struct adm_hpfx_iir_filter_enable_cfg {
uint32_t enable_flag;
/* Specifies whether the HPF tuning filter is disabled (0) or
* enabled (nonzero).
*/
} __packed;
/* Structure for the pregain parameter for the HPF
* IIR tuning filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter
* in the Tx path HPF Tuning Filter module.
*/
struct adm_hpfx_iir_filter_pre_gain {
uint16_t pre_gain;
/* Linear gain in Q13 format. */
uint16_t reserved;
/* Clients must set this field to zero.*/
} __packed;
/* Structure for the configuration parameter for the
* HPF IIR tuning filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
* parameters in the Tx path HPF Tuning Filter module. \n
* \n
* This structure is followed by tuning filter coefficients as follows: \n
* - Sequence of int32_t FilterCoeffs.
* Each band has five coefficients, each in int32_t format in the order of
* b0, b1, b2, a1, a2.
* - Sequence of int16_t NumShiftFactor.
* One int16_t per band. The numerator shift factor is related to the Q
* factor of the filter coefficients.
* - Sequence of uint16_t PanSetting.
* One uint16_t for each band to indicate application of the filter to
* left (0), right (1), or both (2) channels.
*/
struct adm_hpfx_iir_filter_cfg_params {
uint16_t num_biquad_stages;
/*< Number of bands.
* Supported values: 0 to 20
*/
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* end_addtogroup audio_pp_module_ids */
/* @addtogroup audio_pp_module_ids */
/* ID of the Tx path IIR Tuning Filter module.
* This module supports the following parameter IDs:
* - #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
*/
#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41
/* ID of the Rx path IIR Tuning Filter module for the left channel.
* The parameter IDs of the IIR tuning filter module
* (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning
* filter.
*
* Pan parameters are not required for this per-channel IIR filter; the pan
* parameters are ignored by this module.
*/
#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER 0x00010705
/* ID of the the Rx path IIR Tuning Filter module for the right
* channel.
* The parameter IDs of the IIR tuning filter module
* (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx
* tuning filter.
*
* Pan parameters are not required for this per-channel IIR filter;
* the pan parameters are ignored by this module.
*/
#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER 0x00010706
/* end_addtogroup audio_pp_module_ids */
/* @addtogroup audio_pp_param_ids */
/* ID of the Tx IIR filter enable parameter used by the
* #ADM_MODULE_IDX_IIR_FILTER module.
* @parspace Message payload
* @structure{admx_iir_filter_enable_cfg}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex}
*/
#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG 0x00010C42
/* ID of the Tx IIR filter pregain parameter used by the
* #ADM_MODULE_IDX_IIR_FILTER module.
* @parspace Message payload
* @structure{admx_iir_filter_pre_gain}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex}
*/
#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN 0x00010C43
/* ID of the Tx IIR filter configuration parameters used by the
* #ADM_MODULE_IDX_IIR_FILTER module.
* @parspace Message payload
* @structure{admx_iir_filter_cfg_params}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex}
*/
#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS 0x00010C44
/* Structure for enabling the configuration parameter for the
* IIR filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
* parameter in the Tx Path IIR Tuning Filter module.
*/
struct admx_iir_filter_enable_cfg {
uint32_t enable_flag;
/*< Specifies whether the IIR tuning filter is disabled (0) or
* enabled (nonzero).
*/
} __packed;
/* Structure for the pregain parameter for the
* IIR filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN
* parameter in the Tx Path IIR Tuning Filter module.
*/
struct admx_iir_filter_pre_gain {
uint16_t pre_gain;
/*< Linear gain in Q13 format. */
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* Structure for the configuration parameter for the
* IIR filter module on the Tx path.
*/
/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS
* parameter in the Tx Path IIR Tuning Filter module. \n
* \n
* This structure is followed by the HPF IIR filter coefficients on
* the Tx path as follows: \n
* - Sequence of int32_t ulFilterCoeffs. Each band has five
* coefficients, each in int32_t format in the order of b0, b1, b2,
* a1, a2.
* - Sequence of int16_t sNumShiftFactor. One int16_t per band. The
* numerator shift factor is related to the Q factor of the filter
* coefficients.
* - Sequence of uint16_t usPanSetting. One uint16_t for each band
* to indicate if the filter is applied to left (0), right (1), or
* both (2) channels.
*/
struct admx_iir_filter_cfg_params {
uint16_t num_biquad_stages;
/*< Number of bands.
* Supported values: 0 to 20
*/
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* end_addtogroup audio_pp_module_ids */
/* @ingroup audio_pp_module_ids
* ID of the QEnsemble module.
* This module supports the following parameter IDs:
* - #ADM_PARAM_ID_QENSEMBLE_ENABLE
* - #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
* - #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
*/
#define ADM_MODULE_ID_QENSEMBLE 0x00010C59
/* @addtogroup audio_pp_param_ids */
/* ID of the QEnsemble enable parameter used by the
* #ADM_MODULE_ID_QENSEMBLE module.
* @messagepayload
* @structure{adm_qensemble_enable}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex}
*/
#define ADM_PARAM_ID_QENSEMBLE_ENABLE 0x00010C60
/* ID of the QEnsemble back gain parameter used by the
* #ADM_MODULE_ID_QENSEMBLE module.
* @messagepayload
* @structure{adm_qensemble_param_backgain}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex}
*/
#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN 0x00010C61
/* ID of the QEnsemble new angle parameter used by the
* #ADM_MODULE_ID_QENSEMBLE module.
* @messagepayload
* @structure{adm_qensemble_param_set_new_angle}
* @tablespace
* @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex}
*/
#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE 0x00010C62
/* Structure for enabling the configuration parameter for the
* QEnsemble module.
*/
/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE
* parameter used by the QEnsemble module.
*/
struct adm_qensemble_enable {
uint32_t enable_flag;
/*< Specifies whether the QEnsemble module is disabled (0) or enabled
* (nonzero).
*/
} __packed;
/* Structure for the background gain for the QEnsemble module. */
/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
* parameter used by
* the QEnsemble module.
*/
struct adm_qensemble_param_backgain {
int16_t back_gain;
/*< Linear gain in Q15 format.
* Supported values: 0 to 32767
*/
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* Structure for setting a new angle for the QEnsemble module. */
/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
* parameter used
* by the QEnsemble module.
*/
struct adm_qensemble_param_set_new_angle {
int16_t new_angle;
/*< New angle in degrees.
* Supported values: 0 to 359
*/
int16_t time_ms;
/*< Transition time in milliseconds to set the new angle.
* Supported values: 0 to 32767
*/
} __packed;
#define ADM_CMD_GET_PP_TOPO_MODULE_LIST 0x00010349
#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST 0x00010350
#define AUDPROC_PARAM_ID_ENABLE 0x00010904
/*
* Payload of the ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
*/
struct adm_cmd_get_pp_topo_module_list_t {
struct apr_hdr hdr;
/* Lower 32 bits of the 64-bit parameter data payload address. */
uint32_t data_payload_addr_lsw;
/*
* Upper 32 bits of the 64-bit parameter data payload address.
*
*
* The size of the shared memory, if specified, must be large enough to
* contain the entire parameter data payload, including the module ID,
* parameter ID, parameter size, and parameter values.
*/
uint32_t data_payload_addr_msw;
/*
* Unique identifier for an address.
*
* This memory map handle is returned by the aDSP through the
* #ADM_CMD_SHARED_MEM_MAP_REGIONS command.
*
* @values
* - Non-NULL -- On acknowledgment, the parameter data payloads begin at
* the address specified (out-of-band)
* - NULL -- The acknowledgment's payload contains the parameter data
* (in-band) @tablebulletend
*/
uint32_t mem_map_handle;
/*
* Maximum data size of the list of modules. This
* field is a multiple of 4 bytes.
*/
uint16_t param_max_size;
/* This field must be set to zero. */
uint16_t reserved;
} __packed;
/*
* Payload of the ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST message, which returns
* module ids in response to an ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
* Immediately following this structure is the acknowledgment <b>module id
* data variable payload</b> containing the pre/postprocessing module id
* values. For an in-band scenario, the variable payload depends on the size
* of the parameter.
*/
struct adm_cmd_rsp_get_pp_topo_module_list_t {
/* Status message (error code). */
uint32_t status;
} __packed;
struct audproc_topology_module_id_info_t {
uint32_t num_modules;
} __packed;
/* end_addtogroup audio_pp_module_ids */
/* @ingroup audio_pp_module_ids
* ID of the Volume Control module pre/postprocessing block.
* This module supports the following parameter IDs:
* - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
* - #ASM_PARAM_ID_MULTICHANNEL_GAIN
* - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
* - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
* - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
* - #ASM_PARAM_ID_MULTICHANNEL_GAIN
* - #ASM_PARAM_ID_MULTICHANNEL_MUTE
*/
#define ASM_MODULE_ID_VOL_CTRL 0x00010BFE
#define ASM_MODULE_ID_VOL_CTRL2 0x00010910
#define AUDPROC_MODULE_ID_VOL_CTRL ASM_MODULE_ID_VOL_CTRL
/* @addtogroup audio_pp_param_ids */
/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL
* module.
* @messagepayload
* @structure{asm_volume_ctrl_master_gain}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex}
*/
#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN 0x00010BFF
#define AUDPROC_PARAM_ID_VOL_CTRL_MASTER_GAIN ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
/* ID of the left/right channel gain parameter used by the
* #ASM_MODULE_ID_VOL_CTRL module.
* @messagepayload
* @structure{asm_volume_ctrl_lr_chan_gain}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MULTICHANNEL_GAIN.tex}
*/
#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN 0x00010C00
/* ID of the mute configuration parameter used by the
* #ASM_MODULE_ID_VOL_CTRL module.
* @messagepayload
* @structure{asm_volume_ctrl_mute_config}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex}
*/
#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG 0x00010C01
/* ID of the soft stepping volume parameters used by the
* #ASM_MODULE_ID_VOL_CTRL module.
* @messagepayload
* @structure{asm_soft_step_volume_params}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET
* ERS.tex}
*/
#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS 0x00010C29
#define AUDPROC_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS\
ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL
* module.
*/
#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A
/* ID of the multiple-channel volume control parameters used by the
* #ASM_MODULE_ID_VOL_CTRL module.
*/
#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713
/* ID of the multiple-channel mute configuration parameters used by the
* #ASM_MODULE_ID_VOL_CTRL module.
*/
#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714
/* Structure for the master gain parameter for a volume control
* module.
*/
/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
* parameter used by the Volume Control module.
*/
struct asm_volume_ctrl_master_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint16_t master_gain;
/* Linear gain in Q13 format. */
uint16_t reserved;
/* Clients must set this field to zero. */
} __packed;
struct asm_volume_ctrl_lr_chan_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint16_t l_chan_gain;
/*< Linear gain in Q13 format for the left channel. */
uint16_t r_chan_gain;
/*< Linear gain in Q13 format for the right channel.*/
} __packed;
/* Structure for the mute configuration parameter for a
* volume control module.
*/
/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
* parameter used by the Volume Control module.
*/
struct asm_volume_ctrl_mute_config {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t mute_flag;
/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/
} __packed;
/*
* Supported parameters for a soft stepping linear ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0
/*
* Exponential ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1
/*
* Logarithmic ramping curve.
*/
#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2
/* Structure for holding soft stepping volume parameters. */
/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
* parameters used by the Volume Control module.
*/
struct asm_soft_step_volume_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t period;
/*< Period in milliseconds.
* Supported values: 0 to 15000
*/
uint32_t step;
/*< Step in microseconds.
* Supported values: 0 to 15000000
*/
uint32_t ramping_curve;
/*< Ramping curve type.
* Supported values:
* - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
* - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
* - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
*/
} __packed;
/* Structure for holding soft pause parameters. */
/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
* parameters used by the Volume Control module.
*/
struct asm_soft_pause_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t enable_flag;
/*< Specifies whether soft pause is disabled (0) or enabled
* (nonzero).
*/
uint32_t period;
/*< Period in milliseconds.
* Supported values: 0 to 15000
*/
uint32_t step;
/*< Step in microseconds.
* Supported values: 0 to 15000000
*/
uint32_t ramping_curve;
/*< Ramping curve.
* Supported values:
* - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
* - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
* - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
*/
} __packed;
/* Maximum number of channels.*/
#define VOLUME_CONTROL_MAX_CHANNELS 8
/* Structure for holding one channel type - gain pair. */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel
* type/gain pairs used by the Volume Control module. \n \n This
* structure immediately follows the
* asm_volume_ctrl_multichannel_gain structure.
*/
struct asm_volume_ctrl_channeltype_gain_pair {
uint8_t channeltype;
/*
* Channel type for which the gain setting is to be applied.
* Supported values:
* - #PCM_CHANNEL_L
* - #PCM_CHANNEL_R
* - #PCM_CHANNEL_C
* - #PCM_CHANNEL_LS
* - #PCM_CHANNEL_RS
* - #PCM_CHANNEL_LFE
* - #PCM_CHANNEL_CS
* - #PCM_CHANNEL_LB
* - #PCM_CHANNEL_RB
* - #PCM_CHANNELS
* - #PCM_CHANNEL_CVH
* - #PCM_CHANNEL_MS
* - #PCM_CHANNEL_FLC
* - #PCM_CHANNEL_FRC
* - #PCM_CHANNEL_RLC
* - #PCM_CHANNEL_RRC
*/
uint8_t reserved1;
/* Clients must set this field to zero. */
uint8_t reserved2;
/* Clients must set this field to zero. */
uint8_t reserved3;
/* Clients must set this field to zero. */
uint32_t gain;
/*
* Gain value for this channel in Q28 format.
* Supported values: Any
*/
} __packed;
/* Structure for the multichannel gain command */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN
* parameters used by the Volume Control module.
*/
struct asm_volume_ctrl_multichannel_gain {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t num_channels;
/*
* Number of channels for which gain values are provided. Any
* channels present in the data for which gain is not provided are
* set to unity gain.
* Supported values: 1 to 8
*/
struct asm_volume_ctrl_channeltype_gain_pair
gain_data[VOLUME_CONTROL_MAX_CHANNELS];
/* Array of channel type/gain pairs.*/
} __packed;
/* Structure for holding one channel type - mute pair. */
/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel
* type/mute setting pairs used by the Volume Control module. \n \n
* This structure immediately follows the
* asm_volume_ctrl_multichannel_mute structure.
*/
struct asm_volume_ctrl_channelype_mute_pair {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint8_t channelype;
/*< Channel type for which the mute setting is to be applied.
* Supported values:
* - #PCM_CHANNEL_L
* - #PCM_CHANNEL_R
* - #PCM_CHANNEL_C
* - #PCM_CHANNEL_LS
* - #PCM_CHANNEL_RS
* - #PCM_CHANNEL_LFE
* - #PCM_CHANNEL_CS
* - #PCM_CHANNEL_LB
* - #PCM_CHANNEL_RB
* - #PCM_CHANNELS
* - #PCM_CHANNEL_CVH
* - #PCM_CHANNEL_MS
* - #PCM_CHANNEL_FLC
* - #PCM_CHANNEL_FRC
* - #PCM_CHANNEL_RLC
* - #PCM_CHANNEL_RRC
*/
uint8_t reserved1;
/*< Clients must set this field to zero. */
uint8_t reserved2;
/*< Clients must set this field to zero. */
uint8_t reserved3;
/*< Clients must set this field to zero. */
uint32_t mute;
/*< Mute setting for this channel.
* Supported values:
* - 0 = Unmute
* - Nonzero = Mute
*/
} __packed;
/* Structure for the multichannel mute command */
/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE
* parameters used by the Volume Control module.
*/
struct asm_volume_ctrl_multichannel_mute {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t num_channels;
/*< Number of channels for which mute configuration is
* provided. Any channels present in the data for which mute
* configuration is not provided are set to unmute.
* Supported values: 1 to 8
*/
struct asm_volume_ctrl_channelype_mute_pair
mute_data[VOLUME_CONTROL_MAX_CHANNELS];
/*< Array of channel type/mute setting pairs.*/
} __packed;
/* end_addtogroup audio_pp_param_ids */
/* audio_pp_module_ids
* ID of the IIR Tuning Filter module.
* This module supports the following parameter IDs:
* - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
* - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
* - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
*/
#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02
/* @addtogroup audio_pp_param_ids */
/* ID of the IIR tuning filter enable parameter used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
* @messagepayload
* @structure{asm_iiruning_filter_enable}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO
* NFIG.tex}
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03
/* ID of the IIR tuning filter pregain parameter used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04
/* ID of the IIR tuning filter configuration parameters used by the
* #ASM_MODULE_ID_IIRUNING_FILTER module.
*/
#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05
/* Structure for an enable configuration parameter for an
* IIR tuning filter module.
*/
/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
* parameter used by the IIR Tuning Filter module.
*/
struct asm_iiruning_filter_enable {
uint32_t enable_flag;
/*< Specifies whether the IIR tuning filter is disabled (0) or
* enabled (1).
*/
} __packed;
/* Structure for the pregain parameter for an IIR tuning filter module. */
/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
* parameters used by the IIR Tuning Filter module.
*/
struct asm_iiruning_filter_pregain {
uint16_t pregain;
/*< Linear gain in Q13 format. */
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* Structure for the configuration parameter for an IIR tuning filter
* module.
*/
/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
* parameters used by the IIR Tuning Filter module. \n
* \n
* This structure is followed by the IIR filter coefficients: \n
* - Sequence of int32_t FilterCoeffs \n
* Five coefficients for each band. Each coefficient is in int32_t format, in
* the order of b0, b1, b2, a1, a2.
* - Sequence of int16_t NumShiftFactor \n
* One int16_t per band. The numerator shift factor is related to the Q
* factor of the filter coefficients.
* - Sequence of uint16_t PanSetting \n
* One uint16_t per band, indicating if the filter is applied to left (0),
* right (1), or both (2) channels.
*/
struct asm_iir_filter_config_params {
uint16_t num_biquad_stages;
/*< Number of bands.
* Supported values: 0 to 20
*/
uint16_t reserved;
/*< Clients must set this field to zero.*/
} __packed;
/* audio_pp_module_ids
* ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx
* paths.
* This module supports the following parameter IDs:
* - #ASM_PARAM_ID_MBDRC_ENABLE
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
*/
#define ASM_MODULE_ID_MBDRC 0x00010C06
/* audio_pp_param_ids */
/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module.
* @messagepayload
* @structure{asm_mbdrc_enable}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex}
*/
#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07
/* ID of the MBDRC configuration parameters used by the
* #ASM_MODULE_ID_MBDRC module.
* @messagepayload
* @structure{asm_mbdrc_config_params}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex}
*
* @parspace Sub-band DRC configuration parameters
* @structure{asm_subband_drc_config_params}
* @tablespace
* @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex}
*
* @keep{6}
* To obtain legacy ADRC from MBDRC, use the calibration tool to:
*
* - Enable MBDRC (EnableFlag = TRUE)
* - Set number of bands to 1 (uiNumBands = 1)
* - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1)
* - Clear the first band mute flag (MuteFlag[0] = 0)
* - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000)
* - Use the legacy ADRC parameters to calibrate the rest of the MBDRC
* parameters.
*/
#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08
/* end_addtogroup audio_pp_param_ids */
/* audio_pp_module_ids
* ID of the MMBDRC module version 2 pre/postprocessing block.
* This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in
* the length of the filters used in each sub-band.
* This module supports the following parameter ID:
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2
*/
#define ASM_MODULE_ID_MBDRCV2 0x0001070B
/* @addtogroup audio_pp_param_ids */
/* ID of the configuration parameters used by the
* #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure
* of the MBDRC v2 pre/postprocessing block.
* The update to this configuration structure from the original
* MBDRC is the number of filter coefficients in the filter
* structure. The sequence for is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t
* padding
* - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags +
* uint16_t padding
* This block uses the same parameter structure as
* #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.
*/
#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \
0x0001070C
#define ASM_MODULE_ID_MBDRCV3 0x0001090B
/*
* ID of the MMBDRC module version 3 pre/postprocessing block.
* This module differs from MBDRCv2 (#ASM_MODULE_ID_MBDRCV2) in
* that it supports both 16- and 24-bit data.
* This module supports the following parameter ID:
* - #ASM_PARAM_ID_MBDRC_ENABLE
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
* - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_V3
* - #ASM_PARAM_ID_MBDRC_FILTER_XOVER_FREQS
*/
/* Structure for the enable parameter for an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the
* MBDRC module.
*/
struct asm_mbdrc_enable {
uint32_t enable_flag;
/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/
} __packed;
/* Structure for the configuration parameters for an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
* parameters used by the MBDRC module. \n \n Following this
* structure is the payload for sub-band DRC configuration
* parameters (asm_subband_drc_config_params). This sub-band
* structure must be repeated for each band.
*/
struct asm_mbdrc_config_params {
uint16_t num_bands;
/*< Number of bands.
* Supported values: 1 to 5
*/
int16_t limiterhreshold;
/*< Threshold in decibels for the limiter output.
* Supported values: -72 to 18 \n
* Recommended value: 3994 (-0.22 db in Q3.12 format)
*/
int16_t limiter_makeup_gain;
/*< Makeup gain in decibels for the limiter output.
* Supported values: -42 to 42 \n
* Recommended value: 256 (0 dB in Q7.8 format)
*/
int16_t limiter_gc;
/*< Limiter gain recovery coefficient.
* Supported values: 0.5 to 0.99 \n
* Recommended value: 32440 (0.99 in Q15 format)
*/
int16_t limiter_delay;
/*< Limiter delay in samples.
* Supported values: 0 to 10 \n
* Recommended value: 262 (0.008 samples in Q15 format)
*/
int16_t limiter_max_wait;
/*< Maximum limiter waiting time in samples.
* Supported values: 0 to 10 \n
* Recommended value: 262 (0.008 samples in Q15 format)
*/
} __packed;
/* DRC configuration structure for each sub-band of an MBDRC module. */
/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC
* configuration parameters for each sub-band in the MBDRC module.
* After this DRC structure is configured for valid bands, the next
* MBDRC setparams expects the sequence of sub-band MBDRC filter
* coefficients (the length depends on the number of bands) plus the
* mute flag for that band plus uint16_t padding.
*
* @keep{10}
* The filter coefficient and mute flag are of type int16_t:
* - FIR coefficient = int16_t firFilter
* - Mute flag = int16_t fMuteFlag
*
* The sequence is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding
* - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding
*
* For improved filterbank, the sequence is as follows:
* - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
* - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
* - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
* - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding
* - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding
*/
struct asm_subband_drc_config_params {
int16_t drc_stereo_linked_flag;
/*< Specifies whether all stereo channels have the same applied
* dynamics (1) or if they process their dynamics independently (0).
* Supported values:
* - 0 -- Not linked
* - 1 -- Linked
*/
int16_t drc_mode;
/*< Specifies whether DRC mode is bypassed for sub-bands.
* Supported values:
* - 0 -- Disabled
* - 1 -- Enabled
*/
int16_t drc_down_sample_level;
/*< DRC down sample level.
* Supported values: @ge 1
*/
int16_t drc_delay;
/*< DRC delay in samples.
* Supported values: 0 to 1200
*/
uint16_t drc_rmsime_avg_const;
/*< RMS signal energy time-averaging constant.
* Supported values: 0 to 2^16-1
*/
uint16_t drc_makeup_gain;
/*< DRC makeup gain in decibels.
* Supported values: 258 to 64917
*/
/* Down expander settings */
int16_t down_expdrhreshold;
/*< Down expander threshold.
* Supported Q7 format values: 1320 to up_cmpsrhreshold
*/
int16_t down_expdr_slope;
/*< Down expander slope.
* Supported Q8 format values: -32768 to 0.
*/
uint32_t down_expdr_attack;
/*< Down expander attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t down_expdr_release;
/*< Down expander release constant.
* Supported Q31 format values: 19685 to 2^31
*/
uint16_t down_expdr_hysteresis;
/*< Down expander hysteresis constant.
* Supported Q14 format values: 1 to 32690
*/
uint16_t reserved;
/*< Clients must set this field to zero. */
int32_t down_expdr_min_gain_db;
/*< Down expander minimum gain.
* Supported Q23 format values: -805306368 to 0.
*/
/* Up compressor settings */
int16_t up_cmpsrhreshold;
/*< Up compressor threshold.
* Supported Q7 format values: down_expdrhreshold to
* down_cmpsrhreshold.
*/
uint16_t up_cmpsr_slope;
/*< Up compressor slope.
* Supported Q16 format values: 0 to 64881.
*/
uint32_t up_cmpsr_attack;
/*< Up compressor attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t up_cmpsr_release;
/*< Up compressor release constant.
* Supported Q31 format values: 19685 to 2^31.
*/
uint16_t up_cmpsr_hysteresis;
/*< Up compressor hysteresis constant.
* Supported Q14 format values: 1 to 32690.
*/
/* Down compressor settings */
int16_t down_cmpsrhreshold;
/*< Down compressor threshold.
* Supported Q7 format values: up_cmpsrhreshold to 11560.
*/
uint16_t down_cmpsr_slope;
/*< Down compressor slope.
* Supported Q16 format values: 0 to 64881.
*/
uint16_t reserved1;
/*< Clients must set this field to zero. */
uint32_t down_cmpsr_attack;
/*< Down compressor attack constant.
* Supported Q31 format values: 196844 to 2^31.
*/
uint32_t down_cmpsr_release;
/*< Down compressor release constant.
* Supported Q31 format values: 19685 to 2^31.
*/
uint16_t down_cmpsr_hysteresis;
/*< Down compressor hysteresis constant.
* Supported Q14 values: 1 to 32690.
*/
uint16_t reserved2;
/*< Clients must set this field to zero.*/
} __packed;
#define ASM_MODULE_ID_EQUALIZER 0x00010C27
#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28
#define ASM_MAX_EQ_BANDS 12
struct asm_eq_per_band_params {
uint32_t band_idx;
/*< Band index.
* Supported values: 0 to 11
*/
uint32_t filterype;
/*< Type of filter.
* Supported values:
* - #ASM_PARAM_EQYPE_NONE
* - #ASM_PARAM_EQ_BASS_BOOST
* - #ASM_PARAM_EQ_BASS_CUT
* - #ASM_PARAM_EQREBLE_BOOST
* - #ASM_PARAM_EQREBLE_CUT
* - #ASM_PARAM_EQ_BAND_BOOST
* - #ASM_PARAM_EQ_BAND_CUT
*/
uint32_t center_freq_hz;
/*< Filter band center frequency in Hertz. */
int32_t filter_gain;
/*< Filter band initial gain.
* Supported values: +12 to -12 dB in 1 dB increments
*/
int32_t q_factor;
/*< Filter band quality factor expressed as a Q8 number, i.e., a
* fixed-point number with q factor of 8. For example, 3000/(2^8).
*/
} __packed;
struct asm_eq_params {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
uint32_t enable_flag;
/*< Specifies whether the equalizer module is disabled (0) or enabled
* (nonzero).
*/
uint32_t num_bands;
/*< Number of bands.
* Supported values: 1 to 12
*/
struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS];
} __packed;
/* No equalizer effect.*/
#define ASM_PARAM_EQYPE_NONE 0
/* Bass boost equalizer effect.*/
#define ASM_PARAM_EQ_BASS_BOOST 1
/*Bass cut equalizer effect.*/
#define ASM_PARAM_EQ_BASS_CUT 2
/* Treble boost equalizer effect */
#define ASM_PARAM_EQREBLE_BOOST 3
/* Treble cut equalizer effect.*/
#define ASM_PARAM_EQREBLE_CUT 4
/* Band boost equalizer effect.*/
#define ASM_PARAM_EQ_BAND_BOOST 5
/* Band cut equalizer effect.*/
#define ASM_PARAM_EQ_BAND_CUT 6
/* Get & set params */
#define VSS_ICOMMON_CMD_SET_PARAM_V2 0x0001133D
#define VSS_ICOMMON_CMD_GET_PARAM_V2 0x0001133E
#define VSS_ICOMMON_RSP_GET_PARAM 0x00011008
/* ID of the Bass Boost module.
* This module supports the following parameter IDs:
* - #AUDPROC_PARAM_ID_BASS_BOOST_ENABLE
* - #AUDPROC_PARAM_ID_BASS_BOOST_MODE
* - #AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH
*/
#define AUDPROC_MODULE_ID_BASS_BOOST 0x000108A1
/* ID of the Bass Boost enable parameter used by
* AUDPROC_MODULE_ID_BASS_BOOST.
*/
#define AUDPROC_PARAM_ID_BASS_BOOST_ENABLE 0x000108A2
/* ID of the Bass Boost mode parameter used by
* AUDPROC_MODULE_ID_BASS_BOOST.
*/
#define AUDPROC_PARAM_ID_BASS_BOOST_MODE 0x000108A3
/* ID of the Bass Boost strength parameter used by
* AUDPROC_MODULE_ID_BASS_BOOST.
*/
#define AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH 0x000108A4
/* ID of the PBE module.
* This module supports the following parameter IDs:
* - #AUDPROC_PARAM_ID_PBE_ENABLE
* - #AUDPROC_PARAM_ID_PBE_PARAM_CONFIG
*/
#define AUDPROC_MODULE_ID_PBE 0x00010C2A
/* ID of the Bass Boost enable parameter used by
* AUDPROC_MODULE_ID_BASS_BOOST.
*/
#define AUDPROC_PARAM_ID_PBE_ENABLE 0x00010C2B
/* ID of the Bass Boost mode parameter used by
* AUDPROC_MODULE_ID_BASS_BOOST.
*/
#define AUDPROC_PARAM_ID_PBE_PARAM_CONFIG 0x00010C49
/* ID of the Virtualizer module. This module supports the
* following parameter IDs:
* - #AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE
* - #AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH
* - #AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE
* - #AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST
*/
#define AUDPROC_MODULE_ID_VIRTUALIZER 0x000108A5
/* ID of the Virtualizer enable parameter used by
* AUDPROC_MODULE_ID_VIRTUALIZER.
*/
#define AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE 0x000108A6
/* ID of the Virtualizer strength parameter used by
* AUDPROC_MODULE_ID_VIRTUALIZER.
*/
#define AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH 0x000108A7
/* ID of the Virtualizer out type parameter used by
* AUDPROC_MODULE_ID_VIRTUALIZER.
*/
#define AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE 0x000108A8
/* ID of the Virtualizer out type parameter used by
* AUDPROC_MODULE_ID_VIRTUALIZER.
*/
#define AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST 0x000108A9
/* ID of the Reverb module. This module supports the following
* parameter IDs:
* - #AUDPROC_PARAM_ID_REVERB_ENABLE
* - #AUDPROC_PARAM_ID_REVERB_MODE
* - #AUDPROC_PARAM_ID_REVERB_PRESET
* - #AUDPROC_PARAM_ID_REVERB_WET_MIX
* - #AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST
* - #AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL
* - #AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL
* - #AUDPROC_PARAM_ID_REVERB_DECAY_TIME
* - #AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO
* - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL
* - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY
* - #AUDPROC_PARAM_ID_REVERB_LEVEL
* - #AUDPROC_PARAM_ID_REVERB_DELAY
* - #AUDPROC_PARAM_ID_REVERB_DIFFUSION
* - #AUDPROC_PARAM_ID_REVERB_DENSITY
*/
#define AUDPROC_MODULE_ID_REVERB 0x000108AA
/* ID of the Reverb enable parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_ENABLE 0x000108AB
/* ID of the Reverb mode parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_MODE 0x000108AC
/* ID of the Reverb preset parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_PRESET 0x000108AD
/* ID of the Reverb wet mix parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_WET_MIX 0x000108AE
/* ID of the Reverb gain adjust parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST 0x000108AF
/* ID of the Reverb room level parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL 0x000108B0
/* ID of the Reverb room hf level parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL 0x000108B1
/* ID of the Reverb decay time parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_DECAY_TIME 0x000108B2
/* ID of the Reverb decay hf ratio parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO 0x000108B3
/* ID of the Reverb reflections level parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL 0x000108B4
/* ID of the Reverb reflections delay parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY 0x000108B5
/* ID of the Reverb level parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_LEVEL 0x000108B6
/* ID of the Reverb delay parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_DELAY 0x000108B7
/* ID of the Reverb diffusion parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_DIFFUSION 0x000108B8
/* ID of the Reverb density parameter used by
* AUDPROC_MODULE_ID_REVERB.
*/
#define AUDPROC_PARAM_ID_REVERB_DENSITY 0x000108B9
/* ID of the Popless Equalizer module. This module supports the
* following parameter IDs:
* - #AUDPROC_PARAM_ID_EQ_ENABLE
* - #AUDPROC_PARAM_ID_EQ_CONFIG
* - #AUDPROC_PARAM_ID_EQ_NUM_BANDS
* - #AUDPROC_PARAM_ID_EQ_BAND_LEVELS
* - #AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE
* - #AUDPROC_PARAM_ID_EQ_BAND_FREQS
* - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE
* - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ
* - #AUDPROC_PARAM_ID_EQ_BAND_INDEX
* - #AUDPROC_PARAM_ID_EQ_PRESET_ID
* - #AUDPROC_PARAM_ID_EQ_NUM_PRESETS
* - #AUDPROC_PARAM_ID_EQ_GET_PRESET_NAME
*/
#define AUDPROC_MODULE_ID_POPLESS_EQUALIZER 0x000108BA
/* ID of the Popless Equalizer enable parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
*/
#define AUDPROC_PARAM_ID_EQ_ENABLE 0x000108BB
/* ID of the Popless Equalizer config parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
*/
#define AUDPROC_PARAM_ID_EQ_CONFIG 0x000108BC
/* ID of the Popless Equalizer number of bands parameter used
* by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
* used for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_NUM_BANDS 0x000108BD
/* ID of the Popless Equalizer band levels parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
* used for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_BAND_LEVELS 0x000108BE
/* ID of the Popless Equalizer band level range parameter used
* by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
* used for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE 0x000108BF
/* ID of the Popless Equalizer band frequencies parameter used
* by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
* used for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_BAND_FREQS 0x000108C0
/* ID of the Popless Equalizer single band frequency range
* parameter used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
* This param ID is used for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE 0x000108C1
/* ID of the Popless Equalizer single band frequency parameter
* used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID
* is used for set param only.
*/
#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ 0x000108C2
/* ID of the Popless Equalizer band index parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
*/
#define AUDPROC_PARAM_ID_EQ_BAND_INDEX 0x000108C3
/* ID of the Popless Equalizer preset id parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
* for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_PRESET_ID 0x000108C4
/* ID of the Popless Equalizer number of presets parameter used
* by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
* for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_NUM_PRESETS 0x000108C5
/* ID of the Popless Equalizer preset name parameter used by
* AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
* for get param only.
*/
#define AUDPROC_PARAM_ID_EQ_PRESET_NAME 0x000108C6
/* Set Q6 topologies */
#define ASM_CMD_ADD_TOPOLOGIES 0x00010DBE
#define ADM_CMD_ADD_TOPOLOGIES 0x00010335
#define AFE_CMD_ADD_TOPOLOGIES 0x000100f8
/* structure used for both ioctls */
struct cmd_set_topologies {
struct apr_hdr hdr;
u32 payload_addr_lsw;
/* LSW of parameter data payload address.*/
u32 payload_addr_msw;
/* MSW of parameter data payload address.*/
u32 mem_map_handle;
/* Memory map handle returned by mem map command */
u32 payload_size;
/* Size in bytes of the variable payload in shared memory */
} __packed;
/* This module represents the Rx processing of Feedback speaker protection.
* It contains the excursion control, thermal protection,
* analog clip manager features in it.
* This module id will support following param ids.
* - AFE_PARAM_ID_FBSP_MODE_RX_CFG
*/
#define AFE_MODULE_FB_SPKR_PROT_RX 0x0001021C
#define AFE_MODULE_FB_SPKR_PROT_V2_RX 0x0001025F
#define AFE_PARAM_ID_FBSP_MODE_RX_CFG 0x0001021D
#define AFE_PARAM_ID_FBSP_PTONE_RAMP_CFG 0x00010260
struct asm_fbsp_mode_rx_cfg {
uint32_t minor_version;
uint32_t mode;
} __packed;
/* This module represents the VI processing of feedback speaker protection.
* It will receive Vsens and Isens from codec and generates necessary
* parameters needed by Rx processing.
* This module id will support following param ids.
* - AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG
* - AFE_PARAM_ID_CALIB_RES_CFG
* - AFE_PARAM_ID_FEEDBACK_PATH_CFG
*/
#define AFE_MODULE_FB_SPKR_PROT_VI_PROC 0x00010226
#define AFE_MODULE_FB_SPKR_PROT_VI_PROC_V2 0x0001026A
#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG 0x0001022A
#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG_V2 0x0001026B
struct asm_spkr_calib_vi_proc_cfg {
uint32_t minor_version;
uint32_t operation_mode;
uint32_t r0_t0_selection_flag[SP_V2_NUM_MAX_SPKR];
int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR];
uint32_t quick_calib_flag;
} __packed;
#define AFE_PARAM_ID_CALIB_RES_CFG 0x0001022B
#define AFE_PARAM_ID_CALIB_RES_CFG_V2 0x0001026E
struct asm_calib_res_cfg {
uint32_t minor_version;
int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
uint32_t th_vi_ca_state;
} __packed;
#define AFE_PARAM_ID_FEEDBACK_PATH_CFG 0x0001022C
#define AFE_MODULE_FEEDBACK 0x00010257
struct asm_feedback_path_cfg {
uint32_t minor_version;
int32_t dst_portid;
int32_t num_channels;
int32_t chan_info[4];
} __packed;
#define AFE_PARAM_ID_MODE_VI_PROC_CFG 0x00010227
struct asm_mode_vi_proc_cfg {
uint32_t minor_version;
uint32_t cal_mode;
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION_V2_TH_VI 0x0001026A
#define AFE_PARAM_ID_SP_V2_TH_VI_MODE_CFG 0x0001026B
#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_CFG 0x0001029F
#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_PARAMS 0x000102A0
struct afe_sp_th_vi_mode_cfg {
uint32_t minor_version;
uint32_t operation_mode;
/*
* Operation mode of thermal VI module.
* 0 -- Normal Running mode
* 1 -- Calibration mode
* 2 -- FTM mode
*/
uint32_t r0t0_selection_flag[SP_V2_NUM_MAX_SPKR];
/*
* Specifies which set of R0, T0 values the algorithm will use.
* This field is valid only in Normal mode (operation_mode = 0).
* 0 -- Use calibrated R0, T0 value
* 1 -- Use safe R0, T0 value
*/
int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
/*
* Calibration point resistance per device. This field is valid
* only in Normal mode (operation_mode = 0).
* values 33554432 to 1073741824 Ohms (in Q24 format)
*/
int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR];
/*
* Calibration point temperature per device. This field is valid
* in both Normal mode and Calibration mode.
* values -1920 to 5120 degrees C (in Q6 format)
*/
uint32_t quick_calib_flag;
/*
* Indicates whether calibration is to be done in quick mode or not.
* This field is valid only in Calibration mode (operation_mode = 1).
* 0 -- Disabled
* 1 -- Enabled
*/
} __packed;
struct afe_sp_th_vi_ftm_cfg {
uint32_t minor_version;
uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
struct afe_sp_th_vi_ftm_params {
uint32_t minor_version;
int32_t dc_res_q24[SP_V2_NUM_MAX_SPKR];
/*
* DC resistance value in q24 format
* values 0 to 2147483647 Ohms (in Q24 format)
*/
int32_t temp_q22[SP_V2_NUM_MAX_SPKR];
/*
* temperature value in q22 format
* values -125829120 to 2147483647 degC (in Q22 format)
*/
uint32_t status[SP_V2_NUM_MAX_SPKR];
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
struct afe_sp_th_vi_get_param {
struct apr_hdr hdr;
struct afe_port_cmd_get_param_v2 get_param;
struct afe_port_param_data_v2 pdata;
struct afe_sp_th_vi_ftm_params param;
} __packed;
struct afe_sp_th_vi_get_param_resp {
uint32_t status;
struct afe_port_param_data_v2 pdata;
struct afe_sp_th_vi_ftm_params param;
} __packed;
#define AFE_MODULE_SPEAKER_PROTECTION_V2_EX_VI 0x0001026F
#define AFE_PARAM_ID_SP_V2_EX_VI_MODE_CFG 0x000102A1
#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_CFG 0x000102A2
#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_PARAMS 0x000102A3
struct afe_sp_ex_vi_mode_cfg {
uint32_t minor_version;
uint32_t operation_mode;
/*
* Operation mode of Excursion VI module.
* 0 - Normal Running mode
* 2 - FTM mode
*/
} __packed;
struct afe_sp_ex_vi_ftm_cfg {
uint32_t minor_version;
uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* Wait time to heat up speaker before collecting statistics
* for ftm mode in ms.
* values 0 to 4294967295 ms
*/
uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
/*
* duration for which FTM statistics are collected in ms.
* values 0 to 2000 ms
*/
} __packed;
struct afe_sp_ex_vi_ftm_params {
uint32_t minor_version;
int32_t freq_q20[SP_V2_NUM_MAX_SPKR];
/*
* Resonance frequency in q20 format
* values 0 to 2147483647 Hz (in Q20 format)
*/
int32_t resis_q24[SP_V2_NUM_MAX_SPKR];
/*
* Mechanical resistance in q24 format
* values 0 to 2147483647 Ohms (in Q24 format)
*/
int32_t qmct_q24[SP_V2_NUM_MAX_SPKR];
/*
* Mechanical Qfactor in q24 format
* values 0 to 2147483647 (in Q24 format)
*/
uint32_t status[SP_V2_NUM_MAX_SPKR];
/*
* FTM packet status
* 0 - Incorrect operation mode.This status is returned
* when GET_PARAM is called in non FTM Mode.
* 1 - Inactive mode -- Port is not yet started.
* 2 - Wait state. wait_time_ms has not yet elapsed
* 3 - In progress state. ftm_time_ms has not yet elapsed.
* 4 - Success.
* 5 - Failed.
*/
} __packed;
struct afe_sp_ex_vi_get_param {
struct apr_hdr hdr;
struct afe_port_cmd_get_param_v2 get_param;
struct afe_port_param_data_v2 pdata;
struct afe_sp_ex_vi_ftm_params param;
} __packed;
struct afe_sp_ex_vi_get_param_resp {
uint32_t status;
struct afe_port_param_data_v2 pdata;
struct afe_sp_ex_vi_ftm_params param;
} __packed;
union afe_spkr_prot_config {
struct asm_fbsp_mode_rx_cfg mode_rx_cfg;
struct asm_spkr_calib_vi_proc_cfg vi_proc_cfg;
struct asm_feedback_path_cfg feedback_path_cfg;
struct asm_mode_vi_proc_cfg mode_vi_proc_cfg;
struct afe_sp_th_vi_mode_cfg th_vi_mode_cfg;
struct afe_sp_th_vi_ftm_cfg th_vi_ftm_cfg;
struct afe_sp_ex_vi_mode_cfg ex_vi_mode_cfg;
struct afe_sp_ex_vi_ftm_cfg ex_vi_ftm_cfg;
} __packed;
struct afe_spkr_prot_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
union afe_spkr_prot_config prot_config;
} __packed;
struct afe_spkr_prot_get_vi_calib {
struct apr_hdr hdr;
struct afe_port_cmd_get_param_v2 get_param;
struct afe_port_param_data_v2 pdata;
struct asm_calib_res_cfg res_cfg;
} __packed;
struct afe_spkr_prot_calib_get_resp {
uint32_t status;
struct afe_port_param_data_v2 pdata;
struct asm_calib_res_cfg res_cfg;
} __packed;
/* SRS TRUMEDIA start */
/* topology */
#define SRS_TRUMEDIA_TOPOLOGY_ID 0x00010D90
/* module */
#define SRS_TRUMEDIA_MODULE_ID 0x10005010
/* parameters */
#define SRS_TRUMEDIA_PARAMS 0x10005011
#define SRS_TRUMEDIA_PARAMS_WOWHD 0x10005012
#define SRS_TRUMEDIA_PARAMS_CSHP 0x10005013
#define SRS_TRUMEDIA_PARAMS_HPF 0x10005014
#define SRS_TRUMEDIA_PARAMS_AEQ 0x10005015
#define SRS_TRUMEDIA_PARAMS_HL 0x10005016
#define SRS_TRUMEDIA_PARAMS_GEQ 0x10005017
#define SRS_ID_GLOBAL 0x00000001
#define SRS_ID_WOWHD 0x00000002
#define SRS_ID_CSHP 0x00000003
#define SRS_ID_HPF 0x00000004
#define SRS_ID_AEQ 0x00000005
#define SRS_ID_HL 0x00000006
#define SRS_ID_GEQ 0x00000007
#define SRS_CMD_UPLOAD 0x7FFF0000
#define SRS_PARAM_OFFSET_MASK 0x3FFF0000
#define SRS_PARAM_VALUE_MASK 0x0000FFFF
struct srs_trumedia_params_GLOBAL {
uint8_t v1;
uint8_t v2;
uint8_t v3;
uint8_t v4;
uint8_t v5;
uint8_t v6;
uint8_t v7;
uint8_t v8;
uint16_t v9;
} __packed;
struct srs_trumedia_params_WOWHD {
uint32_t v1;
uint16_t v2;
uint16_t v3;
uint16_t v4;
uint16_t v5;
uint16_t v6;
uint16_t v7;
uint16_t v8;
uint16_t v____A1;
uint32_t v9;
uint16_t v10;
uint16_t v11;
uint32_t v12[16];
uint32_t v13[16];
uint32_t v14[16];
uint32_t v15[16];
uint32_t v16;
uint16_t v17;
uint16_t v18;
} __packed;
struct srs_trumedia_params_CSHP {
uint32_t v1;
uint16_t v2;
uint16_t v3;
uint16_t v4;
uint16_t v5;
uint16_t v6;
uint16_t v____A1;
uint32_t v7;
uint16_t v8;
uint16_t v9;
uint32_t v10[16];
} __packed;
struct srs_trumedia_params_HPF {
uint32_t v1;
uint32_t v2[26];
} __packed;
struct srs_trumedia_params_AEQ {
uint32_t v1;
uint16_t v2;
uint16_t v3;
uint16_t v4;
uint16_t v____A1;
uint32_t v5[74];
uint32_t v6[74];
uint16_t v7[2048];
} __packed;
struct srs_trumedia_params_HL {
uint16_t v1;
uint16_t v2;
uint16_t v3;
uint16_t v____A1;
int32_t v4;
uint32_t v5;
uint16_t v6;
uint16_t v____A2;
uint32_t v7;
} __packed;
struct srs_trumedia_params_GEQ {
int16_t v1[10];
} __packed;
struct srs_trumedia_params {
struct srs_trumedia_params_GLOBAL global;
struct srs_trumedia_params_WOWHD wowhd;
struct srs_trumedia_params_CSHP cshp;
struct srs_trumedia_params_HPF hpf;
struct srs_trumedia_params_AEQ aeq;
struct srs_trumedia_params_HL hl;
struct srs_trumedia_params_GEQ geq;
} __packed;
/* SRS TruMedia end */
#define AUDPROC_PARAM_ID_ENABLE 0x00010904
#define ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF
/* DTS Eagle */
#define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C
#define AUDPROC_MODULE_ID_DTS_HPX_POSTMIX 0x0001077B
#define ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX 0x00010DED
#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS 0x10015000
#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER 0x10015001
struct asm_dts_eagle_param {
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 param;
struct asm_stream_param_data_v2 data;
} __packed;
struct asm_dts_eagle_param_get {
struct apr_hdr hdr;
struct asm_stream_cmd_get_pp_params_v2 param;
} __packed;
/* Opcode to set BT address and license for aptx decoder */
#define APTX_DECODER_BT_ADDRESS 0x00013201
#define APTX_CLASSIC_DEC_LICENSE_ID 0x00013202
struct aptx_dec_bt_addr_cfg {
uint32_t lap;
uint32_t uap;
uint32_t nap;
} __packed;
struct aptx_dec_bt_dev_addr {
struct apr_hdr hdr;
struct asm_stream_cmd_set_encdec_param encdec;
struct aptx_dec_bt_addr_cfg bt_addr_cfg;
} __packed;
struct asm_aptx_dec_fmt_blk_v2 {
struct apr_hdr hdr;
struct asm_data_cmd_media_fmt_update_v2 fmtblk;
u32 sample_rate;
/* Number of samples per second.
* Supported values: 44100 and 48000 Hz
*/
} __packed;
/* LSM Specific */
#define VW_FEAT_DIM (39)
#define APRV2_IDS_SERVICE_ID_ADSP_LSM_V (0xD)
#define APRV2_IDS_DOMAIN_ID_ADSP_V (0x4)
#define APRV2_IDS_DOMAIN_ID_APPS_V (0x5)
#define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS (0x00012A7F)
#define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS (0x00012A80)
#define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS (0x00012A81)
#define LSM_SESSION_CMD_OPEN_TX (0x00012A82)
#define LSM_SESSION_CMD_CLOSE_TX (0x00012A88)
#define LSM_SESSION_CMD_SET_PARAMS (0x00012A83)
#define LSM_SESSION_CMD_SET_PARAMS_V2 (0x00012A8F)
#define LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x00012A84)
#define LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x00012A85)
#define LSM_SESSION_CMD_START (0x00012A86)
#define LSM_SESSION_CMD_STOP (0x00012A87)
#define LSM_SESSION_CMD_EOB (0x00012A89)
#define LSM_SESSION_CMD_READ (0x00012A8A)
#define LSM_SESSION_CMD_OPEN_TX_V2 (0x00012A8B)
#define LSM_CMD_ADD_TOPOLOGIES (0x00012A8C)
#define LSM_SESSION_EVENT_DETECTION_STATUS (0x00012B00)
#define LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x00012B01)
#define LSM_DATA_EVENT_READ_DONE (0x00012B02)
#define LSM_DATA_EVENT_STATUS (0x00012B03)
#define LSM_SESSION_EVENT_DETECTION_STATUS_V3 (0x00012B04)
#define LSM_MODULE_ID_VOICE_WAKEUP (0x00012C00)
#define LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01)
#define LSM_PARAM_ID_OPERATION_MODE (0x00012C02)
#define LSM_PARAM_ID_GAIN (0x00012C03)
#define LSM_PARAM_ID_CONNECT_TO_PORT (0x00012C04)
#define LSM_PARAM_ID_FEATURE_COMPENSATION_DATA (0x00012C07)
#define LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS (0x00012C07)
#define LSM_MODULE_ID_LAB (0x00012C08)
#define LSM_PARAM_ID_LAB_ENABLE (0x00012C09)
#define LSM_PARAM_ID_LAB_CONFIG (0x00012C0A)
#define LSM_MODULE_ID_FRAMEWORK (0x00012C0E)
#define LSM_PARAM_ID_SWMAD_CFG (0x00012C18)
#define LSM_PARAM_ID_SWMAD_MODEL (0x00012C19)
#define LSM_PARAM_ID_SWMAD_ENABLE (0x00012C1A)
#define LSM_PARAM_ID_POLLING_ENABLE (0x00012C1B)
#define LSM_PARAM_ID_MEDIA_FMT (0x00012C1E)
#define LSM_PARAM_ID_FWK_MODE_CONFIG (0x00012C27)
/* HW MAD specific */
#define AFE_MODULE_HW_MAD (0x00010230)
#define AFE_PARAM_ID_HW_MAD_CFG (0x00010231)
#define AFE_PARAM_ID_HW_MAD_CTRL (0x00010232)
#define AFE_PARAM_ID_SLIMBUS_SLAVE_PORT_CFG (0x00010233)
/* SW MAD specific */
#define AFE_MODULE_SW_MAD (0x0001022D)
#define AFE_PARAM_ID_SW_MAD_CFG (0x0001022E)
#define AFE_PARAM_ID_SVM_MODEL (0x0001022F)
/* Commands/Params to pass the codec/slimbus data to DSP */
#define AFE_SVC_CMD_SET_PARAM (0x000100f3)
#define AFE_MODULE_CDC_DEV_CFG (0x00010234)
#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG (0x00010235)
#define AFE_PARAM_ID_CDC_REG_CFG (0x00010236)
#define AFE_PARAM_ID_CDC_REG_CFG_INIT (0x00010237)
#define AFE_PARAM_ID_CDC_REG_PAGE_CFG (0x00010296)
#define AFE_MAX_CDC_REGISTERS_TO_CONFIG (20)
/* AANC Port Config Specific */
#define AFE_PARAM_ID_AANC_PORT_CONFIG (0x00010215)
#define AFE_API_VERSION_AANC_PORT_CONFIG (0x1)
#define AANC_TX_MIC_UNUSED (0)
#define AANC_TX_VOICE_MIC (1)
#define AANC_TX_ERROR_MIC (2)
#define AANC_TX_NOISE_MIC (3)
#define AFE_PORT_MAX_CHANNEL_CNT (8)
#define AFE_MODULE_AANC (0x00010214)
#define AFE_PARAM_ID_CDC_AANC_VERSION (0x0001023A)
#define AFE_API_VERSION_CDC_AANC_VERSION (0x1)
#define AANC_HW_BLOCK_VERSION_1 (1)
#define AANC_HW_BLOCK_VERSION_2 (2)
/*Clip bank selection*/
#define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1
#define AFE_CLIP_MAX_BANKS 4
#define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242
struct afe_param_aanc_port_cfg {
/* Minor version used for tracking the version of the module's
* source port configuration.
*/
uint32_t aanc_port_cfg_minor_version;
/* Sampling rate of the source Tx port. 8k - 192k*/
uint32_t tx_port_sample_rate;
/* Channel mapping for the Tx port signal carrying Noise (X),
* Error (E), and Voice (V) signals.
*/
uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT];
/* Number of channels on the source Tx port. */
uint16_t tx_port_num_channels;
/* Port ID of the Rx path reference signal. */
uint16_t rx_path_ref_port_id;
/* Sampling rate of the reference port. 8k - 192k*/
uint32_t ref_port_sample_rate;
} __packed;
struct afe_param_id_cdc_aanc_version {
/* Minor version used for tracking the version of the module's
* hw version
*/
uint32_t cdc_aanc_minor_version;
/* HW version. */
uint32_t aanc_hw_version;
} __packed;
struct afe_param_id_clip_bank_sel {
/* Minor version used for tracking the version of the module's
* hw version
*/
uint32_t minor_version;
/* Number of banks to be read */
uint32_t num_banks;
uint32_t bank_map[AFE_CLIP_MAX_BANKS];
} __packed;
/* ERROR CODES */
/* Success. The operation completed with no errors. */
#define ADSP_EOK 0x00000000
/* General failure. */
#define ADSP_EFAILED 0x00000001
/* Bad operation parameter. */
#define ADSP_EBADPARAM 0x00000002
/* Unsupported routine or operation. */
#define ADSP_EUNSUPPORTED 0x00000003
/* Unsupported version. */
#define ADSP_EVERSION 0x00000004
/* Unexpected problem encountered. */
#define ADSP_EUNEXPECTED 0x00000005
/* Unhandled problem occurred. */
#define ADSP_EPANIC 0x00000006
/* Unable to allocate resource. */
#define ADSP_ENORESOURCE 0x00000007
/* Invalid handle. */
#define ADSP_EHANDLE 0x00000008
/* Operation is already processed. */
#define ADSP_EALREADY 0x00000009
/* Operation is not ready to be processed. */
#define ADSP_ENOTREADY 0x0000000A
/* Operation is pending completion. */
#define ADSP_EPENDING 0x0000000B
/* Operation could not be accepted or processed. */
#define ADSP_EBUSY 0x0000000C
/* Operation aborted due to an error. */
#define ADSP_EABORTED 0x0000000D
/* Operation preempted by a higher priority. */
#define ADSP_EPREEMPTED 0x0000000E
/* Operation requests intervention to complete. */
#define ADSP_ECONTINUE 0x0000000F
/* Operation requests immediate intervention to complete. */
#define ADSP_EIMMEDIATE 0x00000010
/* Operation is not implemented. */
#define ADSP_ENOTIMPL 0x00000011
/* Operation needs more data or resources. */
#define ADSP_ENEEDMORE 0x00000012
/* Operation does not have memory. */
#define ADSP_ENOMEMORY 0x00000014
/* Item does not exist. */
#define ADSP_ENOTEXIST 0x00000015
/* Max count for adsp error code sent to HLOS*/
#define ADSP_ERR_MAX (ADSP_ENOTEXIST + 1)
/* Operation is finished. */
#define ADSP_ETERMINATED 0x00011174
/*bharath, adsp_error_codes.h */
/* LPASS clock for I2S Interface */
/* Supported OSR clock values */
#define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ 0xBB8000
#define Q6AFE_LPASS_OSR_CLK_11_P2896_MHZ 0xAC4400
#define Q6AFE_LPASS_OSR_CLK_9_P600_MHZ 0x927C00
#define Q6AFE_LPASS_OSR_CLK_8_P192_MHZ 0x7D0000
#define Q6AFE_LPASS_OSR_CLK_6_P144_MHZ 0x5DC000
#define Q6AFE_LPASS_OSR_CLK_4_P096_MHZ 0x3E8000
#define Q6AFE_LPASS_OSR_CLK_3_P072_MHZ 0x2EE000
#define Q6AFE_LPASS_OSR_CLK_2_P048_MHZ 0x1F4000
#define Q6AFE_LPASS_OSR_CLK_1_P536_MHZ 0x177000
#define Q6AFE_LPASS_OSR_CLK_1_P024_MHZ 0xFA000
#define Q6AFE_LPASS_OSR_CLK_768_kHZ 0xBB800
#define Q6AFE_LPASS_OSR_CLK_512_kHZ 0x7D000
#define Q6AFE_LPASS_OSR_CLK_DISABLE 0x0
/* Supported Bit clock values */
#define Q6AFE_LPASS_IBIT_CLK_12_P288_MHZ 0xBB8000
#define Q6AFE_LPASS_IBIT_CLK_11_P2896_MHZ 0xAC4400
#define Q6AFE_LPASS_IBIT_CLK_8_P192_MHZ 0x7D0000
#define Q6AFE_LPASS_IBIT_CLK_6_P144_MHZ 0x5DC000
#define Q6AFE_LPASS_IBIT_CLK_4_P096_MHZ 0x3E8000
#define Q6AFE_LPASS_IBIT_CLK_3_P072_MHZ 0x2EE000
#define Q6AFE_LPASS_IBIT_CLK_2_P8224_MHZ 0x2b1100
#define Q6AFE_LPASS_IBIT_CLK_2_P048_MHZ 0x1F4000
#define Q6AFE_LPASS_IBIT_CLK_1_P536_MHZ 0x177000
#define Q6AFE_LPASS_IBIT_CLK_1_P4112_MHZ 0x158880
#define Q6AFE_LPASS_IBIT_CLK_1_P024_MHZ 0xFA000
#define Q6AFE_LPASS_IBIT_CLK_768_KHZ 0xBB800
#define Q6AFE_LPASS_IBIT_CLK_512_KHZ 0x7D000
#define Q6AFE_LPASS_IBIT_CLK_256_KHZ 0x3E800
#define Q6AFE_LPASS_IBIT_CLK_DISABLE 0x0
/* Supported LPASS CLK sources */
#define Q6AFE_LPASS_CLK_SRC_EXTERNAL 0
#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
/* Supported LPASS CLK root*/
#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
enum afe_lpass_clk_mode {
Q6AFE_LPASS_MODE_BOTH_INVALID,
Q6AFE_LPASS_MODE_CLK1_VALID,
Q6AFE_LPASS_MODE_CLK2_VALID,
Q6AFE_LPASS_MODE_BOTH_VALID,
} __packed;
/* Clock ID Enumeration Define. */
/* Clock ID for Primary I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100
/* Clock ID for Primary I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101
/* Clock ID for Secondary I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102
/* Clock ID for Secondary I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103
/* Clock ID for Tertiary I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104
/* Clock ID for Tertiary I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105
/* Clock ID for Quartnery I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106
/* Clock ID for Quartnery I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107
/* Clock ID for Speaker I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108
/* Clock ID for Speaker I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109
/* Clock ID for Speaker I2S OSR */
#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A
/* Clock ID for QUINARY I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B
/* Clock ID for QUINARY I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C
/* Clock ID for SENARY I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D
/* Clock ID for SENARY I2S EBIT */
#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E
/* Clock ID for INT0 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
/* Clock ID for INT1 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
/* Clock ID for INT2 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
/* Clock ID for INT3 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
/* Clock ID for INT4 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
/* Clock ID for INT5 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
/* Clock ID for INT6 I2S IBIT */
#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
/* Clock ID for Primary PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200
/* Clock ID for Primary PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201
/* Clock ID for Secondary PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202
/* Clock ID for Secondary PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203
/* Clock ID for Tertiary PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204
/* Clock ID for Tertiary PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205
/* Clock ID for Quartery PCM IBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206
/* Clock ID for Quartery PCM EBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207
/** Clock ID for Primary TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200
/** Clock ID for Primary TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201
/** Clock ID for Secondary TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202
/** Clock ID for Secondary TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203
/** Clock ID for Tertiary TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204
/** Clock ID for Tertiary TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205
/** Clock ID for Quartery TDM IBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206
/** Clock ID for Quartery TDM EBIT */
#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207
/* Clock ID for MCLK1 */
#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300
/* Clock ID for MCLK2 */
#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301
/* Clock ID for MCLK3 */
#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302
/* Clock ID for MCLK4 */
#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
/* Clock ID for Internal Digital Codec Core */
#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
/* Clock ID for INT MCLK0 */
#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
/* Clock ID for INT MCLK1 */
#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
/*
* Clock ID for soundwire NPL.
* This is the clock to be used to enable NPL clock for internal Soundwire.
*/
#define AFE_CLOCK_SET_CLOCK_ID_SWR_NPL_CLK 0x307
/* Clock ID for AHB HDMI input */
#define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT 0x400
/* Clock ID for SPDIF core */
#define Q6AFE_LPASS_CLK_ID_SPDIF_CORE 0x500
/* Clock attribute for invalid use (reserved for internal usage) */
#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0
/* Clock attribute for no couple case */
#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1
/* Clock attribute for dividend couple case */
#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2
/* Clock attribute for divisor couple case */
#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3
/* Clock attribute for invert and no couple case */
#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4
/* Clock set API version */
#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1
struct afe_clk_set {
/*
* Minor version used for tracking clock set.
* @values #AFE_API_VERSION_CLOCK_SET
*/
uint32_t clk_set_minor_version;
/*
* Clock ID
* @values
* - 0x100 to 0x10A - MSM8996
* - 0x200 to 0x207 - MSM8996
* - 0x300 to 0x302 - MSM8996 @tablebulletend
*/
uint32_t clk_id;
/*
* Clock frequency (in Hertz) to be set.
* @values
* - >= 0 for clock frequency to set @tablebulletend
*/
uint32_t clk_freq_in_hz;
/* Use to specific divider for two clocks if needed.
* Set to Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO for no divider
* relation clocks
* @values
* - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO
* - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND
* - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR @tablebulletend
*/
uint16_t clk_attri;
/*
* Specifies the root clock source.
* Currently, only Q6AFE_LPASS_CLK_ROOT_DEFAULT is valid
* @values
* - 0 @tablebulletend
*/
uint16_t clk_root;
/*
* for enable and disable clock.
* "clk_freq_in_hz", "clk_attri", and "clk_root"
* are ignored in disable clock case.
* @values 
* - 0 -- Disabled
* - 1 -- Enabled @tablebulletend
*/
uint32_t enable;
};
struct afe_clk_cfg {
/* Minor version used for tracking the version of the I2S
* configuration interface.
* Supported values: #AFE_API_VERSION_I2S_CONFIG
*/
u32 i2s_cfg_minor_version;
/* clk value 1 in MHz. */
u32 clk_val1;
/* clk value 2 in MHz. */
u32 clk_val2;
/* clk_src
* #Q6AFE_LPASS_CLK_SRC_EXTERNAL
* #Q6AFE_LPASS_CLK_SRC_INTERNAL
*/
u16 clk_src;
/* clk_root -0 for default */
u16 clk_root;
/* clk_set_mode
* #Q6AFE_LPASS_MODE_BOTH_INVALID
* #Q6AFE_LPASS_MODE_CLK1_VALID
* #Q6AFE_LPASS_MODE_CLK2_VALID
* #Q6AFE_LPASS_MODE_BOTH_VALID
*/
u16 clk_set_mode;
/* This param id is used to configure I2S clk */
u16 reserved;
} __packed;
/* This param id is used to configure I2S clk */
#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
#define AFE_MODULE_CLOCK_SET 0x0001028F
#define AFE_PARAM_ID_CLOCK_SET 0x00010290
struct afe_lpass_clk_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_clk_cfg clk_cfg;
} __packed;
enum afe_lpass_digital_clk_src {
Q6AFE_LPASS_DIGITAL_ROOT_INVALID,
Q6AFE_LPASS_DIGITAL_ROOT_PRI_MI2S_OSR,
Q6AFE_LPASS_DIGITAL_ROOT_SEC_MI2S_OSR,
Q6AFE_LPASS_DIGITAL_ROOT_TER_MI2S_OSR,
Q6AFE_LPASS_DIGITAL_ROOT_QUAD_MI2S_OSR,
Q6AFE_LPASS_DIGITAL_ROOT_CDC_ROOT_CLK,
} __packed;
/* This param id is used to configure internal clk */
#define AFE_PARAM_ID_INTERNAL_DIGIATL_CDC_CLK_CONFIG 0x00010239
struct afe_digital_clk_cfg {
/* Minor version used for tracking the version of the I2S
* configuration interface.
* Supported values: #AFE_API_VERSION_I2S_CONFIG
*/
u32 i2s_cfg_minor_version;
/* clk value in MHz. */
u32 clk_val;
/* INVALID
* PRI_MI2S_OSR
* SEC_MI2S_OSR
* TER_MI2S_OSR
* QUAD_MI2S_OSR
* DIGT_CDC_ROOT
*/
u16 clk_root;
/* This field must be set to zero. */
u16 reserved;
} __packed;
struct afe_lpass_digital_clk_config_command {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_digital_clk_cfg clk_cfg;
} __packed;
/*
* Opcode for AFE to start DTMF.
*/
#define AFE_PORTS_CMD_DTMF_CTL 0x00010102
/** DTMF payload.*/
struct afe_dtmf_generation_command {
struct apr_hdr hdr;
/*
* Duration of the DTMF tone in ms.
* -1 -> continuous,
* 0 -> disable
*/
int64_t duration_in_ms;
/*
* The DTMF high tone frequency.
*/
uint16_t high_freq;
/*
* The DTMF low tone frequency.
*/
uint16_t low_freq;
/*
* The DTMF volume setting
*/
uint16_t gain;
/*
* The number of ports to enable/disable on.
*/
uint16_t num_ports;
/*
* The Destination ports - array .
* For DTMF on multiple ports, portIds needs to
* be populated numPorts times.
*/
uint16_t port_ids;
/*
* variable for 32 bit alignment of APR packet.
*/
uint16_t reserved;
} __packed;
enum afe_config_type {
AFE_SLIMBUS_SLAVE_PORT_CONFIG,
AFE_SLIMBUS_SLAVE_CONFIG,
AFE_CDC_REGISTERS_CONFIG,
AFE_AANC_VERSION,
AFE_CDC_CLIP_REGISTERS_CONFIG,
AFE_CLIP_BANK_SEL,
AFE_CDC_REGISTER_PAGE_CONFIG,
AFE_MAX_CONFIG_TYPES,
};
struct afe_param_slimbus_slave_port_cfg {
uint32_t minor_version;
uint16_t slimbus_dev_id;
uint16_t slave_dev_pgd_la;
uint16_t slave_dev_intfdev_la;
uint16_t bit_width;
uint16_t data_format;
uint16_t num_channels;
uint16_t slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
} __packed;
struct afe_param_cdc_slimbus_slave_cfg {
uint32_t minor_version;
uint32_t device_enum_addr_lsw;
uint32_t device_enum_addr_msw;
uint16_t tx_slave_port_offset;
uint16_t rx_slave_port_offset;
} __packed;
struct afe_param_cdc_reg_cfg {
uint32_t minor_version;
uint32_t reg_logical_addr;
uint32_t reg_field_type;
uint32_t reg_field_bit_mask;
uint16_t reg_bit_width;
uint16_t reg_offset_scale;
} __packed;
#define AFE_API_VERSION_CDC_REG_PAGE_CFG 1
enum {
AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_0 = 0,
AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_1,
AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_2,
AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_3,
};
struct afe_param_cdc_reg_page_cfg {
uint32_t minor_version;
uint32_t enable;
uint32_t proc_id;
} __packed;
struct afe_param_cdc_reg_cfg_data {
uint32_t num_registers;
struct afe_param_cdc_reg_cfg *reg_data;
} __packed;
struct afe_svc_cmd_set_param {
uint32_t payload_size;
uint32_t payload_address_lsw;
uint32_t payload_address_msw;
uint32_t mem_map_handle;
} __packed;
struct afe_svc_param_data {
uint32_t module_id;
uint32_t param_id;
uint16_t param_size;
uint16_t reserved;
} __packed;
struct afe_param_hw_mad_ctrl {
uint32_t minor_version;
uint16_t mad_type;
uint16_t mad_enable;
} __packed;
struct afe_cmd_hw_mad_ctrl {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_hw_mad_ctrl payload;
} __packed;
struct afe_cmd_hw_mad_slimbus_slave_port_cfg {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
struct afe_param_slimbus_slave_port_cfg sb_port_cfg;
} __packed;
struct afe_cmd_sw_mad_enable {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
} __packed;
struct afe_param_cdc_reg_cfg_payload {
struct afe_svc_param_data common;
struct afe_param_cdc_reg_cfg reg_cfg;
} __packed;
struct afe_lpass_clk_config_command_v2 {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_svc_param_data pdata;
struct afe_clk_set clk_cfg;
} __packed;
/*
* reg_data's size can be up to AFE_MAX_CDC_REGISTERS_TO_CONFIG
*/
struct afe_svc_cmd_cdc_reg_cfg {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_param_cdc_reg_cfg_payload reg_data[0];
} __packed;
struct afe_svc_cmd_init_cdc_reg_cfg {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 init;
} __packed;
struct afe_svc_cmd_sb_slave_cfg {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 pdata;
struct afe_param_cdc_slimbus_slave_cfg sb_slave_cfg;
} __packed;
struct afe_svc_cmd_cdc_reg_page_cfg {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 pdata;
struct afe_param_cdc_reg_page_cfg cdc_reg_page_cfg;
} __packed;
struct afe_svc_cmd_cdc_aanc_version {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_cdc_aanc_version version;
} __packed;
struct afe_port_cmd_set_aanc_param {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
struct afe_port_param_data_v2 pdata;
union {
struct afe_param_aanc_port_cfg aanc_port_cfg;
struct afe_mod_enable_param mod_enable;
} __packed data;
} __packed;
struct afe_port_cmd_set_aanc_acdb_table {
struct apr_hdr hdr;
struct afe_port_cmd_set_param_v2 param;
} __packed;
/* Dolby DAP topology */
#define DOLBY_ADM_COPP_TOPOLOGY_ID 0x0001033B
#define DS2_ADM_COPP_TOPOLOGY_ID 0x1301033B
/* RMS value from DSP */
#define RMS_MODULEID_APPI_PASSTHRU 0x10009011
#define RMS_PARAM_FIRST_SAMPLE 0x10009012
#define RMS_PAYLOAD_LEN 4
/* Customized mixing in matix mixer */
#define MTMX_MODULE_ID_DEFAULT_CHMIXER 0x00010341
#define DEFAULT_CHMIXER_PARAM_ID_COEFF 0x00010342
#define CUSTOM_STEREO_PAYLOAD_SIZE 9
#define CUSTOM_STEREO_CMD_PARAM_SIZE 24
#define CUSTOM_STEREO_NUM_OUT_CH 0x0002
#define CUSTOM_STEREO_NUM_IN_CH 0x0002
#define CUSTOM_STEREO_INDEX_PARAM 0x0002
#define Q14_GAIN_ZERO_POINT_FIVE 0x2000
#define Q14_GAIN_UNITY 0x4000
struct afe_svc_cmd_set_clip_bank_selection {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_clip_bank_sel bank_sel;
} __packed;
/* Ultrasound supported formats */
#define US_POINT_EPOS_FORMAT_V2 0x0001272D
#define US_RAW_FORMAT_V2 0x0001272C
#define US_PROX_FORMAT_V4 0x0001273B
#define US_RAW_SYNC_FORMAT 0x0001272F
#define US_GES_SYNC_FORMAT 0x00012730
#define AFE_MODULE_GROUP_DEVICE 0x00010254
#define AFE_PARAM_ID_GROUP_DEVICE_CFG 0x00010255
#define AFE_PARAM_ID_GROUP_DEVICE_ENABLE 0x00010256
#define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX 0x1102
/* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_CFG
* parameter, which configures max of 8 AFE ports
* into a group.
* The fixed size of this structure is sixteen bytes.
*/
struct afe_group_device_group_cfg {
u32 minor_version;
u16 group_id;
u16 num_channels;
u16 port_id[8];
} __packed;
#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX \
(AFE_PORT_ID_PRIMARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX \
(AFE_PORT_ID_PRIMARY_TDM_TX + 0x100)
#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX \
(AFE_PORT_ID_SECONDARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX \
(AFE_PORT_ID_SECONDARY_TDM_TX + 0x100)
#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX \
(AFE_PORT_ID_TERTIARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX \
(AFE_PORT_ID_TERTIARY_TDM_TX + 0x100)
#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX \
(AFE_PORT_ID_QUATERNARY_TDM_RX + 0x100)
#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX \
(AFE_PORT_ID_QUATERNARY_TDM_TX + 0x100)
/* ID of the parameter used by #AFE_MODULE_GROUP_DEVICE to configure the
* group device. #AFE_SVC_CMD_SET_PARAM can use this parameter ID.
*
* Requirements:
* - Configure the group before the member ports in the group are
* configured and started.
* - Enable the group only after it is configured.
* - Stop all member ports in the group before disabling the group.
*/
#define AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG 0x0001029E
/* Version information used to handle future additions to
* AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG processing (for backward compatibility).
*/
#define AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG 0x1
/* Number of AFE ports in group device */
#define AFE_GROUP_DEVICE_NUM_PORTS 8
/* Payload of the AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG parameter ID
* used by AFE_MODULE_GROUP_DEVICE.
*/
struct afe_param_id_group_device_tdm_cfg {
u32 group_device_cfg_minor_version;
/* Minor version used to track group device configuration.
* @values #AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG
*/
u16 group_id;
/* ID for the group device.
* @values
* - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX
* - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX
* - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX
* - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX
* - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX
* - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX
* - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX
* - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX
*/
u16 reserved;
/* 0 */
u16 port_id[AFE_GROUP_DEVICE_NUM_PORTS];
/* Array of member port IDs of this group.
* @values
* - #AFE_PORT_ID_PRIMARY_TDM_RX
* - #AFE_PORT_ID_PRIMARY_TDM_RX_1
* - #AFE_PORT_ID_PRIMARY_TDM_RX_2
* - #AFE_PORT_ID_PRIMARY_TDM_RX_3
* - #AFE_PORT_ID_PRIMARY_TDM_RX_4
* - #AFE_PORT_ID_PRIMARY_TDM_RX_5
* - #AFE_PORT_ID_PRIMARY_TDM_RX_6
* - #AFE_PORT_ID_PRIMARY_TDM_RX_7
* - #AFE_PORT_ID_PRIMARY_TDM_TX
* - #AFE_PORT_ID_PRIMARY_TDM_TX_1
* - #AFE_PORT_ID_PRIMARY_TDM_TX_2
* - #AFE_PORT_ID_PRIMARY_TDM_TX_3
* - #AFE_PORT_ID_PRIMARY_TDM_TX_4
* - #AFE_PORT_ID_PRIMARY_TDM_TX_5
* - #AFE_PORT_ID_PRIMARY_TDM_TX_6
* - #AFE_PORT_ID_PRIMARY_TDM_TX_7
* - #AFE_PORT_ID_SECONDARY_TDM_RX
* - #AFE_PORT_ID_SECONDARY_TDM_RX_1
* - #AFE_PORT_ID_SECONDARY_TDM_RX_2
* - #AFE_PORT_ID_SECONDARY_TDM_RX_3
* - #AFE_PORT_ID_SECONDARY_TDM_RX_4
* - #AFE_PORT_ID_SECONDARY_TDM_RX_5
* - #AFE_PORT_ID_SECONDARY_TDM_RX_6
* - #AFE_PORT_ID_SECONDARY_TDM_RX_7
* - #AFE_PORT_ID_SECONDARY_TDM_TX
* - #AFE_PORT_ID_SECONDARY_TDM_TX_1
* - #AFE_PORT_ID_SECONDARY_TDM_TX_2
* - #AFE_PORT_ID_SECONDARY_TDM_TX_3
* - #AFE_PORT_ID_SECONDARY_TDM_TX_4
* - #AFE_PORT_ID_SECONDARY_TDM_TX_5
* - #AFE_PORT_ID_SECONDARY_TDM_TX_6
* - #AFE_PORT_ID_SECONDARY_TDM_TX_7
* - #AFE_PORT_ID_TERTIARY_TDM_RX
* - #AFE_PORT_ID_TERTIARY_TDM_RX_1
* - #AFE_PORT_ID_TERTIARY_TDM_RX_2
* - #AFE_PORT_ID_TERTIARY_TDM_RX_3
* - #AFE_PORT_ID_TERTIARY_TDM_RX_4
* - #AFE_PORT_ID_TERTIARY_TDM_RX_5
* - #AFE_PORT_ID_TERTIARY_TDM_RX_6
* - #AFE_PORT_ID_TERTIARY_TDM_RX_7
* - #AFE_PORT_ID_TERTIARY_TDM_TX
* - #AFE_PORT_ID_TERTIARY_TDM_TX_1
* - #AFE_PORT_ID_TERTIARY_TDM_TX_2
* - #AFE_PORT_ID_TERTIARY_TDM_TX_3
* - #AFE_PORT_ID_TERTIARY_TDM_TX_4
* - #AFE_PORT_ID_TERTIARY_TDM_TX_5
* - #AFE_PORT_ID_TERTIARY_TDM_TX_6
* - #AFE_PORT_ID_TERTIARY_TDM_TX_7
* - #AFE_PORT_ID_QUATERNARY_TDM_RX
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_1
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_2
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_3
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_4
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_5
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_6
* - #AFE_PORT_ID_QUATERNARY_TDM_RX_7
* - #AFE_PORT_ID_QUATERNARY_TDM_TX
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_1
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_2
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_3
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_4
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_5
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_6
* - #AFE_PORT_ID_QUATERNARY_TDM_TX_7
* @tablebulletend
*/
u32 num_channels;
/* Number of enabled slots for TDM frame.
* @values 1 to 8
*/
u32 sample_rate;
/* Sampling rate of the port.
* @values
* - #AFE_PORT_SAMPLE_RATE_8K
* - #AFE_PORT_SAMPLE_RATE_16K
* - #AFE_PORT_SAMPLE_RATE_24K
* - #AFE_PORT_SAMPLE_RATE_32K
* - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend
*/
u32 bit_width;
/* Bit width of the sample.
* @values 16, 24, (32)
*/
u16 nslots_per_frame;
/* Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
* @values 1 - 32
*/
u16 slot_width;
/* Slot width of the slot in a TDM frame. (slot_width >= bit_width)
* have to be satisfied.
* @values 16, 24, 32
*/
u32 slot_mask;
/* Position of active slots. When that bit is set, that paricular
* slot is active.
* Number of active slots can be inferred by number of bits set in
* the mask. Only 8 individual bits can be enabled.
* Bits 0..31 corresponding to slot 0..31
* @values 1 to 2^32 -1
*/
} __packed;
/* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_ENABLE
* parameter, which enables or
* disables any module.
* The fixed size of this structure is four bytes.
*/
struct afe_group_device_enable {
u16 group_id;
/* valid value is AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX */
u16 enable;
/* Enables (1) or disables (0) the module. */
} __packed;
union afe_port_group_config {
struct afe_group_device_group_cfg group_cfg;
struct afe_group_device_enable group_enable;
struct afe_param_id_group_device_tdm_cfg tdm_cfg;
} __packed;
struct afe_port_group_create {
struct apr_hdr hdr;
struct afe_svc_cmd_set_param param;
struct afe_port_param_data_v2 pdata;
union afe_port_group_config data;
} __packed;
/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to specify
* the timing statistics of the corresponding device interface.
* Client can periodically query for the device time statistics to help adjust
* the PLL based on the drift value. The get param command must be sent to
* AFE port ID corresponding to device interface
* This parameter ID supports following get param commands:
* #AFE_PORT_CMD_GET_PARAM_V2 and
* #AFE_PORT_CMD_GET_PARAM_V3.
*/
#define AFE_PARAM_ID_DEV_TIMING_STATS 0x000102AD
/* Version information used to handle future additions to AFE device
* interface timing statistics (for backward compatibility).
*/
#define AFE_API_VERSION_DEV_TIMING_STATS 0x1
/* Enumeration for specifying a sink(Rx) device */
#define AFE_SINK_DEVICE 0x0
/* Enumeration for specifying a source(Tx) device */
#define AFE_SOURCE_DEVICE 0x1
/* Enumeration for specifying the drift reference is of type AV Timer */
#define AFE_REF_TIMER_TYPE_AVTIMER 0x0
/* Message payload structure for the
* AFE_PARAM_ID_DEV_TIMING_STATS parameter.
*/
struct afe_param_id_dev_timing_stats {
/* Minor version used to track the version of device interface timing
* statistics. Currently, the supported version is 1.
* @values #AFE_API_VERSION_DEV_TIMING_STATS
*/
u32 minor_version;
/* Indicates the device interface direction as either
* source (Tx) or sink (Rx).
* @values
* #AFE_SINK_DEVICE
* #AFE_SOURCE_DEVICE
*/
u16 device_direction;
/* Reference timer for drift accumulation and time stamp information.
* @values
* #AFE_REF_TIMER_TYPE_AVTIMER @tablebulletend
*/
u16 reference_timer;
/*
* Flag to indicate if resync is required on the client side for
* drift correction. Flag is set to TRUE for the first get_param
* response after device interface starts. This flag value can be
* used by client to identify if device interface restart has
* happened and if any re-sync is required at their end for drift
* correction.
* @values
* 0: FALSE (Resync not required)
* 1: TRUE (Resync required) @tablebulletend
*/
u32 resync_flag;
/* Accumulated drift value in microseconds. This value is updated
* every 100th ms.
* Positive drift value indicates AV timer is running faster than device
* Negative drift value indicates AV timer is running slower than device
* @values Any valid int32 number
*/
s32 acc_drift_value;
/* Lower 32 bits of the 64-bit absolute timestamp of reference
* timer in microseconds.
* This timestamp corresponds to the time when the drift values
* are accumlated for every 100th ms.
* @values Any valid uint32 number
*/
u32 ref_timer_abs_ts_lsw;
/* Upper 32 bits of the 64-bit absolute timestamp of reference
* timer in microseconds.
* This timestamp corresponds to the time when the drift values
* are accumlated for every 100th ms.
* @values Any valid uint32 number
*/
u32 ref_timer_abs_ts_msw;
} __packed;
struct afe_av_dev_drift_get_param {
struct apr_hdr hdr;
struct afe_port_cmd_get_param_v2 get_param;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_dev_timing_stats timing_stats;
} __packed;
struct afe_av_dev_drift_get_param_resp {
uint32_t status;
struct afe_port_param_data_v2 pdata;
struct afe_param_id_dev_timing_stats timing_stats;
} __packed;
/* Command for Matrix or Stream Router */
#define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2 0x00010DCE
/* Module for AVSYNC */
#define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC 0x00010DC6
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
* render window start value. This parameter is supported only for a Set
* command (not a Get command) in the Rx direction
* (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2).
* Render window start is a value (session time minus timestamp, or ST-TS)
* below which frames are held, and after which frames are immediately
* rendered.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
* render window end value. This parameter is supported only for a Set
* command (not a Get command) in the Rx direction
* (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value
* (session time minus timestamp) above which frames are dropped, and below
* which frames are immediately rendered.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 0x00010DD2
/* Generic payload of the window parameters in the
* #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module.
* This payload is supported only for a Set command
* (not a Get command) on the Rx path.
*/
struct asm_session_mtmx_strtr_param_window_v2_t {
u32 window_lsw;
/* Lower 32 bits of the render window start value. */
u32 window_msw;
/* Upper 32 bits of the render window start value.
*
* The 64-bit number formed by window_lsw and window_msw specifies a
* signed 64-bit window value in microseconds. The sign extension is
* necessary. This value is used by the following parameter IDs:
* #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2
* #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2
* #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2
* #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2
* The value depends on which parameter ID is used.
* The aDSP honors the windows at a granularity of 1 ms.
*/
};
struct asm_session_cmd_set_mtmx_strstr_params_v2 {
uint32_t data_payload_addr_lsw;
/* Lower 32 bits of the 64-bit data payload address. */
uint32_t data_payload_addr_msw;
/* Upper 32 bits of the 64-bit data payload address.
* If the address is not sent (NULL), the message is in the payload.
* If the address is sent (non-NULL), the parameter data payloads
* begin at the specified address.
*/
uint32_t mem_map_handle;
/* Unique identifier for an address. This memory map handle is returned
* by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
* values
* - NULL -- Parameter data payloads are within the message payload
* (in-band).
* - Non-NULL -- Parameter data payloads begin at the address specified
* in the data_payload_addr_lsw and data_payload_addr_msw fields
* (out-of-band).
*/
uint32_t data_payload_size;
/* Actual size of the variable payload accompanying the message, or in
* shared memory. This field is used for parsing the parameter payload.
* values > 0 bytes
*/
uint32_t direction;
/* Direction of the entity (matrix mixer or stream router) on which
* the parameter is to be set.
* values
* - 0 -- Rx (for Rx stream router or Rx matrix mixer)
* - 1 -- Tx (for Tx stream router or Tx matrix mixer)
*/
};
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
* audio client choose the rendering decision that the audio DSP should use.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD 0x00012F0D
/* Indicates that rendering decision will be based on default rate
* (session clock based rendering, device driven).
* 1. The default session clock based rendering is inherently driven
* by the timing of the device.
* 2. After the initial decision is made (first buffer after a run
* command), subsequent data rendering decisions are made with
* respect to the rate at which the device is rendering, thus deriving
* its timing from the device.
* 3. While this decision making is simple, it has some inherent limitations
* (mentioned in the next section).
* 4. If this API is not set, the session clock based rendering will be assumed
* and this will ensure that the DSP is backward compatible.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT 0
/* Indicates that rendering decision will be based on local clock rate.
* 1. In the DSP loopback/client loopback use cases (frame based
* inputs), the incoming data into audio DSP is time-stamped at the
* local clock rate (STC).
* 2. This TS rate may match the incoming data rate or maybe different
* from the incoming data rate.
* 3. Regardless, the data will be time-stamped with local STC and
* therefore, the client is recommended to set this mode for these
* use cases. This method is inherently more robust to sequencing
* (AFE Start/Stop) and device switches, among other benefits.
* 4. This API will inform the DSP to compare every incoming buffer TS
* against local STC.
* 5. DSP will continue to honor render windows APIs, as before.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC 1
/* Structure for rendering decision parameter */
struct asm_session_mtmx_strtr_param_render_mode_t {
/* Specifies the type of rendering decision the audio DSP should use.
*
* @values
* - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT
* - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC
*/
u32 flags;
} __packed;
/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
* audio client to specify the clock recovery mechanism that the audio DSP
* should use.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD 0x00012F0E
/* Indicates that default clock recovery will be used (no clock recovery).
* If the client wishes that no clock recovery be done, the client can
* choose this. This means that no attempt will made by the DSP to try and
* match the rates of the input and output audio.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE 0
/* Indicates that independent clock recovery needs to be used.
* 1. In the DSP loopback/client loopback use cases (frame based inputs),
* the client should choose the independent clock recovery option.
* 2. This basically de-couples the audio and video from knowing each others
* clock sources and lets the audio DSP independently rate match the input
* and output rates.
* 3. After drift detection, the drift correction is achieved by either pulling
* the PLLs (if applicable) or by stream to device rate matching
* (for PCM use cases) by comparing drift with respect to STC.
* 4. For passthrough use cases, since the PLL pulling is the only option,
* a best effort will be made.
* If PLL pulling is not possible / available, the rendering will be
* done without rate matching.
*/
#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO 1
/* Payload of the #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC parameter.
*/
struct asm_session_mtmx_strtr_param_clk_rec_t {
/* Specifies the type of clock recovery that the audio DSP should
* use for rate matching.
*/
/* @values
* #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_DEFAULT
* #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_INDEPENDENT
*/
u32 flags;
} __packed;
union asm_session_mtmx_strtr_param_config {
struct asm_session_mtmx_strtr_param_window_v2_t window_param;
struct asm_session_mtmx_strtr_param_render_mode_t render_param;
struct asm_session_mtmx_strtr_param_clk_rec_t clk_rec_param;
} __packed;
struct asm_mtmx_strtr_params {
struct apr_hdr hdr;
struct asm_session_cmd_set_mtmx_strstr_params_v2 param;
struct asm_stream_param_data_v2 data;
union asm_session_mtmx_strtr_param_config config;
} __packed;
#define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF
#define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0
#define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B
#define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL)
struct asm_session_cmd_get_mtmx_strstr_params_v2 {
uint32_t data_payload_addr_lsw;
/* Lower 32 bits of the 64-bit data payload address. */
uint32_t data_payload_addr_msw;
/*
* Upper 32 bits of the 64-bit data payload address.
* If the address is not sent (NULL), the message is in the payload.
* If the address is sent (non-NULL), the parameter data payloads
* begin at the specified address.
*/
uint32_t mem_map_handle;
/*
* Unique identifier for an address. This memory map handle is returned
* by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
* values
* - NULL -- Parameter data payloads are within the message payload
* (in-band).
* - Non-NULL -- Parameter data payloads begin at the address specified
* in the data_payload_addr_lsw and data_payload_addr_msw fields
* (out-of-band).
*/
uint32_t direction;
/*
* Direction of the entity (matrix mixer or stream router) on which
* the parameter is to be set.
* values
* - 0 -- Rx (for Rx stream router or Rx matrix mixer)
* - 1 -- Tx (for Tx stream router or Tx matrix mixer)
*/
uint32_t module_id;
/* Unique module ID. */
uint32_t param_id;
/* Unique parameter ID. */
uint32_t param_max_size;
};
struct asm_session_mtmx_strtr_param_session_time_v3_t {
uint32_t session_time_lsw;
/* Lower 32 bits of the current session time in microseconds */
uint32_t session_time_msw;
/*
* Upper 32 bits of the current session time in microseconds.
* The 64-bit number formed by session_time_lsw and session_time_msw
* is treated as signed.
*/
uint32_t absolute_time_lsw;
/*
* Lower 32 bits of the 64-bit absolute time in microseconds.
* This is the time when the sample corresponding to the
* session_time_lsw is rendered to the hardware. This absolute
* time can be slightly in the future or past.
*/
uint32_t absolute_time_msw;
/*
* Upper 32 bits of the 64-bit absolute time in microseconds.
* This is the time when the sample corresponding to the
* session_time_msw is rendered to hardware. This absolute
* time can be slightly in the future or past. The 64-bit number
* formed by absolute_time_lsw and absolute_time_msw is treated as
* unsigned.
*/
uint32_t time_stamp_lsw;
/* Lower 32 bits of the last processed timestamp in microseconds */
uint32_t time_stamp_msw;
/*
* Upper 32 bits of the last processed timestamp in microseconds.
* The 64-bit number formed by time_stamp_lsw and time_stamp_lsw
* is treated as unsigned.
*/
uint32_t flags;
/*
* Keeps track of any additional flags needed.
* @values{for bit 31}
* - 0 -- Uninitialized/invalid
* - 1 -- Valid
* All other bits are reserved; clients must set them to zero.
*/
};
union asm_session_mtmx_strtr_data_type {
struct asm_session_mtmx_strtr_param_session_time_v3_t session_time;
};
struct asm_mtmx_strtr_get_params {
struct apr_hdr hdr;
struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info;
} __packed;
struct asm_mtmx_strtr_get_params_cmdrsp {
uint32_t err_code;
struct asm_stream_param_data_v2 param_info;
union asm_session_mtmx_strtr_data_type param_data;
} __packed;
#define AUDPROC_MODULE_ID_RESAMPLER 0x00010719
enum {
LEGACY_PCM = 0,
COMPRESSED_PASSTHROUGH,
COMPRESSED_PASSTHROUGH_CONVERT,
COMPRESSED_PASSTHROUGH_DSD,
LISTEN,
COMPRESSED_PASSTHROUGH_GEN,
};
#define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770
#define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771
struct adm_set_compressed_device_mute {
struct adm_cmd_set_pp_params_v5 command;
struct adm_param_data_v5 params;
u32 mute_on;
} __packed;
#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E
#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F
struct adm_set_compressed_device_latency {
struct adm_cmd_set_pp_params_v5 command;
struct adm_param_data_v5 params;
u32 latency;
} __packed;
#define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6
#define VOICEPROC_PARAM_ID_FLUENCE_SOUNDFOCUS 0x00010E37
#define VOICEPROC_PARAM_ID_FLUENCE_SOURCETRACKING 0x00010E38
#define MAX_SECTORS 8
#define MAX_NOISE_SOURCE_INDICATORS 3
#define MAX_POLAR_ACTIVITY_INDICATORS 360
struct sound_focus_param {
uint16_t start_angle[MAX_SECTORS];
uint8_t enable[MAX_SECTORS];
uint16_t gain_step;
} __packed;
struct source_tracking_param {
uint8_t vad[MAX_SECTORS];
uint16_t doa_speech;
uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
} __packed;
struct adm_param_fluence_soundfocus_t {
uint16_t start_angles[MAX_SECTORS];
uint8_t enables[MAX_SECTORS];
uint16_t gain_step;
uint16_t reserved;
} __packed;
struct adm_set_fluence_soundfocus_param {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
struct adm_param_fluence_soundfocus_t soundfocus_data;
} __packed;
struct adm_param_fluence_sourcetracking_t {
uint8_t vad[MAX_SECTORS];
uint16_t doa_speech;
uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
} __packed;
#define AUDPROC_MODULE_ID_AUDIOSPHERE 0x00010916
#define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE 0x00010917
#define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH 0x00010918
#define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE 0x00010919
#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT 0x0001091A
#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT 0x0001091B
#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT 0x0001091C
#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT 0x0001091D
#define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO 0x0001091E
#define AUDPROC_MODULE_ID_VOICE_TX_SECNS 0x10027059
#define AUDPROC_PARAM_IDX_SEC_PRIMARY_MIC_CH 0x10014444
struct admx_sec_primary_mic_ch {
uint16_t version;
uint16_t reserved;
uint16_t sec_primary_mic_ch;
uint16_t reserved1;
} __packed;
struct adm_set_sec_primary_ch_params {
struct adm_cmd_set_pp_params_v5 params;
struct adm_param_data_v5 data;
struct admx_sec_primary_mic_ch sec_primary_mic_ch_data;
} __packed;
#endif /*_APR_AUDIO_V2_H_ */