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Diffstat (limited to 'include/sound/apr_audio-v2.h')
| -rw-r--r-- | include/sound/apr_audio-v2.h | 9887 |
1 files changed, 9887 insertions, 0 deletions
diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h new file mode 100644 index 000000000000..06d952a07c2a --- /dev/null +++ b/include/sound/apr_audio-v2.h @@ -0,0 +1,9887 @@ +/* Copyright (c) 2012-2016, The Linux Foundation. All rights reserved. +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 and +* only version 2 as published by the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +*/ + + +#ifndef _APR_AUDIO_V2_H_ +#define _APR_AUDIO_V2_H_ + +#include <linux/qdsp6v2/apr.h> + +/* size of header needed for passing data out of band */ +#define APR_CMD_OB_HDR_SZ 12 + +/* size of header needed for getting data */ +#define APR_CMD_GET_HDR_SZ 16 + +struct param_outband { + size_t size; + void *kvaddr; + phys_addr_t paddr; +}; + +#define ADSP_ADM_VERSION 0x00070000 + +#define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322 +#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323 +#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324 + +#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325 +#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D +/* Enumeration for an audio Rx matrix ID.*/ +#define ADM_MATRIX_ID_AUDIO_RX 0 + +#define ADM_MATRIX_ID_AUDIO_TX 1 + +#define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX 2 +/* Enumeration for an audio Tx matrix ID.*/ +#define ADM_MATRIX_ID_AUDIOX 1 + +#define ADM_MAX_COPPS 5 + +/* make sure this matches with msm_audio_calibration */ +#define SP_V2_NUM_MAX_SPKR 2 + +/* Session map node structure. +* Immediately following this structure are num_copps +* entries of COPP IDs. The COPP IDs are 16 bits, so +* there might be a padding 16-bit field if num_copps +* is odd. +*/ +struct adm_session_map_node_v5 { + u16 session_id; +/* Handle of the ASM session to be routed. Supported values: 1 +* to 8. +*/ + + + u16 num_copps; + /* Number of COPPs to which this session is to be routed. + Supported values: 0 < num_copps <= ADM_MAX_COPPS. + */ +} __packed; + +/* Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command. +* Immediately following this structure are num_sessions of the session map +* node payload (adm_session_map_node_v5). +*/ + +struct adm_cmd_matrix_map_routings_v5 { + struct apr_hdr hdr; + + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx +* (1). Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX +* macros to set this field. +*/ + u32 num_sessions; + /* Number of sessions being updated by this command (optional).*/ +} __packed; + +/* This command allows a client to open a COPP/Voice Proc. TX module +* and sets up the device session: Matrix -> COPP -> AFE on the RX +* and AFE -> COPP -> Matrix on the TX. This enables PCM data to +* be transferred to/from the endpoint (AFEPortID). +* +* @return +* #ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and +* COPP ID. +*/ +#define ADM_CMD_DEVICE_OPEN_V5 0x00010326 + +/* Definition for a low latency stream session. */ +#define ADM_LOW_LATENCY_DEVICE_SESSION 0x2000 + +/* Definition for a ultra low latency stream session. */ +#define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION 0x4000 + +/* Definition for a ultra low latency with Post Processing stream session. */ +#define ADM_ULL_POST_PROCESSING_DEVICE_SESSION 0x8000 + +/* Definition for a legacy device session. */ +#define ADM_LEGACY_DEVICE_SESSION 0 + +/* Indicates that endpoint_id_2 is to be ignored.*/ +#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1 + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2 + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3 + +/* Indicates that an audio COPP is to send/receive a mono PCM + * stream to/from + * END_POINT_ID_1. + */ +#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1 + +/* Indicates that an audio COPP is to send/receive a + * stereo PCM stream to/from END_POINT_ID_1. + */ +#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2 + +/* Sample rate is 8000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000 + +/* Sample rate is 16000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000 + +/* Sample rate is 48000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000 + +/* Definition for a COPP live input flag bitmask.*/ +#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U) + +/* Definition for a COPP live shift value bitmask.*/ +#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0 + +/* Definition for the COPP ID bitmask.*/ +#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL) + +/* Definition for the COPP ID shift value.*/ +#define ADM_SHIFT_COPP_ID 0 + +/* Definition for the service ID bitmask.*/ +#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL) + +/* Definition for the service ID shift value.*/ +#define ADM_SHIFT_SERVICE_ID 16 + +/* Definition for the domain ID bitmask.*/ +#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL) + +/* Definition for the domain ID shift value.*/ +#define ADM_SHIFT_DOMAIN_ID 24 + +/* ADM device open command payload of the + #ADM_CMD_DEVICE_OPEN_V5 command. +*/ +struct adm_cmd_device_open_v5 { + struct apr_hdr hdr; + u16 flags; +/* Reserved for future use. Clients must set this field + * to zero. + */ + + u16 mode_of_operation; +/* Specifies whether the COPP must be opened on the Tx or Rx + * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for + * supported values and interpretation. + * Supported values: + * - 0x1 -- Rx path COPP + * - 0x2 -- Tx path live COPP + * - 0x3 -- Tx path nonlive COPP + * Live connections cause sample discarding in the Tx device + * matrix if the destination output ports do not pull them + * fast enough. Nonlive connections queue the samples + * indefinitely. + */ + + u16 endpoint_id_1; +/* Logical and physical endpoint ID of the audio path. + * If the ID is a voice processor Tx block, it receives near + * samples. Supported values: Any pseudoport, AFE Rx port, + * or AFE Tx port For a list of valid IDs, refer to + * @xhyperref{Q4,[Q4]}. + * Q4 = Hexagon Multimedia: AFE Interface Specification + */ + + u16 endpoint_id_2; +/* Logical and physical endpoint ID 2 for a voice processor + * Tx block. + * This is not applicable to audio COPP. + * Supported values: + * - AFE Rx port + * - 0xFFFF -- Endpoint 2 is unavailable and the voice + * processor Tx + * block ignores this endpoint + * When the voice processor Tx block is created on the audio + * record path, + * it can receive far-end samples from an AFE Rx port if the + * voice call + * is active. The ID of the AFE port is provided in this + * field. + * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. + */ + + u32 topology_id; + /* Audio COPP topology ID; 32-bit GUID. */ + + u16 dev_num_channel; +/* Number of channels the audio COPP sends to/receives from + * the endpoint. + * Supported values: 1 to 8. + * The value is ignored for the voice processor Tx block, + * where channel + * configuration is derived from the topology ID. + */ + + u16 bit_width; +/* Bit width (in bits) that the audio COPP sends to/receives + * from the + * endpoint. The value is ignored for the voice processing + * Tx block, + * where the PCM width is 16 bits. + */ + + u32 sample_rate; +/* Sampling rate at which the audio COPP/voice processor + * Tx block + * interfaces with the endpoint. + * Supported values for voice processor Tx: 8000, 16000, + * 48000 Hz + * Supported values for audio COPP: >0 and <=192 kHz + */ + + u8 dev_channel_mapping[8]; +/* Array of channel mapping of buffers that the audio COPP + * sends to the endpoint. Channel[i] mapping describes channel + * I inside the buffer, where 0 < i < dev_num_channel. + * This value is relevent only for an audio Rx COPP. + * For the voice processor block and Tx audio block, this field + * is set to zero and is ignored. + */ +} __packed; + +/* + * This command allows the client to close a COPP and disconnect + * the device session. + */ +#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327 + +/* Sets one or more parameters to a COPP. +*/ +#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328 + +/* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command. + * If the data_payload_addr_lsw and data_payload_addr_msw element + * are NULL, a series of adm_param_datastructures immediately + * follows, whose total size is data_payload_size bytes. + */ +struct adm_cmd_set_pp_params_v5 { + struct apr_hdr hdr; + u32 payload_addr_lsw; + /* LSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* MSW of parameter data payload address.*/ + + u32 mem_map_handle; +/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS + * command */ +/* If mem_map_handle is zero implies the message is in + * the payload */ + + u32 payload_size; +/* Size in bytes of the variable payload accompanying this + * message or + * in shared memory. This is used for parsing the parameter + * payload. + */ +} __packed; + +/* Payload format for COPP parameter data. + * Immediately following this structure are param_size bytes + * of parameter + * data. + */ +struct adm_param_data_v5 { + u32 module_id; + /* Unique ID of the module. */ + u32 param_id; + /* Unique ID of the parameter. */ + u16 param_size; + /* Data size of the param_id/module_id combination. + This value is a + multiple of 4 bytes. */ + u16 reserved; + /* Reserved for future enhancements. + * This field must be set to zero. + */ +} __packed; + +/* set customized mixing on matrix mixer */ +#define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 0x00010344 +struct adm_cmd_set_pspd_mtmx_strtr_params_v5 { + struct apr_hdr hdr; + /* LSW of parameter data payload address.*/ + u32 payload_addr_lsw; + /* MSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */ + /* command. If mem_map_handle is zero implies the message is in */ + /* the payload */ + u32 mem_map_handle; + /* Size in bytes of the variable payload accompanying this */ + /* message or in shared memory. This is used for parsing the */ + /* parameter payload. */ + u32 payload_size; + u16 direction; + u16 sessionid; + u16 deviceid; + u16 reserved; +} __packed; + +/* Defined specifically for in-band use, includes params */ +struct adm_cmd_set_pp_params_inband_v5 { + struct apr_hdr hdr; + /* LSW of parameter data payload address.*/ + u32 payload_addr_lsw; + /* MSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */ + /* command. If mem_map_handle is zero implies the message is in */ + /* the payload */ + u32 mem_map_handle; + /* Size in bytes of the variable payload accompanying this */ + /* message or in shared memory. This is used for parsing the */ + /* parameter payload. */ + u32 payload_size; + /* Parameters passed for in band payload */ + struct adm_param_data_v5 params; +} __packed; + +/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. + */ +#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 + +/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message, + * which returns the + * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. + */ +struct adm_cmd_rsp_device_open_v5 { + u32 status; + /* Status message (error code).*/ + + u16 copp_id; + /* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/ + + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +/* This command allows a query of one COPP parameter. +*/ +#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A + +/* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command. +*/ +struct adm_cmd_get_pp_params_v5 { + struct apr_hdr hdr; + u32 data_payload_addr_lsw; + /* LSW of parameter data payload address.*/ + + u32 data_payload_addr_msw; + /* MSW of parameter data payload address.*/ + + /* If the mem_map_handle is non zero, + * on ACK, the ParamData payloads begin at + * the address specified (out-of-band). + */ + + u32 mem_map_handle; + /* Memory map handle returned + * by ADM_CMD_SHARED_MEM_MAP_REGIONS command. + * If the mem_map_handle is 0, it implies that + * the ACK's payload will contain the ParamData (in-band). + */ + + u32 module_id; + /* Unique ID of the module. */ + + u32 param_id; + /* Unique ID of the parameter. */ + + u16 param_max_size; + /* Maximum data size of the parameter + *ID/module ID combination. This + * field is a multiple of 4 bytes. + */ + u16 reserved; + /* Reserved for future enhancements. + * This field must be set to zero. + */ +} __packed; + +/* Returns parameter values + * in response to an #ADM_CMD_GET_PP_PARAMS_V5 command. + */ +#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B + +/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message, + * which returns parameter values in response + * to an #ADM_CMD_GET_PP_PARAMS_V5 command. + * Immediately following this + * structure is the adm_param_data_v5 + * structure containing the pre/postprocessing + * parameter data. For an in-band + * scenario, the variable payload depends + * on the size of the parameter. +*/ +struct adm_cmd_rsp_get_pp_params_v5 { + u32 status; + /* Status message (error code).*/ +} __packed; + +/* Structure for holding soft stepping volume parameters. */ + +/* + * Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * parameters used by the Volume Control module. + */ + +struct audproc_softvolume_params { + u32 period; + u32 step; + u32 rampingcurve; +} __packed; + +/* + * 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_TX_VI 0x4f09 +#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>® + * </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 + +struct afe_sidetone_iir_filter_config_params { + u16 num_biquad_stages; +/* Number of stages. + * Supported values: Minimum of 5 and maximum of 10 + */ + + u16 pregain; +/* Pregain for the compensating filter response. + * Supported values: Any number in Q13 format + */ +} __packed; + +#define AFE_MODULE_LOOPBACK 0x00010205 +#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206 + +/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter, + * which gets/sets loopback gain of a port to an Rx port. + * The Tx port ID of the loopback is part of the set_param command. + */ + +/* Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's + * configuration/calibration settings for the AFE port. + */ +struct afe_port_cmd_set_param_v2 { + u16 port_id; +/* Port interface and direction (Rx or Tx) to start. + */ + + u16 payload_size; +/* Actual size of the payload in bytes. + * This is used for parsing the parameter payload. + * Supported values: > 0 + */ + +u32 payload_address_lsw; +/* LSW of 64 bit Payload address. + * Address should be 32-byte, + * 4kbyte aligned and must be contiguous memory. + */ + +u32 payload_address_msw; +/* MSW of 64 bit Payload address. + * In case of 32-bit shared memory address, + * this field must be set to zero. + * In case of 36-bit shared memory address, + * bit-4 to bit-31 must be set to zero. + * Address should be 32-byte, 4kbyte aligned + * and must be contiguous memory. + */ + +u32 mem_map_handle; +/* Memory map handle returned by + * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands. + * Supported Values: + * - NULL -- Message. The parameter data is in-band. + * - Non-NULL -- The parameter data is Out-band.Pointer to + * the physical address + * in shared memory of the payload data. + * An optional field is available if parameter + * data is in-band: + * afe_param_data_v2 param_data[...]. + * For detailed payload content, see the + * afe_port_param_data_v2 structure. + */ +} __packed; + +#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF + +struct afe_port_param_data_v2 { + u32 module_id; +/* ID of the module to be configured. + * Supported values: Valid module ID + */ + +u32 param_id; +/* ID of the parameter corresponding to the supported parameters + * for the module ID. + * Supported values: Valid parameter ID + */ + +u16 param_size; +/* Actual size of the data for the + * module_id/param_id pair. The size is a + * multiple of four bytes. + * Supported values: > 0 + */ + +u16 reserved; +/* This field must be set to zero. + */ +} __packed; + +struct afe_loopback_gain_per_path_param { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + u16 rx_port_id; +/* Rx port of the loopback. */ + +u16 gain; +/* Loopback gain per path of the port. + * Supported values: Any number in Q13 format + */ +} __packed; + +/* Parameter ID used to configure and enable/disable the + * loopback path. The difference with respect to the existing + * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be + * configured as source port in loopback path. Port-id in + * AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be + * Tx or Rx port. In addition, we can configure the type of + * routing mode to handle different use cases. + */ +#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B +#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1 + +enum afe_loopback_routing_mode { + LB_MODE_DEFAULT = 1, + /* Regular loopback from source to destination port */ + LB_MODE_SIDETONE, + /* Sidetone feed from Tx source to Rx destination port */ + LB_MODE_EC_REF_VOICE_AUDIO, + /* Echo canceller reference, voice + audio + DTMF */ + LB_MODE_EC_REF_VOICE + /* Echo canceller reference, voice alone */ +} __packed; + +/* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG , + * which enables/disables one AFE loopback. + */ +struct afe_loopback_cfg_v1 { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + u32 loopback_cfg_minor_version; +/* Minor version used for tracking the version of the RMC module + * configuration interface. + * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG + */ + u16 dst_port_id; + /* Destination Port Id. */ + u16 routing_mode; +/* Specifies data path type from src to dest port. + * Supported values: + * #LB_MODE_DEFAULT + * #LB_MODE_SIDETONE + * #LB_MODE_EC_REF_VOICE_AUDIO + * #LB_MODE_EC_REF_VOICE_A + * #LB_MODE_EC_REF_VOICE + */ + + u16 enable; +/* Specifies whether to enable (1) or + * disable (0) an AFE loopback. + */ + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0. + */ + +} __packed; + +#define AFE_MODULE_SPEAKER_PROTECTION 0x00010209 +#define AFE_PARAM_ID_SPKR_PROT_CONFIG 0x0001020a +#define AFE_API_VERSION_SPKR_PROT_CONFIG 0x1 +#define AFE_SPKR_PROT_EXCURSIONF_LEN 512 +struct afe_spkr_prot_cfg_param_v1 { + u32 spkr_prot_minor_version; +/* + * Minor version used for tracking the version of the + * speaker protection module configuration interface. + * Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG + */ + +int16_t win_size; +/* Analysis and synthesis window size (nWinSize). + * Supported values: 1024, 512, 256 samples + */ + +int16_t margin; +/* Allowable margin for excursion prediction, + * in L16Q15 format. This is a + * control parameter to allow + * for overestimation of peak excursion. + */ + +int16_t spkr_exc_limit; +/* Speaker excursion limit, in L16Q15 format.*/ + +int16_t spkr_resonance_freq; +/* Resonance frequency of the speaker; used + * to define a frequency range + * for signal modification. + * + * Supported values: 0 to 2000 Hz */ + +int16_t limhresh; +/* Threshold of the hard limiter; used to + * prevent overshooting beyond a + * signal level that was set by the limiter + * prior to speaker protection. + * Supported values: 0 to 32767 + */ + +int16_t hpf_cut_off_freq; +/* High pass filter cutoff frequency. + * Supported values: 100, 200, 300 Hz + */ + +int16_t hpf_enable; +/* Specifies whether the high pass filter + * is enabled (0) or disabled (1). + */ + +int16_t reserved; +/* This field must be set to zero. */ + +int32_t amp_gain; +/* Amplifier gain in L32Q15 format. + * This is the RMS voltage at the + * loudspeaker when a 0dBFS tone + * is played in the digital domain. + */ + +int16_t excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN]; +/* Array of the excursion transfer function. + * The peak excursion of the + * loudspeaker diaphragm is + * measured in millimeters for 1 Vrms Sine + * tone at all FFT bin frequencies. + * Supported values: Q15 format + */ +} __packed; + + +#define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER 0x000100E0 + +/* Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER + * command, which registers a real-time port driver + * with the AFE service. + */ +struct afe_service_cmd_register_rt_port_driver { + struct apr_hdr hdr; + u16 port_id; +/* Port ID with which the real-time driver exchanges data + * (registers for events). + * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to + * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END + */ + + u16 reserved; + /* This field must be set to zero. */ +} __packed; + +#define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER 0x000100E1 + +/* Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER + * command, which unregisters a real-time port driver from + * the AFE service. + */ +struct afe_service_cmd_unregister_rt_port_driver { + struct apr_hdr hdr; + u16 port_id; +/* Port ID from which the real-time + * driver unregisters for events. + * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to + * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END + */ + + u16 reserved; + /* This field must be set to zero. */ +} __packed; + +#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105 +#define AFE_EVENTYPE_RT_PROXY_PORT_START 0 +#define AFE_EVENTYPE_RT_PROXY_PORT_STOP 1 +#define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK 2 +#define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK 3 +#define AFE_EVENTYPE_RT_PROXY_PORT_INVALID 0xFFFF + +/* Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS + * message, which sends an event from the AFE service + * to a registered client. + */ +struct afe_event_rt_proxy_port_status { + u16 port_id; +/* Port ID to which the event is sent. + * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to + * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END + */ + + u16 eventype; +/* Type of event. + * Supported values: + * - #AFE_EVENTYPE_RT_PROXY_PORT_START + * - #AFE_EVENTYPE_RT_PROXY_PORT_STOP + * - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK + * - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK + */ +} __packed; + +#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED + +struct afe_port_data_cmd_rt_proxy_port_write_v2 { + struct apr_hdr hdr; + u16 port_id; +/* Tx (mic) proxy port ID with which the real-time + * driver exchanges data. + * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to + * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END + */ + + u16 reserved; + /* This field must be set to zero. */ + + u32 buffer_address_lsw; +/* LSW Address of the buffer containing the + * data from the real-time source + * device on a client. + */ + + u32 buffer_address_msw; +/* MSW Address of the buffer containing the + * data from the real-time source + * device on a client. + */ + + u32 mem_map_handle; +/* A memory map handle encapsulating shared memory + * attributes is returned if + * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS + * command is successful. + * Supported Values: + * - Any 32 bit value + */ + + u32 available_bytes; +/* Number of valid bytes available + * in the buffer (including all + * channels: number of bytes per + * channel = availableBytesumChannels). + * Supported values: > 0 + * + * This field must be equal to the frame + * size specified in the #AFE_PORT_AUDIO_IF_CONFIG + * command that was sent to configure this + * port. + */ +} __packed; + +#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 0x000100EE + +/* Payload of the + * #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which + * delivers an empty buffer to the AFE service. On + * acknowledgment, data is filled in the buffer. + */ +struct afe_port_data_cmd_rt_proxy_port_read_v2 { + struct apr_hdr hdr; + u16 port_id; +/* Rx proxy port ID with which the real-time + * driver exchanges data. + * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to + * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END + * (This must be an Rx (speaker) port.) + */ + + u16 reserved; + /* This field must be set to zero. */ + + u32 buffer_address_lsw; +/* LSW Address of the buffer containing the data sent from the AFE + * service to a real-time sink device on the client. + */ + + + u32 buffer_address_msw; +/* MSW Address of the buffer containing the data sent from the AFE + * service to a real-time sink device on the client. + */ + + u32 mem_map_handle; +/* A memory map handle encapsulating shared memory attributes is + * returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is + * successful. + * Supported Values: + * - Any 32 bit value + */ + + u32 available_bytes; +/* Number of valid bytes available in the buffer (including all + * channels). + * Supported values: > 0 + * This field must be equal to the frame size specified in the + * #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure + * this port. + */ +} __packed; + +/* This module ID is related to device configuring like I2S,PCM, + * HDMI, SLIMBus etc. This module supports follwing parameter ids. + * - #AFE_PARAM_ID_I2S_CONFIG + * - #AFE_PARAM_ID_PCM_CONFIG + * - #AFE_PARAM_ID_DIGI_MIC_CONFIG + * - #AFE_PARAM_ID_HDMI_CONFIG + * - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG + * - #AFE_PARAM_ID_SLIMBUS_CONFIG + * - #AFE_PARAM_ID_RT_PROXY_CONFIG + */ + +#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C +#define AFE_PORT_SAMPLE_RATE_8K 8000 +#define AFE_PORT_SAMPLE_RATE_16K 16000 +#define AFE_PORT_SAMPLE_RATE_48K 48000 +#define AFE_PORT_SAMPLE_RATE_96K 96000 +#define AFE_PORT_SAMPLE_RATE_192K 192000 +#define AFE_LINEAR_PCM_DATA 0x0 +#define AFE_NON_LINEAR_DATA 0x1 +#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2 +#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3 + +/* This param id is used to configure I2S interface */ +#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D +#define AFE_API_VERSION_I2S_CONFIG 0x1 +/* Enumeration for setting the I2S configuration + * channel_mode parameter to + * serial data wire number 1-3 (SD3). + */ +#define AFE_PORT_I2S_SD0 0x1 +#define AFE_PORT_I2S_SD1 0x2 +#define AFE_PORT_I2S_SD2 0x3 +#define AFE_PORT_I2S_SD3 0x4 +#define AFE_PORT_I2S_QUAD01 0x5 +#define AFE_PORT_I2S_QUAD23 0x6 +#define AFE_PORT_I2S_6CHS 0x7 +#define AFE_PORT_I2S_8CHS 0x8 +#define AFE_PORT_I2S_MONO 0x0 +#define AFE_PORT_I2S_STEREO 0x1 +#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0 +#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 + +/* Payload of the #AFE_PARAM_ID_I2S_CONFIG + * command's (I2S configuration + * parameter). + */ +struct afe_param_id_i2s_cfg { + u32 i2s_cfg_minor_version; +/* Minor version used for tracking the version of the I2S + * configuration interface. + * Supported values: #AFE_API_VERSION_I2S_CONFIG + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + + u16 channel_mode; +/* I2S lines and multichannel operation. + * Supported values: + * - #AFE_PORT_I2S_SD0 + * - #AFE_PORT_I2S_SD1 + * - #AFE_PORT_I2S_SD2 + * - #AFE_PORT_I2S_SD3 + * - #AFE_PORT_I2S_QUAD01 + * - #AFE_PORT_I2S_QUAD23 + * - #AFE_PORT_I2S_6CHS + * - #AFE_PORT_I2S_8CHS + */ + + u16 mono_stereo; +/* Specifies mono or stereo. This applies only when + * a single I2S line is used. + * Supported values: + * - #AFE_PORT_I2S_MONO + * - #AFE_PORT_I2S_STEREO + */ + + u16 ws_src; +/* Word select source: internal or external. + * Supported values: + * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL + * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K + */ + + u16 data_format; +/* data format + * Supported values: + * - #LINEAR_PCM_DATA + * - #NON_LINEAR_DATA + * - #LINEAR_PCM_DATA_PACKED_IN_60958 + * - #NON_LINEAR_DATA_PACKED_IN_60958 + */ + u16 reserved; + /* This field must be set to zero. */ +} __packed; + +/* + * This param id is used to configure PCM interface + */ + +#define AFE_API_VERSION_SPDIF_CONFIG 0x1 +#define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1 +#define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1 +#define AFE_CH_STATUS_A 1 +#define AFE_CH_STATUS_B 2 + +#define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244 +#define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245 +#define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246 + +#define AFE_PORT_CLK_ROOT_LPAPLL 0x3 +#define AFE_PORT_CLK_ROOT_LPAQ6PLL 0x4 + +struct afe_param_id_spdif_cfg { +/* Minor version used for tracking the version of the SPDIF + * configuration interface. + * Supported values: #AFE_API_VERSION_SPDIF_CONFIG + */ + u32 spdif_cfg_minor_version; + +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_22_05K + * - #AFE_PORT_SAMPLE_RATE_32K + * - #AFE_PORT_SAMPLE_RATE_44_1K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_176_4K + * - #AFE_PORT_SAMPLE_RATE_192K + */ + u32 sample_rate; + +/* data format + * Supported values: + * - #AFE_LINEAR_PCM_DATA + * - #AFE_NON_LINEAR_DATA + */ + u16 data_format; +/* Number of channels supported by the port + * - PCM - 1, Compressed Case - 2 + */ + u16 num_channels; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 bit_width; +/* This field must be set to zero. */ + u16 reserved; +} __packed; + +struct afe_param_id_spdif_ch_status_cfg { + u32 ch_status_cfg_minor_version; +/* Minor version used for tracking the version of channel + * status configuration. Current supported version is 1 + */ + + u32 status_type; +/* Indicate if the channel status is for channel A or B + * Supported values: + * - #AFE_CH_STATUS_A + * - #AFE_CH_STATUS_B + */ + + u8 status_bits[24]; +/* Channel status - 192 bits for channel + * Byte ordering as defined by IEC60958-3 + */ + + u8 status_mask[24]; +/* Channel status with mask bits 1 will be applied. + * Byte ordering as defined by IEC60958-3 + */ +} __packed; + +struct afe_param_id_spdif_clk_cfg { + u32 clk_cfg_minor_version; +/* Minor version used for tracking the version of SPDIF + * interface clock configuration. Current supported version + * is 1 + */ + + u32 clk_value; +/* Specifies the clock frequency in Hz to set + * Supported values: + * 0 - Disable the clock + * 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2 + * (channels A and B) + */ + + u32 clk_root; +/* Specifies SPDIF root clk source + * Supported Values: + * - #AFE_PORT_CLK_ROOT_LPAPLL + * - #AFE_PORT_CLK_ROOT_LPAQ6PLL + */ +} __packed; + +struct afe_spdif_clk_config_command { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + struct afe_param_id_spdif_clk_cfg clk_cfg; +} __packed; + +struct afe_spdif_chstatus_config_command { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + struct afe_param_id_spdif_ch_status_cfg ch_status; +} __packed; + +struct afe_spdif_port_config { + struct afe_param_id_spdif_cfg cfg; + struct afe_param_id_spdif_ch_status_cfg ch_status; +} __packed; + +#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E +#define AFE_API_VERSION_PCM_CONFIG 0x1 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an external source. + */ + +#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an internal source. + */ +#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1 +/* Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use + * short synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_PCM 0x0 +/* + * Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use long + * synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_AUX 0x1 +/* + * Enumeration for setting the PCM configuration frame to 8. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0 +/* + * Enumeration for setting the PCM configuration frame to 16. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1 + +/* Enumeration for setting the PCM configuration frame to 32.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2 + +/* Enumeration for setting the PCM configuration frame to 64.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3 + +/* Enumeration for setting the PCM configuration frame to 128.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4 + +/* Enumeration for setting the PCM configuration frame to 256.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5 + +/* Enumeration for setting the PCM configuration + * quantype parameter to A-law with no padding. + */ +#define AFE_PORT_PCM_ALAW_NOPADDING 0x0 + +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with no padding. + */ +#define AFE_PORT_PCM_MULAW_NOPADDING 0x1 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with no padding. + */ +#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2 +/* Enumeration for setting the PCM configuration quantype + * parameter to A-law with padding. + */ +#define AFE_PORT_PCM_ALAW_PADDING 0x3 +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with padding. + */ +#define AFE_PORT_PCM_MULAW_PADDING 0x4 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with padding. + */ +#define AFE_PORT_PCM_LINEAR_PADDING 0x5 +/* Enumeration for disabling the PCM configuration + * ctrl_data_out_enable parameter. + * The PCM block is the only master. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0 +/* + * Enumeration for enabling the PCM configuration + * ctrl_data_out_enable parameter. The PCM block shares + * the signal with other masters. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1 + +/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's + * (PCM configuration parameter). + */ + +struct afe_param_id_pcm_cfg { + u32 pcm_cfg_minor_version; +/* Minor version used for tracking the version of the AUX PCM + * configuration interface. + * Supported values: #AFE_API_VERSION_PCM_CONFIG + */ + + u16 aux_mode; +/* PCM synchronization setting. + * Supported values: + * - #AFE_PORT_PCM_AUX_MODE_PCM + * - #AFE_PORT_PCM_AUX_MODE_AUX + */ + + u16 sync_src; +/* Synchronization source. + * Supported values: + * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL + * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL + */ + + u16 frame_setting; +/* Number of bits per frame. + * Supported values: + * - #AFE_PORT_PCM_BITS_PER_FRAME_8 + * - #AFE_PORT_PCM_BITS_PER_FRAME_16 + * - #AFE_PORT_PCM_BITS_PER_FRAME_32 + * - #AFE_PORT_PCM_BITS_PER_FRAME_64 + * - #AFE_PORT_PCM_BITS_PER_FRAME_128 + * - #AFE_PORT_PCM_BITS_PER_FRAME_256 + */ + + u16 quantype; +/* PCM quantization type. + * Supported values: + * - #AFE_PORT_PCM_ALAW_NOPADDING + * - #AFE_PORT_PCM_MULAW_NOPADDING + * - #AFE_PORT_PCM_LINEAR_NOPADDING + * - #AFE_PORT_PCM_ALAW_PADDING + * - #AFE_PORT_PCM_MULAW_PADDING + * - #AFE_PORT_PCM_LINEAR_PADDING + */ + + u16 ctrl_data_out_enable; +/* Specifies whether the PCM block shares the data-out + * signal to the drive with other masters. + * Supported values: + * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE + * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE + */ + u16 reserved; + /* This field must be set to zero. */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to 4 + */ + + u16 slot_number_mapping[4]; +/* Specifies the slot number for the each channel in + * multi channel scenario. + * Supported values: 1 to 32 + */ +} __packed; + +/* + * This param id is used to configure DIGI MIC interface + */ +#define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F +/* This version information is used to handle the new + * additions to the config interface in future in backward + * compatible manner. + */ +#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to left 0. + */ + +#define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1 + +/*Enumeration for setting the digital mic configuration + * channel_mode parameter to right 0. + */ + + +#define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to left 1. + */ + +#define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to right 1. + */ + +#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to stereo 0. + */ +#define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to stereo 1. + */ + + +#define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6 + +/* Enumeration for setting the digital mic configuration + * channel_mode parameter to quad. + */ + +#define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7 + +/* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's + * (DIGI MIC configuration + * parameter). + */ +struct afe_param_id_digi_mic_cfg { + u32 digi_mic_cfg_minor_version; +/* Minor version used for tracking the version of the DIGI Mic + * configuration interface. + * Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 channel_mode; +/* Digital mic and multichannel operation. + * Supported values: + * - #AFE_PORT_DIGI_MIC_MODE_LEFT0 + * - #AFE_PORT_DIGI_MIC_MODE_RIGHT0 + * - #AFE_PORT_DIGI_MIC_MODE_LEFT1 + * - #AFE_PORT_DIGI_MIC_MODE_RIGHT1 + * - #AFE_PORT_DIGI_MIC_MODE_STEREO0 + * - #AFE_PORT_DIGI_MIC_MODE_STEREO1 + * - #AFE_PORT_DIGI_MIC_MODE_QUAD + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + */ +} __packed; + +/* +* This param id is used to configure HDMI interface +*/ +#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210 + +/* This version information is used to handle the new +* additions to the config interface in future in backward +* compatible manner. +*/ +#define AFE_API_VERSION_HDMI_CONFIG 0x1 + +/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command, + * which configures a multichannel HDMI audio interface. + */ +struct afe_param_id_hdmi_multi_chan_audio_cfg { + u32 hdmi_cfg_minor_version; +/* Minor version used for tracking the version of the HDMI + * configuration interface. + * Supported values: #AFE_API_VERSION_HDMI_CONFIG + */ + +u16 datatype; +/* data type + * Supported values: + * - #LINEAR_PCM_DATA + * - #NON_LINEAR_DATA + * - #LINEAR_PCM_DATA_PACKED_IN_60958 + * - #NON_LINEAR_DATA_PACKED_IN_60958 + */ + +u16 channel_allocation; +/* HDMI channel allocation information for programming an HDMI + * frame. The default is 0 (Stereo). + * + * This information is defined in the HDMI standard, CEA 861-D + * (refer to @xhyperref{S1,[S1]}). The number of channels is also + * inferred from this parameter. +*/ + + +u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - 22050, 44100, 176400 for compressed streams + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 reserved; + /* This field must be set to zero. */ +} __packed; + +/* +* This param id is used to configure BT or FM(RIVA) interface +*/ +#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211 + +/* This version information is used to handle the new +* additions to the config interface in future in backward +* compatible manner. +*/ +#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1 + +/* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG + * command's BT voice/BT audio/FM configuration parameter. + */ +struct afe_param_id_internal_bt_fm_cfg { + u32 bt_fm_cfg_minor_version; +/* Minor version used for tracking the version of the BT and FM + * configuration interface. + * Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to 2 + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO) + * - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO) + * - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP) + */ +} __packed; + +/* This param id is used to configure SLIMBUS interface using + * shared channel approach. + */ + + +#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 + +/* This version information is used to handle the new +* additions to the config interface in future in backward +* compatible manner. +*/ +#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 + +/* Enumeration for setting SLIMbus device ID 1. +*/ +#define AFE_SLIMBUS_DEVICE_1 0x0 + +/* Enumeration for setting SLIMbus device ID 2. +*/ +#define AFE_SLIMBUS_DEVICE_2 0x1 + +/* Enumeration for setting the SLIMbus data formats. +*/ +#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0 + +/* Enumeration for setting the maximum number of streams per + * device. + */ + +#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8 + +/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus + * port configuration parameter. + */ + +struct afe_param_id_slimbus_cfg { + u32 sb_cfg_minor_version; +/* Minor version used for tracking the version of the SLIMBUS + * configuration interface. + * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG + */ + + u16 slimbus_dev_id; +/* SLIMbus hardware device ID, which is required to handle + * multiple SLIMbus hardware blocks. + * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2 + */ + + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + + u16 data_format; +/* Data format supported by the SLIMbus hardware. The default is + * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the + * hardware does not perform any format conversions before the data + * transfer. + */ + + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT + */ + + u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; +/* Mapping of shared channel IDs (128 to 255) to which the + * master port is to be connected. + * Shared_channel_mapping[i] represents the shared channel assigned + * for audio channel i in multichannel audio data. + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K + */ +} __packed; + + +/* 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 + +/* 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; + +/* 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; +} __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; + struct afe_param_id_usb_audio_dev_params usb_dev; +} __packed; + +/* +* This param id is used to configure Real Time Proxy interface. +*/ +#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213 + +/* This version information is used to handle the new +* additions to the config interface in future in backward +* compatible manner. +*/ +#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1 + +/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG + * command (real-time proxy port configuration parameter). + */ +struct afe_param_id_rt_proxy_port_cfg { + u32 rt_proxy_cfg_minor_version; +/* Minor version used for tracking the version of rt-proxy + * config interface. + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 interleaved; +/* Specifies whether the data exchanged between the AFE + * interface and real-time port is interleaved. + * Supported values: - 0 -- Non-interleaved (samples from each + * channel are contiguous in the buffer) - 1 -- Interleaved + * (corresponding samples from each input channel are interleaved + * within the buffer) + */ + + + u16 frame_size; + /* Size of the frames that are used for PCM exchanges with this + * port. + * Supported values: > 0, in bytes + * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples + * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320 + * bytes. + */ + u16 jitter_allowance; +/* Configures the amount of jitter that the port will allow. + * Supported values: > 0 + * For example, if +/-10 ms of jitter is anticipated in the timing + * of sending frames to the port, and the configuration is 16 kHz + * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2 + * bytes/sample = 320. + */ + + u16 low_water_mark; +/* Low watermark in bytes (including all channels). + * Supported values: + * - 0 -- Do not send any low watermark events + * - > 0 -- Low watermark for triggering an event + * If the number of bytes in an internal circular buffer is lower + * than this low_water_mark parameter, a LOW_WATER_MARK event is + * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS + * event). + * Use of watermark events is optional for debugging purposes. + */ + + u16 high_water_mark; +/* High watermark in bytes (including all channels). + * Supported values: + * - 0 -- Do not send any high watermark events + * - > 0 -- High watermark for triggering an event + * If the number of bytes in an internal circular buffer exceeds + * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event + * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS + * event). + * The use of watermark events is optional and for debugging + * purposes. + */ + + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT + */ + + u16 reserved; + /* For 32 bit alignment. */ +} __packed; + + +/* This param id is used to configure the Pseudoport interface */ + +#define AFE_PARAM_ID_PSEUDO_PORT_CONFIG 0x00010219 + +/* Version information used to handle future additions to the configuration + * interface (for backward compatibility). + */ +#define AFE_API_VERSION_PSEUDO_PORT_CONFIG 0x1 + +/* Enumeration for setting the timing_mode parameter to faster than real + * time. + */ +#define AFE_PSEUDOPORT_TIMING_MODE_FTRT 0x0 + +/* Enumeration for setting the timing_mode parameter to real time using + * timers. + */ +#define AFE_PSEUDOPORT_TIMING_MODE_TIMER 0x1 + +/* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by + AFE_MODULE_AUDIO_DEV_INTERFACE. +*/ +struct afe_param_id_pseudo_port_cfg { + u32 pseud_port_cfg_minor_version; + /* + * Minor version used for tracking the version of the pseudoport + * configuration interface. + */ + + u16 bit_width; + /* Bit width of the sample at values 16, 24 */ + + u16 num_channels; + /* Number of channels at values 1 to 8 */ + + u16 data_format; + /* Non-linear data format supported by the pseudoport (for future use). + * At values #AFE_LINEAR_PCM_DATA + */ + + u16 timing_mode; + /* Indicates whether the pseudoport synchronizes to the clock or + * operates faster than real time. + * at values + * - #AFE_PSEUDOPORT_TIMING_MODE_FTRT + * - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend + */ + + u32 sample_rate; + /* Sample rate at which the pseudoport will run. + * at values + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_32K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend + */ +} __packed; + +#define AFE_PARAM_ID_TDM_CONFIG 0x0001029D + +#define AFE_API_VERSION_TDM_CONFIG 1 + +#define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE 0 +#define AFE_PORT_TDM_LONG_SYNC_MODE 1 +#define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE 2 + +#define AFE_PORT_TDM_SYNC_SRC_EXTERNAL 0 +#define AFE_PORT_TDM_SYNC_SRC_INTERNAL 1 + +#define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE 0 +#define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE 1 + +#define AFE_PORT_TDM_SYNC_NORMAL 0 +#define AFE_PORT_TDM_SYNC_INVERT 1 + +#define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE 0 +#define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE 1 +#define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE 2 + +/* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by + AFE_MODULE_AUDIO_DEV_INTERFACE. +*/ +struct afe_param_id_tdm_cfg { + u32 tdm_cfg_minor_version; + /**< Minor version used to track TDM configuration. + @values #AFE_API_VERSION_TDM_CONFIG */ + + u32 num_channels; + /**< Number of enabled slots for TDM frame. + @values 1 to 8 */ + + u32 sample_rate; + /**< Sampling rate of the port. + @values + - #AFE_PORT_SAMPLE_RATE_8K + - #AFE_PORT_SAMPLE_RATE_16K + - #AFE_PORT_SAMPLE_RATE_24K + - #AFE_PORT_SAMPLE_RATE_32K + - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend */ + + u32 bit_width; + /**< Bit width of the sample. + @values 16, 24 */ + + u16 data_format; + /**< Data format: linear and compressed + + @values + - #AFE_LINEAR_PCM_DATA + - #AFE_NON_LINEAR_DATA @tablebulletend */ + + u16 sync_mode; + /**< TDM synchronization setting. + @values (short, long, slot) sync mode + - #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE + - #AFE_PORT_TDM_LONG_SYNC_MODE + - #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend */ + + u16 sync_src; + /**< Synchronization source. + @values + - #AFE_PORT_TDM_SYNC_SRC_EXTERNAL + - #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend */ + + u16 nslots_per_frame; + /**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32. + @values 1 - 32 */ + + u16 ctrl_data_out_enable; + /**< Specifies whether the TDM block shares the data-out signal to the + drive with other masters. + @values + - #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE + - #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend */ + + u16 ctrl_invert_sync_pulse; + /**< Specifies whether to invert the sync or not. + @values + - #AFE_PORT_TDM_SYNC_NORMAL + - #AFE_PORT_TDM_SYNC_INVERT @tablebulletend */ + + u16 ctrl_sync_data_delay; + /**< Specifies the number of bit clock to delay data with respect to + sync edge. + @values + - #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE + - #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE + - #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend */ + + u16 slot_width; + /**< Slot width of the slot in a TDM frame. (slot_width >= bit_width) + have to be satisfied. + @values 16, 24, 32 */ + + u32 slot_mask; + /**< Position of active slots. When that bit is set, + that paricular slot is active. + Number of active slots can be inferred by number of + bits set in the mask. Only 8 individual bits can be enabled. + Bits 0..31 corresponding to slot 0..31 + @values 1 to 2^32 - 1 */ +} __packed; + +/** ID of Time Divsion Multiplexing (TDM) module, + which is used for configuring the AFE TDM. + + This module supports following parameter IDs: + - #AFE_PORT_TDM_SLOT_CONFIG + + To configure the TDM interface, the client must use the + #AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the + respective parameter IDs as listed above. +*/ + +#define AFE_MODULE_TDM 0x0001028A + +/** ID of the parameter used by #AFE_MODULE_TDM to configure + the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. +*/ +#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 + +/** Version information used to handle future additions to slot mapping + configuration (for backward compatibility). +*/ +#define AFE_API_VERSION_SLOT_MAPPING_CONFIG 0x1 + +/** Data align type */ +#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB 0 +#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB 1 + +#define AFE_SLOT_MAPPING_OFFSET_INVALID 0xFFFF + +/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG + command's TDM configuration parameter. +*/ +struct afe_param_id_slot_mapping_cfg { + u32 minor_version; + /**< Minor version used for tracking TDM slot configuration. + @values #AFE_API_VERSION_TDM_SLOT_CONFIG */ + + u16 num_channel; + /**< number of channel of the audio sample. + @values 1, 2, 4, 6, 8 @tablebulletend */ + + u16 bitwidth; + /**< Slot bit width for each channel + @values 16, 24, 32 */ + + u32 data_align_type; + /**< indicate how data packed from slot_offset for 32 slot bit width + in case of sample bit width is 24. + @values + #AFE_SLOT_MAPPING_DATA_ALIGN_MSB + #AFE_SLOT_MAPPING_DATA_ALIGN_LSB */ + + u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT]; + /**< Array of the slot mapping start offset in bytes for this frame. + The bytes is counted from 0. The 0 is mapped to the 1st byte + in or out of the digital serial data line this sub-frame belong to. + slot_offset[] setting is per-channel based. + The max num of channel supported is 8. + The valid offset value must always be continuly placed in from index 0. + Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays. + If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24, + "data_align_type" is used to indicate how 24 bit sample data in aligning + with 32 bit slot width per-channel. + @values, in byte*/ +} __packed; + +/** ID of the parameter used by #AFE_MODULE_TDM to configure + the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. +*/ +#define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG 0x00010298 + +/** Version information used to handle future additions to custom TDM header + configuration (for backward compatibility). +*/ +#define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG 0x1 + +#define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID 0x0 +#define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT 0x1 +#define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST 0x2 + +#define AFE_CUSTOM_TDM_HEADER_MAX_CNT 0x8 + +/** Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID +*/ +struct afe_param_id_custom_tdm_header_cfg { + u32 minor_version; + /**< Minor version used for tracking custom TDM header configuration. + @values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG */ + + u16 start_offset; + /**< the slot mapping start offset in bytes from this sub-frame + The bytes is counted from 0. The 0 is mapped to the 1st byte in or out of + the digital serial data line this sub-frame belong to. + @values, in byte, + supported values are 0, 4, 8, */ + + u16 header_width; + /**< the header width per-frame followed. + 2 bytes for MOST/TDM case + @values, in byte + supported value is 2 */ + + u16 header_type; + /**< Indicate what kind of custom TDM header it is. + @values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0 + #AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1 (for AAN channel per MOST) + #AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2 + (for entertainment channel, which will overwrite + AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST) */ + + u16 num_frame_repeat; + /**< num of header followed. + @values, supported value is 8*/ + u16 header[AFE_CUSTOM_TDM_HEADER_MAX_CNT]; + /** < SAD header for MOST/TDM case is followed as payload as below. + The size of followed SAD header in bytes is num_of_frame_repeat * header_width_per_frame + which is 2 * 8 = 16 bytes here. + the supported payload format is in uint16_t as below + uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00 + uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218 + uint16_t header2; DTCP Info Info[6] - unused -> 0x0 + uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0 + uint16_t header4; Reserved Info[0] - Num of bytes following -> 0x7 + uint16_t header5; Reserved Info[1] - Media Type -> 0x0 + uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0 + uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low Byte -> 0x0 */ +} __packed; + +struct afe_slot_mapping_config_command { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + struct afe_param_id_slot_mapping_cfg slot_mapping; +} __packed; + +struct afe_custom_tdm_header_config_command { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + struct afe_param_id_custom_tdm_header_cfg custom_tdm_header; +} __packed; + +struct afe_tdm_port_config { + struct afe_param_id_tdm_cfg tdm; + struct afe_param_id_slot_mapping_cfg slot_mapping; + struct afe_param_id_custom_tdm_header_cfg custom_tdm_header; +} __packed; + +#define AFE_PARAM_ID_DEVICE_HW_DELAY 0x00010243 +#define AFE_API_VERSION_DEVICE_HW_DELAY 0x1 + +struct afe_param_id_device_hw_delay_cfg { + uint32_t device_hw_delay_minor_version; + uint32_t delay_in_us; +} __packed; + +#define AFE_PARAM_ID_SET_TOPOLOGY 0x0001025A +#define AFE_API_VERSION_TOPOLOGY_V1 0x1 + +struct afe_param_id_set_topology_cfg { + /* + * Minor version used for tracking afe topology id configuration. + * @values #AFE_API_VERSION_TOPOLOGY_V1 + */ + u32 minor_version; + /* + * Id of the topology for the afe session. + * @values Any valid AFE topology ID + */ + u32 topology_id; +} __packed; + + +/* + * 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 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 COMPRESS_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_MAX_EQ_BANDS 12 + +#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98 + +struct asm_data_cmd_media_fmt_update_v2 { +u32 fmt_blk_size; + /* Media format block size in bytes.*/ +} __packed; + +struct asm_multi_channel_pcm_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + + u16 num_channels; + /* Number of channels. Supported values: 1 to 8 */ + u16 bits_per_sample; +/* Number of bits per sample per channel. * Supported values: + * 16, 24 * When used for playback, the client must send 24-bit + * samples packed in 32-bit words. The 24-bit samples must be placed + * in the most significant 24 bits of the 32-bit word. When used for + * recording, the aDSP sends 24-bit samples packed in 32-bit words. + * The 24-bit samples are placed in the most significant 24 bits of + * the 32-bit word. + */ + + + u32 sample_rate; +/* Number of samples per second (in Hertz). + * Supported values: 2000 to 48000 + */ + + u16 is_signed; + /* Flag that indicates the samples are signed (1). */ + + u16 reserved; + /* reserved field for 32 bit alignment. must be set to zero. */ + + u8 channel_mapping[8]; +/* Channel array of size 8. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_CHANNELS + * - #PCM_CHANNEL_CVH + * - #PCM_CHANNEL_MS + * - #PCM_CHANNEL_FLC + * - #PCM_CHANNEL_FRC + * - #PCM_CHANNEL_RLC + * - #PCM_CHANNEL_RRC + * + * Channel[i] mapping describes channel I. Each element i of the + * array describes channel I inside the buffer where 0 @le I < + * num_channels. An unused channel is set to zero. + */ +} __packed; + +struct asm_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 + +#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 doesnt reset unless the system is reset. + */ + + +u32 timestamp_msw; +/* Upper 32 bits of the 64-bit session time in microseconds of + * the first sample in the buffer. + */ + + +u32 flags; +/* Bitfield of flags. Bit 30 indicates whether frame metadata is + * present. If frame metadata is present, num_frames consecutive + * instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start + * at the buffer address. + * Supported values for bit 31: + * - 1 -- Timestamp is valid. + * - 0 -- Timestamp is invalid. + * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and + * #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit. + * + * Supported values for bit 30: + * - 1 -- Frame metadata is present. + * - 0 -- Frame metadata is absent. + * - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and + * #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit. + * + * All other bits are reserved; the aDSP sets them to 0. + */ + +u32 num_frames; +/* Number of encoded frames in the buffer. */ + +u32 seq_id; +/* Optional buffer sequence ID. */ +} __packed; + +struct asm_data_read_buf_metadata_v2 { + u32 offset; +/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to + * the frame associated with this metadata. + * Supported values: > 0 + */ + +u32 frm_size; +/* Size of the encoded frame in bytes. + * Supported values: > 0 + */ + +u32 num_encoded_pcm_samples; +/* Number of encoded PCM samples (per channel) in the frame + * associated with this metadata. + * Supported values: > 0 + */ + +u32 timestamp_lsw; +/* Lower 32 bits of the 64-bit session time in microseconds of the + * first sample for this frame. + * If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1 + * then the 64 bit timestamp is absolute capture time otherwise it + * is relative session time. The absolute timestamp doesnt reset + * unless the system is reset. + */ + + +u32 timestamp_msw; +/* Lower 32 bits of the 64-bit session time in microseconds of the + * first sample for this frame. + */ + +u32 flags; +/* Frame flags. + * Supported values for bit 31: + * - 1 -- Time stamp is valid + * - 0 -- Time stamp is not valid + * - All other bits are reserved; the aDSP sets them to 0. +*/ +} __packed; + +/* Notifies the client of a change in the data sampling rate or + * Channel mode. This event is raised by the decoder service. The + * event is enabled through the mode flags of + * #ASM_STREAM_CMD_OPEN_WRITE_V2 or + * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change + * in the output sampling frequency or the number/positioning of + * output channels, or if it is the first frame decoded.The new + * sampling frequency or the new channel configuration is + * communicated back to the client asynchronously. + */ + +#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65 + +/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event. + * This event is raised when the following conditions are both true: + * - The event is enabled through the mode_flags of + * #ASM_STREAM_CMD_OPEN_WRITE_V2 or + * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change + * in either the output sampling frequency or the number/positioning + * of output channels, or if it is the first frame decoded. + * This event is not raised (even if enabled) if the decoder is + * MIDI, because + */ + + +struct asm_data_event_sr_cm_change_notify { + u32 sample_rate; +/* New sampling rate (in Hertz) after detecting a change in the + * bitstream. + * Supported values: 2000 to 48000 + */ + + u16 num_channels; +/* New number of channels after detecting a change in the + * bitstream. + * Supported values: 1 to 8 + */ + + + u16 reserved; + /* Reserved for future use. This field must be set to 0.*/ + + u8 channel_mapping[8]; + +} __packed; + +/* Notifies the client of a data sampling rate or channel mode + * change. This event is raised by the encoder service. + * This event is raised when : + * - Native mode encoding was requested in the encoder + * configuration (i.e., the channel number was 0), the sample rate + * was 0, or both were 0. + * + * - The input data frame at the encoder is the first one, or the + * sampling rate/channel mode is different from the previous input + * data frame. + * + */ +#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE + +struct asm_data_event_enc_sr_cm_change_notify { + u32 sample_rate; +/* New sampling rate (in Hertz) after detecting a change in the + * input data. + * Supported values: 2000 to 48000 + */ + + + u16 num_channels; +/* New number of channels after detecting a change in the input + * data. Supported values: 1 to 8 + */ + + + u16 bits_per_sample; +/* New bits per sample after detecting a change in the input + * data. + * Supported values: 16, 24 + */ + + + u8 channel_mapping[8]; + +} __packed; +#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87 + + +/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command, + * which is used to indicate the IEC 60958 frame rate of a given + * packetized audio stream. + */ + +struct asm_data_cmd_iec_60958_frame_rate { + u32 frame_rate; +/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream. + * Supported values: Any valid frame rate + */ +} __packed; + +/* adsp_asm_data_commands.h*/ +/* Definition of the stream ID bitmask.*/ +#define ASM_BIT_MASK_STREAM_ID (0x000000FFUL) + +/* Definition of the stream ID shift value.*/ +#define ASM_SHIFT_STREAM_ID 0 + +/* Definition of the session ID bitmask.*/ +#define ASM_BIT_MASK_SESSION_ID (0x0000FF00UL) + +/* Definition of the session ID shift value.*/ +#define ASM_SHIFT_SESSION_ID 8 + +/* Definition of the service ID bitmask.*/ +#define ASM_BIT_MASK_SERVICE_ID (0x00FF0000UL) + +/* Definition of the service ID shift value.*/ +#define ASM_SHIFT_SERVICE_ID 16 + +/* Definition of the domain ID bitmask.*/ +#define ASM_BIT_MASK_DOMAIN_ID (0xFF000000UL) + +/* Definition of the domain ID shift value.*/ +#define ASM_SHIFT_DOMAIN_ID 24 + +#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92 +#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93 +#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94 + +/* adsp_asm_service_commands.h */ + +#define ASM_MAX_SESSION_ID (15) + +/* Maximum number of sessions.*/ +#define ASM_MAX_NUM_SESSIONS ASM_MAX_SESSION_ID + +/* Maximum number of streams per session.*/ +#define ASM_MAX_STREAMS_PER_SESSION (8) +#define ASM_SESSION_CMD_RUN_V2 0x00010DAA +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE 0 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY 3 + +#define ASM_BIT_MASK_RUN_STARTIME (0x00000003UL) + +/* Bit shift value used to specify the start time for the + * ASM_SESSION_CMD_RUN_V2 command. + */ +#define ASM_SHIFT_RUN_STARTIME 0 +struct asm_session_cmd_run_v2 { + struct apr_hdr hdr; + u32 flags; +/* Specifies whether to run immediately or at a specific + * rendering time or with a specified delay. Run with delay is + * useful for delaying in case of ASM loopback opened through + * ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME + * and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag. + * + * + *Bits 0 and 1 can take one of four possible values: + * + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY + * + *All other bits are reserved; clients must set them to zero. + */ + + u32 time_lsw; +/* Lower 32 bits of the time in microseconds used to align the + * session origin time. When bits 0-1 of flags is + * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of + * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, + * maximum value of the 64 bit delay is 150 ms. + */ + + u32 time_msw; +/* Upper 32 bits of the time in microseconds used to align the + * session origin time. When bits 0-1 of flags is + * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of + * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, + * maximum value of the 64 bit delay is 150 ms. + */ + +} __packed; + +#define ASM_SESSION_CMD_PAUSE 0x00010BD3 +#define ASM_SESSION_CMD_SUSPEND 0x00010DEC +#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D +#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5 + +struct asm_session_cmd_rgstr_rx_underflow { + struct apr_hdr hdr; + u16 enable_flag; +/* Specifies whether a client is to receive events when an Rx + * session underflows. + * Supported values: + * - 0 -- Do not send underflow events + * - 1 -- Send underflow events + */ + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6 + +struct asm_session_cmd_regx_overflow { + struct apr_hdr hdr; + u16 enable_flag; +/* Specifies whether a client is to receive events when a Tx +* session overflows. + * Supported values: + * - 0 -- Do not send overflow events + * - 1 -- Send overflow events + */ + + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17 +#define ASM_SESSION_EVENTX_OVERFLOW 0x00010C18 +#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E + +struct asm_session_cmdrsp_get_sessiontime_v3 { + u32 status; + /* Status message (error code). + * Supported values: Refer to @xhyperref{Q3,[Q3]} + */ + + u32 sessiontime_lsw; + /* Lower 32 bits of the current session time in microseconds.*/ + + u32 sessiontime_msw; + /* Upper 32 bits of the current session time in microseconds.*/ + + u32 absolutetime_lsw; +/* Lower 32 bits in micro seconds of the absolute time at which + * the * sample corresponding to the above session time gets + * rendered * to hardware. This absolute time may be slightly in the + * future or past. + */ + + + u32 absolutetime_msw; +/* Upper 32 bits in micro seconds of the absolute time at which + * the * sample corresponding to the above session time gets + * rendered to * hardware. This absolute time may be slightly in the + * future or past. + */ + +} __packed; + +#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2 0x00010D9F + +struct asm_session_cmd_adjust_session_clock_v2 { + struct apr_hdr hdr; +u32 adjustime_lsw; +/* Lower 32 bits of the signed 64-bit quantity that specifies the + * adjustment time in microseconds to the session clock. + * + * Positive values indicate advancement of the session clock. + * Negative values indicate delay of the session clock. + */ + + + u32 adjustime_msw; +/* Upper 32 bits of the signed 64-bit quantity that specifies + * the adjustment time in microseconds to the session clock. + * Positive values indicate advancement of the session clock. + * Negative values indicate delay of the session clock. + */ + +} __packed; + +#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 0x00010DA0 + +struct asm_session_cmdrsp_adjust_session_clock_v2 { + u32 status; +/* Status message (error code). + * Supported values: Refer to @xhyperref{Q3,[Q3]} + * An error means the session clock is not adjusted. In this case, + * the next two fields are irrelevant. + */ + + + u32 actual_adjustime_lsw; +/* Lower 32 bits of the signed 64-bit quantity that specifies + * the actual adjustment in microseconds performed by the aDSP. + * A positive value indicates advancement of the session clock. A + * negative value indicates delay of the session clock. + */ + + + u32 actual_adjustime_msw; +/* Upper 32 bits of the signed 64-bit quantity that specifies + * the actual adjustment in microseconds performed by the aDSP. + * A positive value indicates advancement of the session clock. A + * negative value indicates delay of the session clock. + */ + + + u32 cmd_latency_lsw; +/* Lower 32 bits of the unsigned 64-bit quantity that specifies + * the amount of time in microseconds taken to perform the session + * clock adjustment. + */ + + + u32 cmd_latency_msw; +/* Upper 32 bits of the unsigned 64-bit quantity that specifies + * the amount of time in microseconds taken to perform the session + * clock adjustment. + */ + +} __packed; + +#define ASM_SESSION_CMD_GET_PATH_DELAY_V2 0x00010DAF +#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0 + +struct asm_session_cmdrsp_get_path_delay_v2 { + u32 status; +/* Status message (error code). Whether this get delay operation + * is successful or not. Delay value is valid only if status is + * success. + * Supported values: Refer to @xhyperref{Q5,[Q5]} + */ + + u32 audio_delay_lsw; + /* Upper 32 bits of the aDSP delay in microseconds. */ + + u32 audio_delay_msw; + /* Lower 32 bits of the aDSP delay in microseconds. */ + +} __packed; + +/* adsp_asm_session_command.h*/ +#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3 + +#define ASM_LOW_LATENCY_STREAM_SESSION 0x10000000 + +#define ASM_ULTRA_LOW_LATENCY_STREAM_SESSION 0x20000000 + +#define ASM_ULL_POST_PROCESSING_STREAM_SESSION 0x40000000 + +#define ASM_LEGACY_STREAM_SESSION 0 + + +struct asm_stream_cmd_open_write_v3 { + struct apr_hdr hdr; + uint32_t mode_flags; +/* Mode flags that configure the stream to notify the client + * whenever it detects an SR/CM change at the input to its POPP. + * Supported values for bits 0 to 1: + * - Reserved; clients must set them to zero. + * Supported values for bit 2: + * - 0 -- SR/CM change notification event is disabled. + * - 1 -- SR/CM change notification event is enabled. + * - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and + * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit. + * + * Supported values for bit 31: + * - 0 -- Stream to be opened in on-Gapless mode. + * - 1 -- Stream to be opened in Gapless mode. In Gapless mode, + * successive streams must be opened with same session ID but + * different stream IDs. + * + * - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and + * #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit. + * + * + * @note1hang MIDI and DTMF streams cannot be opened in Gapless mode. + */ + + uint16_t sink_endpointype; +/*< Sink point type. + * Supported values: + * - 0 -- Device matrix + * - Other values are reserved. + * + * The device matrix is the gateway to the hardware ports. + */ + + uint16_t bits_per_sample; +/*< Number of bits per sample processed by ASM modules. + * Supported values: 16 and 24 bits per sample + */ + + uint32_t postprocopo_id; +/*< Specifies the topology (order of processing) of + * postprocessing algorithms. <i>None</i> means no postprocessing. + * Supported values: + * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT + * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL + * - #ASM_STREAM_POSTPROCOPO_ID_NONE + * + * This field can also be enabled through SetParams flags. + */ + + uint32_t dec_fmt_id; +/*< Configuration ID of the decoder media format. + * + * Supported values: + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_ADPCM + * - #ASM_MEDIA_FMT_MP3 + * - #ASM_MEDIA_FMT_AAC_V2 + * - #ASM_MEDIA_FMT_DOLBY_AAC + * - #ASM_MEDIA_FMT_AMRNB_FS + * - #ASM_MEDIA_FMT_AMRWB_FS + * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2 + * - #ASM_MEDIA_FMT_V13K_FS + * - #ASM_MEDIA_FMT_EVRC_FS + * - #ASM_MEDIA_FMT_EVRCB_FS + * - #ASM_MEDIA_FMT_EVRCWB_FS + * - #ASM_MEDIA_FMT_SBC + * - #ASM_MEDIA_FMT_WMA_V10PRO_V2 + * - #ASM_MEDIA_FMT_WMA_V9_V2 + * - #ASM_MEDIA_FMT_AC3 + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_G711_ALAW_FS + * - #ASM_MEDIA_FMT_G711_MLAW_FS + * - #ASM_MEDIA_FMT_G729A_FS + * - #ASM_MEDIA_FMT_FR_FS + * - #ASM_MEDIA_FMT_VORBIS + * - #ASM_MEDIA_FMT_FLAC + * - #ASM_MEDIA_FMT_ALAC + * - #ASM_MEDIA_FMT_APE + * - #ASM_MEDIA_FMT_EXAMPLE + */ +} __packed; + +#define ASM_STREAM_CMD_OPEN_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_CLOSE 0x00010BCD +#define ASM_STREAM_CMD_FLUSH 0x00010BCE + + +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09 +#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1 + +struct asm_stream_cmd_set_pp_params_v2 { + u32 data_payload_addr_lsw; +/* LSW of parameter data payload address. Supported values: any. */ + u32 data_payload_addr_msw; +/* MSW of Parameter data payload address. Supported values: any. + * - Must be set to zero for in-band data. + * - In the case of 32 bit Shared memory address, msw field must be + * - set to zero. + * - In the case of 36 bit shared memory address, bit 31 to bit 4 of + * msw + * + * - must be set to zero. + */ + u32 mem_map_handle; +/* Supported Values: Any. +* memory map handle returned by DSP through +* ASM_CMD_SHARED_MEM_MAP_REGIONS +* command. +* if mmhandle is NULL, the ParamData payloads are within the +* message payload (in-band). +* If mmhandle is non-NULL, the ParamData payloads begin at the +* address specified in the address msw and lsw (out-of-band). +*/ + + u32 data_payload_size; +/* Size in bytes of the variable payload accompanying the +message, or in shared memory. This field is used for parsing the +parameter payload. */ + +} __packed; + + +struct asm_stream_param_data_v2 { + u32 module_id; + /* Unique module ID. */ + + u32 param_id; + /* Unique parameter ID. */ + + u16 param_size; +/* Data size of the param_id/module_id combination. This is + * a multiple of 4 bytes. + */ + + u16 reserved; +/* Reserved for future enhancements. This field must be set to + * zero. + */ + +} __packed; + +#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2 + +struct asm_stream_cmd_get_pp_params_v2 { + u32 data_payload_addr_lsw; + /* LSW of the parameter data payload address. */ + u32 data_payload_addr_msw; +/* MSW of the parameter data payload address. + * - Size of the shared memory, if specified, shall be large enough + * to contain the whole ParamData payload, including Module ID, + * Param ID, Param Size, and Param Values + * - Must be set to zero for in-band data + * - In the case of 32 bit Shared memory address, msw field must be + * set to zero. + * - In the case of 36 bit shared memory address, bit 31 to bit 4 of + * msw must be set to zero. + */ + + u32 mem_map_handle; +/* Supported Values: Any. +* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS +* command. +* if mmhandle is NULL, the ParamData payloads in the ACK are within the +* message payload (in-band). +* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the +* address specified in the address msw and lsw. +* (out-of-band). +*/ + + u32 module_id; + /* Unique module ID. */ + + u32 param_id; + /* Unique parameter ID. */ + + u16 param_max_size; +/* Maximum data size of the module_id/param_id combination. This + * is a multiple of 4 bytes. + */ + + + u16 reserved; +/* Reserved for backward compatibility. Clients must set this +* field to zero. +*/ + +} __packed; + +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10 + +#define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13 + +struct asm_bitrate_param { + u32 bitrate; +/* Maximum supported bitrate. Only the AAC encoder is supported.*/ + +} __packed; + +#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3 +#define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63 + +/* Flag to turn off both SBR and PS processing, if they are + * present in the bitstream. + */ + +#define ASM_AAC_SBR_OFF_PS_OFF (2) + +/* Flag to turn on SBR but turn off PS processing,if they are + * present in the bitstream. + */ + +#define ASM_AAC_SBR_ON_PS_OFF (1) + +/* Flag to turn on both SBR and PS processing, if they are + * present in the bitstream (default behavior). + */ + + +#define ASM_AAC_SBR_ON_PS_ON (0) + +/* Structure for an AAC SBR PS processing flag. */ + +/* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +struct asm_aac_sbr_ps_flag_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u32 sbr_ps_flag; +/* Control parameter to enable or disable SBR/PS processing in + * the AAC bitstream. Use the following macros to set this field: + * - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS + * processing, if they are present in the bitstream. + * - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS + * processing, if they are present in the bitstream. + * - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing, + * if they are present in the bitstream (default behavior). + * - All other values are invalid. + * Changes are applied to the next decoded frame. + */ +} __packed; + +#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64 + +/* First single channel element in a dual mono bitstream.*/ +#define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1) + +/* Second single channel element in a dual mono bitstream.*/ +#define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2) + +/* Structure for AAC decoder dual mono channel mapping. */ + + +struct asm_aac_dual_mono_mapping_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + u16 left_channel_sce; + u16 right_channel_sce; + +} __packed; + +#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4 + +struct asm_stream_cmdrsp_get_pp_params_v2 { + u32 status; +} __packed; + +#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73 + +/* Enumeration for both vocals in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_NO_VOCAL (0) + +/* Enumeration for only the left vocal in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_LEFT_VOCAL (1) + +/* Enumeration for only the right vocal in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2) + +/* Enumeration for both vocal channels in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_BOTH_VOCAL (3) +#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74 +/* Enumeration for the Custom Analog mode.*/ +#define AC3_DRC_MODE_CUSTOM_ANALOG (0) + +/* Enumeration for the Custom Digital mode.*/ +#define AC3_DRC_MODE_CUSTOM_DIGITAL (1) +/* Enumeration for the Line Out mode (light compression).*/ +#define AC3_DRC_MODE_LINE_OUT (2) + +/* Enumeration for the RF remodulation mode (heavy compression).*/ +#define AC3_DRC_MODE_RF_REMOD (3) +#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75 + +/* Enumeration for playing dual mono in stereo mode.*/ +#define AC3_DUAL_MONO_MODE_STEREO (0) + +/* Enumeration for playing left mono.*/ +#define AC3_DUAL_MONO_MODE_LEFT_MONO (1) + +/* Enumeration for playing right mono.*/ +#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2) + +/* Enumeration for mixing both dual mono channels and playing them.*/ +#define AC3_DUAL_MONO_MODE_MIXED_MONO (3) +#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76 + +/* Enumeration for using the Downmix mode indicated in the bitstream. */ + +#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0) + +/* Enumeration for Surround Compatible mode (preserves the + * surround information). + */ + +#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1) +/* Enumeration for Mono Compatible mode (if the output is to be + * further downmixed to mono). + */ + +#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2) + +/* ID of the AC3 PCM scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78 + +/* ID of the AC3 DRC boost scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79 + +/* ID of the AC3 DRC cut scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A + +/* Structure for AC3 Generic Parameter. */ + +/* Payload of the AC3 parameters in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +struct asm_ac3_generic_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + u32 generic_parameter; +/* AC3 generic parameter. Select from one of the following + * possible values. + * + * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are: + * - AC3_KARAOKE_MODE_NO_VOCAL + * - AC3_KARAOKE_MODE_LEFT_VOCAL + * - AC3_KARAOKE_MODE_RIGHT_VOCAL + * - AC3_KARAOKE_MODE_BOTH_VOCAL + * + * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are: + * - AC3_DRC_MODE_CUSTOM_ANALOG + * - AC3_DRC_MODE_CUSTOM_DIGITAL + * - AC3_DRC_MODE_LINE_OUT + * - AC3_DRC_MODE_RF_REMOD + * + * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are: + * - AC3_DUAL_MONO_MODE_STEREO + * - AC3_DUAL_MONO_MODE_LEFT_MONO + * - AC3_DUAL_MONO_MODE_RIGHT_MONO + * - AC3_DUAL_MONO_MODE_MIXED_MONO + * + * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are: + * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT + * - AC3_STEREO_DOWNMIX_MODE_LT_RT + * - AC3_STEREO_DOWNMIX_MODE_LO_RO + * + * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + * + * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + * + * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + */ +} __packed; + +/* Enumeration for Raw mode (no downmixing), which specifies + * that all channels in the bitstream are to be played out as is + * without any downmixing. (Default) + */ + +#define WMAPRO_CHANNEL_MASK_RAW (-1) + +/* Enumeration for setting the channel mask to 0. The 7.1 mode + * (Home Theater) is assigned. + */ + + +#define WMAPRO_CHANNEL_MASK_ZERO 0x0000 + +/* Speaker layout mask for one channel (Home Theater, mono). + * - Speaker front center + */ +#define WMAPRO_CHANNEL_MASK_1_C 0x0004 + +/* Speaker layout mask for two channels (Home Theater, stereo). + * - Speaker front left + * - Speaker front right + */ +#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003 + +/* Speaker layout mask for three channels (Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + */ +#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007 + +/* Speaker layout mask for two channels (stereo). + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030 + +/* Speaker layout mask for four channels. + * - Speaker front left + * - Speaker front right + * - Speaker back left + * - Speaker back right +*/ +#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033 + +/* Speaker layout mask for four channels (Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back center +*/ +#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107 +/* Speaker layout mask for five channels. + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037 + +/* Speaker layout mask for five channels (5 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607 +/* Speaker layout mask for six channels (5.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F +/* Speaker layout mask for six channels (5.1 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F +/* Speaker layout mask for six channels (5.1 mode, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker back center + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137 +/* Speaker layout mask for six channels (5.1 mode, Home Theater, + * no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back center + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707 + +/* Speaker layout mask for seven channels (6.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back left + * - Speaker back right + * - Speaker back center + */ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F + +/* Speaker layout mask for seven channels (6.1 mode, Home + * Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back center + * - Speaker side left + * - Speaker side right +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F + +/* Speaker layout mask for seven channels (6.1 mode, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker front left of center + * - Speaker front right of center +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7 + +/* Speaker layout mask for seven channels (6.1 mode, Home + * Theater, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + * - Speaker front left of center + * - Speaker front right of center +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637 + +/* Speaker layout mask for eight channels (7.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker low frequency + * - Speaker front left of center + * - Speaker front right of center + */ +#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \ + 0x00FF + +/* Speaker layout mask for eight channels (7.1 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + * - Speaker low frequency + * - Speaker front left of center + * - Speaker front right of center + * +*/ +#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \ + 0x063F + +#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82 + +/* Maximum number of decoder output channels.*/ +#define MAX_CHAN_MAP_CHANNELS 16 + +/* Structure for decoder output channel mapping. */ + +/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +struct asm_dec_out_chan_map_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + u32 num_channels; +/* Number of decoder output channels. + * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS + * + * A value of 0 indicates native channel mapping, which is valid + * only for NT mode. This means the output of the decoder is to be + * preserved as is. + */ + u8 channel_mapping[MAX_CHAN_MAP_CHANNELS]; +} __packed; + +#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84 + +/* Bitmask for the IEC 61937 enable flag.*/ +#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL) + +/* Shift value for the IEC 61937 enable flag.*/ +#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0 + +/* Bitmask for the IEC 60958 enable flag.*/ +#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL) + +/* Shift value for the IEC 60958 enable flag.*/ +#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1 + +/* Payload format for open write compressed comand */ + +/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED + * comand, which opens a stream for a given session ID and stream ID + * to be rendered in the compressed format. + */ + +struct asm_stream_cmd_open_write_compressed { + struct apr_hdr hdr; + u32 flags; +/* Mode flags that configure the stream for a specific format. + * Supported values: + * - Bit 0 -- IEC 61937 compatibility + * - 0 -- Stream is not in IEC 61937 format + * - 1 -- Stream is in IEC 61937 format + * - Bit 1 -- IEC 60958 compatibility + * - 0 -- Stream is not in IEC 60958 format + * - 1 -- Stream is in IEC 60958 format + * - Bits 2 to 31 -- 0 (Reserved) + * + * For the same stream, bit 0 cannot be set to 0 and bit 1 cannot + * be set to 1. A compressed stream connot have IEC 60958 + * packetization applied without IEC 61937 packetization. + * @note1hang Currently, IEC 60958 packetized input streams are not + * supported. + */ + + + u32 fmt_id; +/* Specifies the media type of the HDMI stream to be opened. + * Supported values: + * - #ASM_MEDIA_FMT_AC3 + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_DTS + * - #ASM_MEDIA_FMT_ATRAC + * - #ASM_MEDIA_FMT_MAT + * + * @note1hang This field must be set to a valid media type even if + * IEC 61937 packetization is not performed by the aDSP. + */ + +} __packed; + + +/* + Indicates the number of samples per channel to be removed from the + beginning of the stream. +*/ +#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67 +/* + Indicates the number of samples per channel to be removed from + the end of the stream. +*/ +#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68 +struct asm_data_cmd_remove_silence { + struct apr_hdr hdr; + u32 num_samples_to_remove; + /**< Number of samples per channel to be removed. + + @values 0 to (2@sscr{32}-1) */ +} __packed; + +#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95 + +struct asm_stream_cmd_open_read_compressed { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode flags that indicate whether meta information per encoded + * frame is to be provided. + * Supported values for bit 4: + * - 0 -- Return data buffer contains all encoded frames only; it does + * not contain frame metadata. + * - 1 -- Return data buffer contains an array of metadata and encoded + * frames. + * - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and + * #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit. + * All other bits are reserved; clients must set them to zero. + */ + + u32 frames_per_buf; +/* Indicates the number of frames that need to be returned per + * read buffer + * Supported values: should be greater than 0 + */ + +} __packed; + +/* adsp_asm_stream_commands.h*/ + + +/* adsp_asm_api.h (no changes)*/ +#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \ + 0x00010BE4 +#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \ + 0x00010D83 +#define ASM_STREAM_POSTPROCOPO_ID_NONE \ + 0x00010C68 +#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \ + 0x00010D8B +#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \ + ASM_STREAM_POSTPROCOPO_ID_DEFAULT +#define ASM_STREAM_PREPROCOPO_ID_NONE \ + ASM_STREAM_POSTPROCOPO_ID_NONE +#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \ + 0x00010312 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \ + 0x00010313 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \ + 0x00010314 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\ + 0x00010704 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\ + 0x0001070D +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\ + 0x0001070E +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\ + 0x0001070F +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRC_V3 \ + 0x11000000 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \ + 0x0001031B +#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP 0x00010315 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316 +#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO 0x00010715 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP 0x00010BE3 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317 +#define AUDPROC_MODULE_ID_AIG 0x00010716 +#define AUDPROC_PARAM_ID_AIG_ENABLE 0x00010717 +#define AUDPROC_PARAM_ID_AIG_CONFIG 0x00010718 + +struct Audio_AigParam { + uint16_t mode; +/*< Mode word for enabling AIG/SIG mode . + * Byte offset: 0 + */ + int16_t staticGainL16Q12; +/*< Static input gain when aigMode is set to 1. + * Byte offset: 2 + */ + int16_t initialGainDBL16Q7; +/*<Initial value that the adaptive gain update starts from dB + * Q7 Byte offset: 4 + */ + int16_t idealRMSDBL16Q7; +/*<Average RMS level that AIG attempts to achieve Q8.7 + * Byte offset: 6 + */ + int32_t noiseGateL32; +/*Threshold below which signal is considered as noise and AIG + * Byte offset: 8 + */ + int32_t minGainL32Q15; +/*Minimum gain that can be provided by AIG Q16.15 + * Byte offset: 12 + */ + int32_t maxGainL32Q15; +/*Maximum gain that can be provided by AIG Q16.15 + * Byte offset: 16 + */ + uint32_t gainAtRtUL32Q31; +/*Attack/release time for AIG update Q1.31 + * Byte offset: 20 + */ + uint32_t longGainAtRtUL32Q31; +/*Long attack/release time while updating gain for + * noise/silence Q1.31 Byte offset: 24 + */ + + uint32_t rmsTavUL32Q32; +/* RMS smoothing time constant used for long-term RMS estimate + * Q0.32 Byte offset: 28 + */ + + uint32_t gainUpdateStartTimMsUL32Q0; +/* The waiting time before which AIG starts to apply adaptive + * gain update Q32.0 Byte offset: 32 + */ + +} __packed; + + +#define ADM_MODULE_ID_EANS 0x00010C4A +#define ADM_PARAM_ID_EANS_ENABLE 0x00010C4B +#define ADM_PARAM_ID_EANS_PARAMS 0x00010C4C + +struct adm_eans_enable { + + uint32_t enable_flag; +/*< Specifies whether EANS is disabled (0) or enabled + * (nonzero). + * This is supported only for sampling rates of 8, 12, 16, 24, 32, + * and 48 kHz. It is not supported for sampling rates of 11.025, + * 22.05, or 44.1 kHz. + */ + +} __packed; + + +struct adm_eans_params { + int16_t eans_mode; +/*< Mode word for enabling/disabling submodules. + * Byte offset: 0 + */ + + int16_t eans_input_gain; +/*< Q2.13 input gain to the EANS module. + * Byte offset: 2 + */ + + int16_t eans_output_gain; +/*< Q2.13 output gain to the EANS module. + * Byte offset: 4 + */ + + int16_t eansarget_ns; +/*< Target noise suppression level in dB. + * Byte offset: 6 + */ + + int16_t eans_s_alpha; +/*< Q3.12 over-subtraction factor for stationary noise + * suppression. + * Byte offset: 8 + */ + + int16_t eans_n_alpha; +/* < Q3.12 over-subtraction factor for nonstationary noise + * suppression. + * Byte offset: 10 + */ + + int16_t eans_n_alphamax; +/*< Q3.12 maximum over-subtraction factor for nonstationary + * noise suppression. + * Byte offset: 12 + */ + int16_t eans_e_alpha; +/*< Q15 scaling factor for excess noise suppression. + * Byte offset: 14 + */ + + int16_t eans_ns_snrmax; +/*< Upper boundary in dB for SNR estimation. + * Byte offset: 16 + */ + + int16_t eans_sns_block; +/*< Quarter block size for stationary noise suppression. + * Byte offset: 18 + */ + + int16_t eans_ns_i; +/*< Initialization block size for noise suppression. + * Byte offset: 20 + */ + int16_t eans_np_scale; +/*< Power scale factor for nonstationary noise update. + * Byte offset: 22 + */ + + int16_t eans_n_lambda; +/*< Smoothing factor for higher level nonstationary noise + * update. + * Byte offset: 24 + */ + + int16_t eans_n_lambdaf; +/*< Medium averaging factor for noise update. + * Byte offset: 26 + */ + + int16_t eans_gs_bias; +/*< Bias factor in dB for gain calculation. + * Byte offset: 28 + */ + + int16_t eans_gs_max; +/*< SNR lower boundary in dB for aggressive gain calculation. + * Byte offset: 30 + */ + + int16_t eans_s_alpha_hb; +/*< Q3.12 over-subtraction factor for high-band stationary + * noise suppression. + * Byte offset: 32 + */ + + int16_t eans_n_alphamax_hb; +/*< Q3.12 maximum over-subtraction factor for high-band + * nonstationary noise suppression. + * Byte offset: 34 + */ + + int16_t eans_e_alpha_hb; +/*< Q15 scaling factor for high-band excess noise suppression. + * Byte offset: 36 + */ + + int16_t eans_n_lambda0; +/*< Smoothing factor for nonstationary noise update during + * speech activity. + * Byte offset: 38 + */ + + int16_t thresh; +/*< Threshold for generating a binary VAD decision. + * Byte offset: 40 + */ + + int16_t pwr_scale; +/*< Indirect lower boundary of the noise level estimate. + * Byte offset: 42 + */ + + int16_t hangover_max; +/*< Avoids mid-speech clipping and reliably detects weak speech + * bursts at the end of speech activity. + * Byte offset: 44 + */ + + int16_t alpha_snr; +/*< Controls responsiveness of the VAD. + * Byte offset: 46 + */ + + int16_t snr_diff_max; +/*< Maximum SNR difference. Decreasing this parameter value may + * help in making correct decisions during abrupt changes; however, + * decreasing too much may increase false alarms during long + * pauses/silences. + * Byte offset: 48 + */ + + int16_t snr_diff_min; +/*< Minimum SNR difference. Decreasing this parameter value may + * help in making correct decisions during abrupt changes; however, + * decreasing too much may increase false alarms during long + * pauses/silences. + * Byte offset: 50 + */ + + int16_t init_length; +/*< Defines the number of frames for which a noise level + * estimate is set to a fixed value. + * Byte offset: 52 + */ + + int16_t max_val; +/*< Defines the upper limit of the noise level. + * Byte offset: 54 + */ + + int16_t init_bound; +/*< Defines the initial bounding value for the noise level + * estimate. This is used during the initial segment defined by the + * init_length parameter. + * Byte offset: 56 + */ + + int16_t reset_bound; +/*< Reset boundary for noise tracking. + * Byte offset: 58 + */ + + int16_t avar_scale; +/*< Defines the bias factor in noise estimation. + * Byte offset: 60 + */ + + int16_t sub_nc; +/*< Defines the window length for noise estimation. + * Byte offset: 62 + */ + + int16_t spow_min; +/*< Defines the minimum signal power required to update the + * boundaries for the noise floor estimate. + * Byte offset: 64 + */ + + int16_t eans_gs_fast; +/*< Fast smoothing factor for postprocessor gain. + * Byte offset: 66 + */ + + int16_t eans_gs_med; +/*< Medium smoothing factor for postprocessor gain. + * Byte offset: 68 + */ + + int16_t eans_gs_slow; +/*< Slow smoothing factor for postprocessor gain. + * Byte offset: 70 + */ + + int16_t eans_swb_salpha; +/*< Q3.12 super wideband aggressiveness factor for stationary + * noise suppression. + * Byte offset: 72 + */ + + int16_t eans_swb_nalpha; +/*< Q3.12 super wideband aggressiveness factor for + * nonstationary noise suppression. + * Byte offset: 74 + */ +} __packed; +#define ADM_MODULE_IDX_MIC_GAIN_CTRL 0x00010C35 + +/* @addtogroup audio_pp_param_ids + * ID of the Tx mic gain control parameter used by the + * #ADM_MODULE_IDX_MIC_GAIN_CTRL module. + * @messagepayload + * @structure{admx_mic_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex} + */ +#define ADM_PARAM_IDX_MIC_GAIN 0x00010C36 + +/* Structure for a Tx mic gain parameter for the mic gain + * control module. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the + * Tx Mic Gain Control module. + */ +struct admx_mic_gain { + uint16_t tx_mic_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero. */ +} __packed; + +/* end_addtogroup audio_pp_param_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the Rx Codec Gain Control module. + * + * This module supports the following parameter ID: + * - #ADM_PARAM_ID_RX_CODEC_GAIN + */ +#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL 0x00010C37 + +/* @addtogroup audio_pp_param_ids + * ID of the Rx codec gain control parameter used by the + * #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module. + * + * @messagepayload + * @structure{adm_rx_codec_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex} +*/ +#define ADM_PARAM_ID_RX_CODEC_GAIN 0x00010C38 + +/* Structure for the Rx common codec gain control module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter + * in the Rx Codec Gain Control module. + */ + + +struct adm_rx_codec_gain { + uint16_t rx_codec_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_param_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the HPF Tuning Filter module on the Tx path. + * This module supports the following parameter IDs: + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS + */ +#define ADM_MODULE_ID_HPF_IIRX_FILTER 0x00010C3D + +/* @addtogroup audio_pp_param_ids */ +/* ID of the Tx HPF IIR filter enable parameter used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_enable_cfg} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG 0x00010C3E + +/* ID of the Tx HPF IIR filter pregain parameter used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_pre_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN 0x00010C3F + +/* ID of the Tx HPF IIR filter configuration parameters used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_cfg_params} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA + * RAMS.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS 0x00010C40 + +/* Structure for enabling a configuration parameter for + * the HPF IIR tuning filter module on the Tx path. + */ + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG + * parameter in the Tx path HPF Tuning Filter module. + */ +struct adm_hpfx_iir_filter_enable_cfg { + uint32_t enable_flag; +/*< Specifies whether the HPF tuning filter is disabled (0) or + * enabled (nonzero). + */ +} __packed; + + +/* Structure for the pregain parameter for the HPF + IIR tuning filter module on the Tx path. */ + + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter + * in the Tx path HPF Tuning Filter module. + */ +struct adm_hpfx_iir_filter_pre_gain { + uint16_t pre_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + + +/* Structure for the configuration parameter for the + HPF IIR tuning filter module on the Tx path. */ + + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS + * parameters in the Tx path HPF Tuning Filter module. \n + * \n + * This structure is followed by tuning filter coefficients as follows: \n + * - Sequence of int32_t FilterCoeffs. + * Each band has five coefficients, each in int32_t format in the order of + * b0, b1, b2, a1, a2. + * - Sequence of int16_t NumShiftFactor. + * One int16_t per band. The numerator shift factor is related to the Q + * factor of the filter coefficients. + * - Sequence of uint16_t PanSetting. + * One uint16_t for each band to indicate application of the filter to + * left (0), right (1), or both (2) channels. + */ +struct adm_hpfx_iir_filter_cfg_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @addtogroup audio_pp_module_ids */ +/* ID of the Tx path IIR Tuning Filter module. + * This module supports the following parameter IDs: + * - #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG + */ +#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41 + +/* ID of the Rx path IIR Tuning Filter module for the left channel. + * The parameter IDs of the IIR tuning filter module + * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning + * filter. + * + * Pan parameters are not required for this per-channel IIR filter; the pan + * parameters are ignored by this module. + */ +#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER 0x00010705 + +/* ID of the the Rx path IIR Tuning Filter module for the right + * channel. + * The parameter IDs of the IIR tuning filter module + * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx + * tuning filter. + * + * Pan parameters are not required for this per-channel IIR filter; + * the pan parameters are ignored by this module. + */ +#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER 0x00010706 + +/* end_addtogroup audio_pp_module_ids */ + +/* @addtogroup audio_pp_param_ids */ + +/* ID of the Tx IIR filter enable parameter used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_enable_cfg} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG 0x00010C42 + +/* ID of the Tx IIR filter pregain parameter used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_pre_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN 0x00010C43 + +/* ID of the Tx IIR filter configuration parameters used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_cfg_params} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS 0x00010C44 + +/* Structure for enabling the configuration parameter for the + * IIR filter module on the Tx path. + */ + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG + * parameter in the Tx Path IIR Tuning Filter module. + */ + +struct admx_iir_filter_enable_cfg { + uint32_t enable_flag; +/*< Specifies whether the IIR tuning filter is disabled (0) or + * enabled (nonzero). + */ + +} __packed; + + +/* Structure for the pregain parameter for the + * IIR filter module on the Tx path. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN + * parameter in the Tx Path IIR Tuning Filter module. + */ + +struct admx_iir_filter_pre_gain { + uint16_t pre_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + + +/* Structure for the configuration parameter for the + * IIR filter module on the Tx path. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS + * parameter in the Tx Path IIR Tuning Filter module. \n + * \n + * This structure is followed by the HPF IIR filter coefficients on + * the Tx path as follows: \n + * - Sequence of int32_t ulFilterCoeffs. Each band has five + * coefficients, each in int32_t format in the order of b0, b1, b2, + * a1, a2. + * - Sequence of int16_t sNumShiftFactor. One int16_t per band. The + * numerator shift factor is related to the Q factor of the filter + * coefficients. + * - Sequence of uint16_t usPanSetting. One uint16_t for each band + * to indicate if the filter is applied to left (0), right (1), or + * both (2) channels. + */ +struct admx_iir_filter_cfg_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the QEnsemble module. + * This module supports the following parameter IDs: + * - #ADM_PARAM_ID_QENSEMBLE_ENABLE + * - #ADM_PARAM_ID_QENSEMBLE_BACKGAIN + * - #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE + */ +#define ADM_MODULE_ID_QENSEMBLE 0x00010C59 + +/* @addtogroup audio_pp_param_ids */ +/* ID of the QEnsemble enable parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_enable} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_ENABLE 0x00010C60 + +/* ID of the QEnsemble back gain parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_param_backgain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN 0x00010C61 + +/* ID of the QEnsemble new angle parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_param_set_new_angle} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE 0x00010C62 + +/* Structure for enabling the configuration parameter for the + * QEnsemble module. + */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE + * parameter used by the QEnsemble module. + */ +struct adm_qensemble_enable { + uint32_t enable_flag; +/*< Specifies whether the QEnsemble module is disabled (0) or enabled + * (nonzero). + */ +} __packed; + + +/* Structure for the background gain for the QEnsemble module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN + * parameter used by + * the QEnsemble module. + */ +struct adm_qensemble_param_backgain { + int16_t back_gain; +/*< Linear gain in Q15 format. + * Supported values: 0 to 32767 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; +/* Structure for setting a new angle for the QEnsemble module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE + * parameter used + * by the QEnsemble module. + */ +struct adm_qensemble_param_set_new_angle { + int16_t new_angle; +/*< New angle in degrees. + * Supported values: 0 to 359 + */ + + int16_t time_ms; +/*< Transition time in milliseconds to set the new angle. + * Supported values: 0 to 32767 + */ +} __packed; + + +#define ADM_CMD_GET_PP_TOPO_MODULE_LIST 0x00010349 +#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST 0x00010350 +#define AUDPROC_PARAM_ID_ENABLE 0x00010904 + /* + * Payload of the ADM_CMD_GET_PP_TOPO_MODULE_LIST command. + */ +struct adm_cmd_get_pp_topo_module_list_t { + struct apr_hdr hdr; + /* Lower 32 bits of the 64-bit parameter data payload address. */ + uint32_t data_payload_addr_lsw; + /* + * Upper 32 bits of the 64-bit parameter data payload address. + * + * + * The size of the shared memory, if specified, must be large enough to + * contain the entire parameter data payload, including the module ID, + * parameter ID, parameter size, and parameter values. + */ + uint32_t data_payload_addr_msw; + /* + * Unique identifier for an address. + * + * This memory map handle is returned by the aDSP through the + * #ADM_CMD_SHARED_MEM_MAP_REGIONS command. + * + * @values + * - Non-NULL -- On acknowledgment, the parameter data payloads begin at + * the address specified (out-of-band) + * - NULL -- The acknowledgment's payload contains the parameter data + * (in-band) @tablebulletend + */ + uint32_t mem_map_handle; + /* + * Maximum data size of the list of modules. This + * field is a multiple of 4 bytes. + */ + uint16_t param_max_size; + /* This field must be set to zero. */ + uint16_t reserved; +} __packed; + +/* + * Payload of the ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST message, which returns + * module ids in response to an ADM_CMD_GET_PP_TOPO_MODULE_LIST command. + * Immediately following this structure is the acknowledgement <b>module id + * data variable payload</b> containing the pre/postprocessing module id + * values. For an in-band scenario, the variable payload depends on the size + * of the parameter. + */ +struct adm_cmd_rsp_get_pp_topo_module_list_t { + /* Status message (error code). */ + uint32_t status; +} __packed; + +struct audproc_topology_module_id_info_t { + uint32_t num_modules; +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the Volume Control module pre/postprocessing block. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + * - #ASM_PARAM_ID_MULTICHANNEL_GAIN + * - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG + * - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS + * - #ASM_PARAM_ID_MULTICHANNEL_GAIN + * - #ASM_PARAM_ID_MULTICHANNEL_MUTE + */ +#define ASM_MODULE_ID_VOL_CTRL 0x00010BFE +#define ASM_MODULE_ID_VOL_CTRL2 0x00010910 +#define AUDPROC_MODULE_ID_VOL_CTRL ASM_MODULE_ID_VOL_CTRL + +/* @addtogroup audio_pp_param_ids */ +/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL + * module. + * @messagepayload + * @structure{asm_volume_ctrl_master_gain} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN 0x00010BFF +#define AUDPROC_PARAM_ID_VOL_CTRL_MASTER_GAIN ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + +/* ID of the left/right channel gain parameter used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_volume_ctrl_lr_chan_gain} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MULTICHANNEL_GAIN.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN 0x00010C00 + +/* ID of the mute configuration parameter used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_volume_ctrl_mute_config} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG 0x00010C01 + +/* ID of the soft stepping volume parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_soft_step_volume_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET + * ERS.tex} + */ +#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS 0x00010C29 +#define AUDPROC_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS\ + ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + +/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL + * module. + */ +#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A + +/* ID of the multiple-channel volume control parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + */ +#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713 + +/* ID of the multiple-channel mute configuration parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + */ + +#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714 + +/* Structure for the master gain parameter for a volume control + * module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + * parameter used by the Volume Control module. + */ + + + +struct asm_volume_ctrl_master_gain { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint16_t master_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero. + */ +} __packed; + + +struct asm_volume_ctrl_lr_chan_gain { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + + uint16_t l_chan_gain; + /*< Linear gain in Q13 format for the left channel. */ + + uint16_t r_chan_gain; + /*< Linear gain in Q13 format for the right channel.*/ +} __packed; + + +/* Structure for the mute configuration parameter for a + volume control module. */ + + +/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG + * parameter used by the Volume Control module. + */ + + +struct asm_volume_ctrl_mute_config { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t mute_flag; +/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/ + +} __packed; + +/* + * Supported parameters for a soft stepping linear ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0 + +/* + * Exponential ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1 + +/* + * Logarithmic ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2 + +/* Structure for holding soft stepping volume parameters. */ + + +/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * parameters used by the Volume Control module. + */ +struct asm_soft_step_volume_params { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t period; +/*< Period in milliseconds. + * Supported values: 0 to 15000 + */ + + uint32_t step; +/*< Step in microseconds. + * Supported values: 0 to 15000000 + */ + + uint32_t ramping_curve; +/*< Ramping curve type. + * Supported values: + * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR + * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP + * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG + */ +} __packed; + + +/* Structure for holding soft pause parameters. */ + + +/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS + * parameters used by the Volume Control module. + */ + + +struct asm_soft_pause_params { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t enable_flag; +/*< Specifies whether soft pause is disabled (0) or enabled + * (nonzero). + */ + + + + uint32_t period; +/*< Period in milliseconds. + * Supported values: 0 to 15000 + */ + + uint32_t step; +/*< Step in microseconds. + * Supported values: 0 to 15000000 + */ + + uint32_t ramping_curve; +/*< Ramping curve. + * Supported values: + * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR + * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP + * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG + */ +} __packed; + + +/* Maximum number of channels.*/ +#define VOLUME_CONTROL_MAX_CHANNELS 8 + +/* Structure for holding one channel type - gain pair. */ + + +/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel + * type/gain pairs used by the Volume Control module. \n \n This + * structure immediately follows the + * asm_volume_ctrl_multichannel_gain structure. + */ + + +struct asm_volume_ctrl_channeltype_gain_pair { + uint8_t channeltype; + /* + * Channel type for which the gain setting is to be applied. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_CHANNELS + * - #PCM_CHANNEL_CVH + * - #PCM_CHANNEL_MS + * - #PCM_CHANNEL_FLC + * - #PCM_CHANNEL_FRC + * - #PCM_CHANNEL_RLC + * - #PCM_CHANNEL_RRC + */ + + uint8_t reserved1; + /* Clients must set this field to zero. */ + + uint8_t reserved2; + /* Clients must set this field to zero. */ + + uint8_t reserved3; + /* Clients must set this field to zero. */ + + uint32_t gain; + /* + * Gain value for this channel in Q28 format. + * Supported values: Any + */ +} __packed; + + +/* Structure for the multichannel gain command */ + + +/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN + * parameters used by the Volume Control module. + */ + + +struct asm_volume_ctrl_multichannel_gain { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t num_channels; + /* + * Number of channels for which gain values are provided. Any + * channels present in the data for which gain is not provided are + * set to unity gain. + * Supported values: 1 to 8 + */ + + struct asm_volume_ctrl_channeltype_gain_pair + gain_data[VOLUME_CONTROL_MAX_CHANNELS]; + /* Array of channel type/gain pairs.*/ +} __packed; + + +/* Structure for holding one channel type - mute pair. */ + + +/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel + * type/mute setting pairs used by the Volume Control module. \n \n + * This structure immediately follows the + * asm_volume_ctrl_multichannel_mute structure. + */ + + +struct asm_volume_ctrl_channelype_mute_pair { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint8_t channelype; +/*< Channel type for which the mute setting is to be applied. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_CHANNELS + * - #PCM_CHANNEL_CVH + * - #PCM_CHANNEL_MS + * - #PCM_CHANNEL_FLC + * - #PCM_CHANNEL_FRC + * - #PCM_CHANNEL_RLC + * - #PCM_CHANNEL_RRC + */ + + uint8_t reserved1; + /*< Clients must set this field to zero. */ + + uint8_t reserved2; + /*< Clients must set this field to zero. */ + + uint8_t reserved3; + /*< Clients must set this field to zero. */ + + uint32_t mute; +/*< Mute setting for this channel. + * Supported values: + * - 0 = Unmute + * - Nonzero = Mute + */ +} __packed; + + +/* Structure for the multichannel mute command */ + + +/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE + * parameters used by the Volume Control module. + */ + + +struct asm_volume_ctrl_multichannel_mute { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t num_channels; +/*< Number of channels for which mute configuration is + * provided. Any channels present in the data for which mute + * configuration is not provided are set to unmute. + * Supported values: 1 to 8 + */ + +struct asm_volume_ctrl_channelype_mute_pair + mute_data[VOLUME_CONTROL_MAX_CHANNELS]; + /*< Array of channel type/mute setting pairs.*/ +} __packed; +/* end_addtogroup audio_pp_param_ids */ + +/* audio_pp_module_ids + * ID of the IIR Tuning Filter module. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG + * - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN + * - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS + */ +#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02 + +/* @addtogroup audio_pp_param_ids */ +/* ID of the IIR tuning filter enable parameter used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + * @messagepayload + * @structure{asm_iiruning_filter_enable} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO + * NFIG.tex} + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03 + +/* ID of the IIR tuning filter pregain parameter used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04 + +/* ID of the IIR tuning filter configuration parameters used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05 + +/* Structure for an enable configuration parameter for an + * IIR tuning filter module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG + * parameter used by the IIR Tuning Filter module. + */ +struct asm_iiruning_filter_enable { + uint32_t enable_flag; +/*< Specifies whether the IIR tuning filter is disabled (0) or + * enabled (1). + */ +} __packed; + +/* Structure for the pregain parameter for an IIR tuning filter module. */ + + +/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN + * parameters used by the IIR Tuning Filter module. + */ +struct asm_iiruning_filter_pregain { + uint16_t pregain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* Structure for the configuration parameter for an IIR tuning filter + * module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS + * parameters used by the IIR Tuning Filter module. \n + * \n + * This structure is followed by the IIR filter coefficients: \n + * - Sequence of int32_t FilterCoeffs \n + * Five coefficients for each band. Each coefficient is in int32_t format, in + * the order of b0, b1, b2, a1, a2. + * - Sequence of int16_t NumShiftFactor \n + * One int16_t per band. The numerator shift factor is related to the Q + * factor of the filter coefficients. + * - Sequence of uint16_t PanSetting \n + * One uint16_t per band, indicating if the filter is applied to left (0), + * right (1), or both (2) channels. + */ +struct asm_iir_filter_config_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* audio_pp_module_ids + * ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx + * paths. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_MBDRC_ENABLE + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + */ +#define ASM_MODULE_ID_MBDRC 0x00010C06 + +/* audio_pp_param_ids */ +/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module. + * @messagepayload + * @structure{asm_mbdrc_enable} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex} + */ +#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07 + +/* ID of the MBDRC configuration parameters used by the + * #ASM_MODULE_ID_MBDRC module. + * @messagepayload + * @structure{asm_mbdrc_config_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex} + * + * @parspace Sub-band DRC configuration parameters + * @structure{asm_subband_drc_config_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex} + * + * @keep{6} + * To obtain legacy ADRC from MBDRC, use the calibration tool to: + * + * - Enable MBDRC (EnableFlag = TRUE) + * - Set number of bands to 1 (uiNumBands = 1) + * - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1) + * - Clear the first band mute flag (MuteFlag[0] = 0) + * - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000) + * - Use the legacy ADRC parameters to calibrate the rest of the MBDRC + * parameters. + */ +#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08 + +/* end_addtogroup audio_pp_param_ids */ + +/* audio_pp_module_ids + * ID of the MMBDRC module version 2 pre/postprocessing block. + * This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in + * the length of the filters used in each sub-band. + * This module supports the following parameter ID: + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 + */ +#define ASM_MODULE_ID_MBDRCV2 0x0001070B + +/* @addtogroup audio_pp_param_ids */ +/* ID of the configuration parameters used by the + * #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure + * of the MBDRC v2 pre/postprocessing block. + * The update to this configuration structure from the original + * MBDRC is the number of filter coefficients in the filter + * structure. The sequence for is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t + * padding + * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + + * uint16_t padding + * This block uses the same parameter structure as + * #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS. + */ +#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \ + 0x0001070C + +#define ASM_MODULE_ID_MBDRCV3 0x0001090B +/* + * ID of the MMBDRC module version 3 pre/postprocessing block. + * This module differs from MBDRCv2 (#ASM_MODULE_ID_MBDRCV2) in + * that it supports both 16- and 24-bit data. + * This module supports the following parameter ID: + * - #ASM_PARAM_ID_MBDRC_ENABLE + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_V3 + * - #ASM_PARAM_ID_MBDRC_FILTER_XOVER_FREQS + */ + +/* Structure for the enable parameter for an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the + * MBDRC module. + */ +struct asm_mbdrc_enable { + uint32_t enable_flag; +/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/ +} __packed; + +/* Structure for the configuration parameters for an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + * parameters used by the MBDRC module. \n \n Following this + * structure is the payload for sub-band DRC configuration + * parameters (asm_subband_drc_config_params). This sub-band + * structure must be repeated for each band. + */ + + +struct asm_mbdrc_config_params { + uint16_t num_bands; +/*< Number of bands. + * Supported values: 1 to 5 + */ + + int16_t limiterhreshold; +/*< Threshold in decibels for the limiter output. + * Supported values: -72 to 18 \n + * Recommended value: 3994 (-0.22 db in Q3.12 format) + */ + + int16_t limiter_makeup_gain; +/*< Makeup gain in decibels for the limiter output. + * Supported values: -42 to 42 \n + * Recommended value: 256 (0 dB in Q7.8 format) + */ + + int16_t limiter_gc; +/*< Limiter gain recovery coefficient. + * Supported values: 0.5 to 0.99 \n + * Recommended value: 32440 (0.99 in Q15 format) + */ + + int16_t limiter_delay; +/*< Limiter delay in samples. + * Supported values: 0 to 10 \n + * Recommended value: 262 (0.008 samples in Q15 format) + */ + + int16_t limiter_max_wait; +/*< Maximum limiter waiting time in samples. + * Supported values: 0 to 10 \n + * Recommended value: 262 (0.008 samples in Q15 format) + */ +} __packed; + +/* DRC configuration structure for each sub-band of an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC + * configuration parameters for each sub-band in the MBDRC module. + * After this DRC structure is configured for valid bands, the next + * MBDRC setparams expects the sequence of sub-band MBDRC filter + * coefficients (the length depends on the number of bands) plus the + * mute flag for that band plus uint16_t padding. + * + * @keep{10} + * The filter coefficient and mute flag are of type int16_t: + * - FIR coefficient = int16_t firFilter + * - Mute flag = int16_t fMuteFlag + * + * The sequence is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding + * - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding + * + * For improved filterbank, the sequence is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding + * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding + */ +struct asm_subband_drc_config_params { + int16_t drc_stereo_linked_flag; +/*< Specifies whether all stereo channels have the same applied + * dynamics (1) or if they process their dynamics independently (0). + * Supported values: + * - 0 -- Not linked + * - 1 -- Linked + */ + + int16_t drc_mode; +/*< Specifies whether DRC mode is bypassed for sub-bands. + * Supported values: + * - 0 -- Disabled + * - 1 -- Enabled + */ + + int16_t drc_down_sample_level; +/*< DRC down sample level. + * Supported values: @ge 1 + */ + + int16_t drc_delay; +/*< DRC delay in samples. + * Supported values: 0 to 1200 + */ + + uint16_t drc_rmsime_avg_const; +/*< RMS signal energy time-averaging constant. + * Supported values: 0 to 2^16-1 + */ + + uint16_t drc_makeup_gain; +/*< DRC makeup gain in decibels. + * Supported values: 258 to 64917 + */ + /* Down expander settings */ + int16_t down_expdrhreshold; +/*< Down expander threshold. + * Supported Q7 format values: 1320 to up_cmpsrhreshold + */ + + int16_t down_expdr_slope; +/*< Down expander slope. + * Supported Q8 format values: -32768 to 0. + */ + + uint32_t down_expdr_attack; +/*< Down expander attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t down_expdr_release; +/*< Down expander release constant. + * Supported Q31 format values: 19685 to 2^31 + */ + + uint16_t down_expdr_hysteresis; +/*< Down expander hysteresis constant. + * Supported Q14 format values: 1 to 32690 + */ + + uint16_t reserved; + /*< Clients must set this field to zero. */ + + int32_t down_expdr_min_gain_db; +/*< Down expander minimum gain. + * Supported Q23 format values: -805306368 to 0. + */ + + /* Up compressor settings */ + + int16_t up_cmpsrhreshold; +/*< Up compressor threshold. + * Supported Q7 format values: down_expdrhreshold to + * down_cmpsrhreshold. + */ + + uint16_t up_cmpsr_slope; +/*< Up compressor slope. + * Supported Q16 format values: 0 to 64881. + */ + + uint32_t up_cmpsr_attack; +/*< Up compressor attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t up_cmpsr_release; +/*< Up compressor release constant. + * Supported Q31 format values: 19685 to 2^31. + */ + + uint16_t up_cmpsr_hysteresis; +/*< Up compressor hysteresis constant. + * Supported Q14 format values: 1 to 32690. + */ + + /* Down compressor settings */ + + int16_t down_cmpsrhreshold; +/*< Down compressor threshold. + * Supported Q7 format values: up_cmpsrhreshold to 11560. + */ + + uint16_t down_cmpsr_slope; +/*< Down compressor slope. + * Supported Q16 format values: 0 to 64881. + */ + + uint16_t reserved1; +/*< Clients must set this field to zero. */ + + uint32_t down_cmpsr_attack; +/*< Down compressor attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t down_cmpsr_release; +/*< Down compressor release constant. + * Supported Q31 format values: 19685 to 2^31. + */ + + uint16_t down_cmpsr_hysteresis; +/*< Down compressor hysteresis constant. + * Supported Q14 values: 1 to 32690. + */ + + uint16_t reserved2; +/*< Clients must set this field to zero.*/ +} __packed; + +#define ASM_MODULE_ID_EQUALIZER 0x00010C27 +#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28 + +#define ASM_MAX_EQ_BANDS 12 + +struct asm_eq_per_band_params { + uint32_t band_idx; +/*< Band index. + * Supported values: 0 to 11 + */ + + uint32_t filterype; +/*< Type of filter. + * Supported values: + * - #ASM_PARAM_EQYPE_NONE + * - #ASM_PARAM_EQ_BASS_BOOST + * - #ASM_PARAM_EQ_BASS_CUT + * - #ASM_PARAM_EQREBLE_BOOST + * - #ASM_PARAM_EQREBLE_CUT + * - #ASM_PARAM_EQ_BAND_BOOST + * - #ASM_PARAM_EQ_BAND_CUT + */ + + uint32_t center_freq_hz; + /*< Filter band center frequency in Hertz. */ + + int32_t filter_gain; +/*< Filter band initial gain. + * Supported values: +12 to -12 dB in 1 dB increments + */ + + int32_t q_factor; +/*< Filter band quality factor expressed as a Q8 number, i.e., a + * fixed-point number with q factor of 8. For example, 3000/(2^8). + */ +} __packed; + +struct asm_eq_params { + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 param; + struct asm_stream_param_data_v2 data; + uint32_t enable_flag; +/*< Specifies whether the equalizer module is disabled (0) or enabled + * (nonzero). + */ + + uint32_t num_bands; +/*< Number of bands. + * Supported values: 1 to 12 + */ + struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS]; + +} __packed; + +/* No equalizer effect.*/ +#define ASM_PARAM_EQYPE_NONE 0 + +/* Bass boost equalizer effect.*/ +#define ASM_PARAM_EQ_BASS_BOOST 1 + +/*Bass cut equalizer effect.*/ +#define ASM_PARAM_EQ_BASS_CUT 2 + +/* Treble boost equalizer effect */ +#define ASM_PARAM_EQREBLE_BOOST 3 + +/* Treble cut equalizer effect.*/ +#define ASM_PARAM_EQREBLE_CUT 4 + +/* Band boost equalizer effect.*/ +#define ASM_PARAM_EQ_BAND_BOOST 5 + +/* Band cut equalizer effect.*/ +#define ASM_PARAM_EQ_BAND_CUT 6 + +/* Voice get & set params */ +#define VOICE_CMD_SET_PARAM 0x0001133D +#define VOICE_CMD_GET_PARAM 0x0001133E +#define VOICE_EVT_GET_PARAM_ACK 0x00011008 + + +/** ID of the Bass Boost module. + This module supports the following parameter IDs: + - #AUDPROC_PARAM_ID_BASS_BOOST_ENABLE + - #AUDPROC_PARAM_ID_BASS_BOOST_MODE + - #AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH +*/ +#define AUDPROC_MODULE_ID_BASS_BOOST 0x000108A1 +/** ID of the Bass Boost enable parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_ENABLE 0x000108A2 +/** ID of the Bass Boost mode parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_MODE 0x000108A3 +/** ID of the Bass Boost strength parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH 0x000108A4 + +/** ID of the PBE module. + This module supports the following parameter IDs: + - #AUDPROC_PARAM_ID_PBE_ENABLE + - #AUDPROC_PARAM_ID_PBE_PARAM_CONFIG +*/ +#define AUDPROC_MODULE_ID_PBE 0x00010C2A +/** ID of the Bass Boost enable parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_PBE_ENABLE 0x00010C2B +/** ID of the Bass Boost mode parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_PBE_PARAM_CONFIG 0x00010C49 + +/** ID of the Virtualizer module. This module supports the + following parameter IDs: + - #AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE + - #AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH + - #AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE + - #AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST +*/ +#define AUDPROC_MODULE_ID_VIRTUALIZER 0x000108A5 +/** ID of the Virtualizer enable parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE 0x000108A6 +/** ID of the Virtualizer strength parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH 0x000108A7 +/** ID of the Virtualizer out type parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE 0x000108A8 +/** ID of the Virtualizer out type parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST 0x000108A9 + +/** ID of the Reverb module. This module supports the following + parameter IDs: + - #AUDPROC_PARAM_ID_REVERB_ENABLE + - #AUDPROC_PARAM_ID_REVERB_MODE + - #AUDPROC_PARAM_ID_REVERB_PRESET + - #AUDPROC_PARAM_ID_REVERB_WET_MIX + - #AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST + - #AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL + - #AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL + - #AUDPROC_PARAM_ID_REVERB_DECAY_TIME + - #AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO + - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL + - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY + - #AUDPROC_PARAM_ID_REVERB_LEVEL + - #AUDPROC_PARAM_ID_REVERB_DELAY + - #AUDPROC_PARAM_ID_REVERB_DIFFUSION + - #AUDPROC_PARAM_ID_REVERB_DENSITY +*/ +#define AUDPROC_MODULE_ID_REVERB 0x000108AA +/** ID of the Reverb enable parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ENABLE 0x000108AB +/** ID of the Reverb mode parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_MODE 0x000108AC +/** ID of the Reverb preset parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_PRESET 0x000108AD +/** ID of the Reverb wet mix parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_WET_MIX 0x000108AE +/** ID of the Reverb gain adjust parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST 0x000108AF +/** ID of the Reverb room level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL 0x000108B0 +/** ID of the Reverb room hf level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL 0x000108B1 +/** ID of the Reverb decay time parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DECAY_TIME 0x000108B2 +/** ID of the Reverb decay hf ratio parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO 0x000108B3 +/** ID of the Reverb reflections level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL 0x000108B4 +/** ID of the Reverb reflections delay parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY 0x000108B5 +/** ID of the Reverb level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_LEVEL 0x000108B6 +/** ID of the Reverb delay parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DELAY 0x000108B7 +/** ID of the Reverb diffusion parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DIFFUSION 0x000108B8 +/** ID of the Reverb density parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DENSITY 0x000108B9 + +/** ID of the Popless Equalizer module. This module supports the + following parameter IDs: + - #AUDPROC_PARAM_ID_EQ_ENABLE + - #AUDPROC_PARAM_ID_EQ_CONFIG + - #AUDPROC_PARAM_ID_EQ_NUM_BANDS + - #AUDPROC_PARAM_ID_EQ_BAND_LEVELS + - #AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE + - #AUDPROC_PARAM_ID_EQ_BAND_FREQS + - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE + - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ + - #AUDPROC_PARAM_ID_EQ_BAND_INDEX + - #AUDPROC_PARAM_ID_EQ_PRESET_ID + - #AUDPROC_PARAM_ID_EQ_NUM_PRESETS + - #AUDPROC_PARAM_ID_EQ_GET_PRESET_NAME +*/ +#define AUDPROC_MODULE_ID_POPLESS_EQUALIZER 0x000108BA +/** ID of the Popless Equalizer enable parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_ENABLE 0x000108BB +/** ID of the Popless Equalizer config parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_CONFIG 0x000108BC +/** ID of the Popless Equalizer number of bands parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_NUM_BANDS 0x000108BD +/** ID of the Popless Equalizer band levels parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_LEVELS 0x000108BE +/** ID of the Popless Equalizer band level range parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE 0x000108BF +/** ID of the Popless Equalizer band frequencies parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_FREQS 0x000108C0 +/** ID of the Popless Equalizer single band frequency range + parameter used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. + This param ID is used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE 0x000108C1 +/** ID of the Popless Equalizer single band frequency parameter + used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID + is used for set param only. +*/ +#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ 0x000108C2 +/** ID of the Popless Equalizer band index parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_INDEX 0x000108C3 +/** ID of the Popless Equalizer preset id parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_PRESET_ID 0x000108C4 +/** ID of the Popless Equalizer number of presets parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_NUM_PRESETS 0x000108C5 +/** ID of the Popless Equalizer preset name parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_PRESET_NAME 0x000108C6 + +/* Set Q6 topologies */ +#define ASM_CMD_ADD_TOPOLOGIES 0x00010DBE +#define ADM_CMD_ADD_TOPOLOGIES 0x00010335 +#define AFE_CMD_ADD_TOPOLOGIES 0x000100f8 +/* structure used for both ioctls */ +struct cmd_set_topologies { + struct apr_hdr hdr; + u32 payload_addr_lsw; + /* LSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* MSW of parameter data payload address.*/ + u32 mem_map_handle; + /* Memory map handle returned by mem map command */ + u32 payload_size; + /* Size in bytes of the variable payload in shared memory */ +} __packed; + +/* This module represents the Rx processing of Feedback speaker protection. + * It contains the excursion control, thermal protection, + * analog clip manager features in it. + * This module id will support following param ids. + * - AFE_PARAM_ID_FBSP_MODE_RX_CFG + */ + +#define AFE_MODULE_FB_SPKR_PROT_RX 0x0001021C +#define AFE_MODULE_FB_SPKR_PROT_V2_RX 0x0001025F + +#define AFE_PARAM_ID_FBSP_MODE_RX_CFG 0x0001021D +#define AFE_PARAM_ID_FBSP_PTONE_RAMP_CFG 0x00010260 + +struct asm_fbsp_mode_rx_cfg { + uint32_t minor_version; + uint32_t mode; +} __packed; + +/* This module represents the VI processing of feedback speaker protection. + * It will receive Vsens and Isens from codec and generates necessary + * parameters needed by Rx processing. + * This module id will support following param ids. + * - AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG + * - AFE_PARAM_ID_CALIB_RES_CFG + * - AFE_PARAM_ID_FEEDBACK_PATH_CFG + */ + +#define AFE_MODULE_FB_SPKR_PROT_VI_PROC 0x00010226 +#define AFE_MODULE_FB_SPKR_PROT_VI_PROC_V2 0x0001026A + +#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG 0x0001022A +#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG_V2 0x0001026B + +struct asm_spkr_calib_vi_proc_cfg { + uint32_t minor_version; + uint32_t operation_mode; + uint32_t r0_t0_selection_flag[SP_V2_NUM_MAX_SPKR]; + int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR]; + int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR]; + uint32_t quick_calib_flag; +} __packed; + +#define AFE_PARAM_ID_CALIB_RES_CFG 0x0001022B +#define AFE_PARAM_ID_CALIB_RES_CFG_V2 0x0001026E + +struct asm_calib_res_cfg { + uint32_t minor_version; + int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR]; + uint32_t th_vi_ca_state; +} __packed; + +#define AFE_PARAM_ID_FEEDBACK_PATH_CFG 0x0001022C +#define AFE_MODULE_FEEDBACK 0x00010257 + +struct asm_feedback_path_cfg { + uint32_t minor_version; + int32_t dst_portid; + int32_t num_channels; + int32_t chan_info[4]; +} __packed; + +#define AFE_PARAM_ID_MODE_VI_PROC_CFG 0x00010227 + +struct asm_mode_vi_proc_cfg { + uint32_t minor_version; + uint32_t cal_mode; +} __packed; + +#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; + +/* LSM Specific */ +#define VW_FEAT_DIM (39) + +#define APRV2_IDS_SERVICE_ID_ADSP_LSM_V (0xD) +#define APRV2_IDS_DOMAIN_ID_ADSP_V (0x4) +#define APRV2_IDS_DOMAIN_ID_APPS_V (0x5) + +#define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS (0x00012A7F) +#define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS (0x00012A80) +#define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS (0x00012A81) +#define LSM_SESSION_CMD_OPEN_TX (0x00012A82) +#define LSM_SESSION_CMD_CLOSE_TX (0x00012A88) +#define LSM_SESSION_CMD_SET_PARAMS (0x00012A83) +#define LSM_SESSION_CMD_SET_PARAMS_V2 (0x00012A8F) +#define LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x00012A84) +#define LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x00012A85) +#define LSM_SESSION_CMD_START (0x00012A86) +#define LSM_SESSION_CMD_STOP (0x00012A87) +#define LSM_SESSION_CMD_EOB (0x00012A89) +#define LSM_SESSION_CMD_READ (0x00012A8A) +#define LSM_SESSION_CMD_OPEN_TX_V2 (0x00012A8B) +#define LSM_CMD_ADD_TOPOLOGIES (0x00012A8C) + +#define LSM_SESSION_EVENT_DETECTION_STATUS (0x00012B00) +#define LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x00012B01) +#define LSM_DATA_EVENT_READ_DONE (0x00012B02) +#define LSM_DATA_EVENT_STATUS (0x00012B03) + +#define LSM_MODULE_ID_VOICE_WAKEUP (0x00012C00) +#define LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01) +#define LSM_PARAM_ID_OPERATION_MODE (0x00012C02) +#define LSM_PARAM_ID_GAIN (0x00012C03) +#define LSM_PARAM_ID_CONNECT_TO_PORT (0x00012C04) +#define LSM_PARAM_ID_FEATURE_COMPENSATION_DATA (0x00012C07) +#define LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS (0x00012C07) +#define LSM_MODULE_ID_LAB (0x00012C08) +#define LSM_PARAM_ID_LAB_ENABLE (0x00012C09) +#define LSM_PARAM_ID_LAB_CONFIG (0x00012C0A) +#define LSM_MODULE_ID_FRAMEWORK (0x00012C0E) + +/* HW MAD specific */ +#define AFE_MODULE_HW_MAD (0x00010230) +#define AFE_PARAM_ID_HW_MAD_CFG (0x00010231) +#define AFE_PARAM_ID_HW_MAD_CTRL (0x00010232) +#define AFE_PARAM_ID_SLIMBUS_SLAVE_PORT_CFG (0x00010233) + +/* SW MAD specific */ +#define AFE_MODULE_SW_MAD (0x0001022D) +#define AFE_PARAM_ID_SW_MAD_CFG (0x0001022E) +#define AFE_PARAM_ID_SVM_MODEL (0x0001022F) + +/* Commands/Params to pass the codec/slimbus data to DSP */ +#define AFE_SVC_CMD_SET_PARAM (0x000100f3) +#define AFE_MODULE_CDC_DEV_CFG (0x00010234) +#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG (0x00010235) +#define AFE_PARAM_ID_CDC_REG_CFG (0x00010236) +#define AFE_PARAM_ID_CDC_REG_CFG_INIT (0x00010237) +#define AFE_PARAM_ID_CDC_REG_PAGE_CFG (0x00010296) + +#define AFE_MAX_CDC_REGISTERS_TO_CONFIG (20) + +/* AANC Port Config Specific */ +#define AFE_PARAM_ID_AANC_PORT_CONFIG (0x00010215) +#define AFE_API_VERSION_AANC_PORT_CONFIG (0x1) +#define AANC_TX_MIC_UNUSED (0) +#define AANC_TX_VOICE_MIC (1) +#define AANC_TX_ERROR_MIC (2) +#define AANC_TX_NOISE_MIC (3) +#define AFE_PORT_MAX_CHANNEL_CNT (8) +#define AFE_MODULE_AANC (0x00010214) +#define AFE_PARAM_ID_CDC_AANC_VERSION (0x0001023A) +#define AFE_API_VERSION_CDC_AANC_VERSION (0x1) +#define AANC_HW_BLOCK_VERSION_1 (1) +#define AANC_HW_BLOCK_VERSION_2 (2) + +/*Clip bank selection*/ +#define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1 +#define AFE_CLIP_MAX_BANKS 4 +#define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242 + +struct afe_param_aanc_port_cfg { + /* Minor version used for tracking the version of the module's + * source port configuration. + */ + uint32_t aanc_port_cfg_minor_version; + + /* Sampling rate of the source Tx port. 8k - 192k*/ + uint32_t tx_port_sample_rate; + + /* Channel mapping for the Tx port signal carrying Noise (X), + * Error (E), and Voice (V) signals. + */ + uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT]; + + /* Number of channels on the source Tx port. */ + uint16_t tx_port_num_channels; + + /* Port ID of the Rx path reference signal. */ + uint16_t rx_path_ref_port_id; + + /* Sampling rate of the reference port. 8k - 192k*/ + uint32_t ref_port_sample_rate; +} __packed; + +struct afe_param_id_cdc_aanc_version { + /* Minor version used for tracking the version of the module's + * hw version + */ + uint32_t cdc_aanc_minor_version; + + /* HW version. */ + uint32_t aanc_hw_version; +} __packed; + +struct afe_param_id_clip_bank_sel { + /* Minor version used for tracking the version of the module's + * hw version + */ + uint32_t minor_version; + + /* Number of banks to be read */ + uint32_t num_banks; + + uint32_t bank_map[AFE_CLIP_MAX_BANKS]; +} __packed; + +/* ERROR CODES */ +/* Success. The operation completed with no errors. */ +#define ADSP_EOK 0x00000000 +/* General failure. */ +#define ADSP_EFAILED 0x00000001 +/* Bad operation parameter. */ +#define ADSP_EBADPARAM 0x00000002 +/* Unsupported routine or operation. */ +#define ADSP_EUNSUPPORTED 0x00000003 +/* Unsupported version. */ +#define ADSP_EVERSION 0x00000004 +/* Unexpected problem encountered. */ +#define ADSP_EUNEXPECTED 0x00000005 +/* Unhandled problem occurred. */ +#define ADSP_EPANIC 0x00000006 +/* Unable to allocate resource. */ +#define ADSP_ENORESOURCE 0x00000007 +/* Invalid handle. */ +#define ADSP_EHANDLE 0x00000008 +/* Operation is already processed. */ +#define ADSP_EALREADY 0x00000009 +/* Operation is not ready to be processed. */ +#define ADSP_ENOTREADY 0x0000000A +/* Operation is pending completion. */ +#define ADSP_EPENDING 0x0000000B +/* Operation could not be accepted or processed. */ +#define ADSP_EBUSY 0x0000000C +/* Operation aborted due to an error. */ +#define ADSP_EABORTED 0x0000000D +/* Operation preempted by a higher priority. */ +#define ADSP_EPREEMPTED 0x0000000E +/* Operation requests intervention to complete. */ +#define ADSP_ECONTINUE 0x0000000F +/* Operation requests immediate intervention to complete. */ +#define ADSP_EIMMEDIATE 0x00000010 +/* Operation is not implemented. */ +#define ADSP_ENOTIMPL 0x00000011 +/* Operation needs more data or resources. */ +#define ADSP_ENEEDMORE 0x00000012 +/* Operation does not have memory. */ +#define ADSP_ENOMEMORY 0x00000014 +/* Item does not exist. */ +#define ADSP_ENOTEXIST 0x00000015 +/* Max count for adsp error code sent to HLOS*/ +#define ADSP_ERR_MAX (ADSP_ENOTEXIST + 1) +/* Operation is finished. */ +#define ADSP_ETERMINATED 0x00011174 + +/*bharath, adsp_error_codes.h */ + +/* LPASS clock for I2S Interface */ + +/* Supported OSR clock values */ +#define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ 0xBB8000 +#define Q6AFE_LPASS_OSR_CLK_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; + +/* Command for Matrix or Stream Router */ +#define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2 0x00010DCE +/* Module for AVSYNC */ +#define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC 0x00010DC6 + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the + * render window start value. This parameter is supported only for a Set + * command (not a Get command) in the Rx direction + * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). + * Render window start is a value (session time minus timestamp, or ST-TS) + * below which frames are held, and after which frames are immediately + * rendered. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1 + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the + * render window end value. This parameter is supported only for a Set + * command (not a Get command) in the Rx direction + * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value + * (session time minus timestamp) above which frames are dropped, and below + * which frames are immediately rendered. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 0x00010DD2 + +/* Generic payload of the window parameters in the + * #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module. + * This payload is supported only for a Set command + * (not a Get command) on the Rx path. + */ +struct asm_session_mtmx_strtr_param_window_v2_t { + u32 window_lsw; + /* Lower 32 bits of the render window start value. */ + + u32 window_msw; + /* Upper 32 bits of the render window start value. + + * The 64-bit number formed by window_lsw and window_msw specifies a + * signed 64-bit window value in microseconds. The sign extension is + * necessary. This value is used by the following parameter IDs: + * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2 + * The value depends on which parameter ID is used. + * The aDSP honors the windows at a granularity of 1 ms. + */ +}; + +struct asm_session_cmd_set_mtmx_strstr_params_v2 { + uint32_t data_payload_addr_lsw; + /* Lower 32 bits of the 64-bit data payload address. */ + + uint32_t data_payload_addr_msw; + /* Upper 32 bits of the 64-bit data payload address. + * If the address is not sent (NULL), the message is in the payload. + * If the address is sent (non-NULL), the parameter data payloads + * begin at the specified address. + */ + + uint32_t mem_map_handle; + /* Unique identifier for an address. This memory map handle is returned + * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. + * values + * - NULL -- Parameter data payloads are within the message payload + * (in-band). + * - Non-NULL -- Parameter data payloads begin at the address specified + * in the data_payload_addr_lsw and data_payload_addr_msw fields + * (out-of-band). + */ + + uint32_t data_payload_size; + /* Actual size of the variable payload accompanying the message, or in + * shared memory. This field is used for parsing the parameter payload. + * values > 0 bytes + */ + + uint32_t direction; + /* Direction of the entity (matrix mixer or stream router) on which + * the parameter is to be set. + * values + * - 0 -- Rx (for Rx stream router or Rx matrix mixer) + * - 1 -- Tx (for Tx stream router or Tx matrix mixer) + */ +}; + +struct asm_mtmx_strtr_params { + struct apr_hdr hdr; + struct asm_session_cmd_set_mtmx_strstr_params_v2 param; + struct asm_stream_param_data_v2 data; + u32 window_lsw; + u32 window_msw; +} __packed; + +#define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF +#define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0 + +#define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B +#define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL) + +struct asm_session_cmd_get_mtmx_strstr_params_v2 { + uint32_t data_payload_addr_lsw; + /* Lower 32 bits of the 64-bit data payload address. */ + + uint32_t data_payload_addr_msw; + /* + * Upper 32 bits of the 64-bit data payload address. + * If the address is not sent (NULL), the message is in the payload. + * If the address is sent (non-NULL), the parameter data payloads + * begin at the specified address. + */ + + uint32_t mem_map_handle; + /* + * Unique identifier for an address. This memory map handle is returned + * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. + * values + * - NULL -- Parameter data payloads are within the message payload + * (in-band). + * - Non-NULL -- Parameter data payloads begin at the address specified + * in the data_payload_addr_lsw and data_payload_addr_msw fields + * (out-of-band). + */ + uint32_t direction; + /* + * Direction of the entity (matrix mixer or stream router) on which + * the parameter is to be set. + * values + * - 0 -- Rx (for Rx stream router or Rx matrix mixer) + * - 1 -- Tx (for Tx stream router or Tx matrix mixer) + */ + uint32_t module_id; + /* Unique module ID. */ + + uint32_t param_id; + /* Unique parameter ID. */ + + uint32_t param_max_size; +}; + +struct asm_session_mtmx_strtr_param_session_time_v3_t { + uint32_t session_time_lsw; + /* Lower 32 bits of the current session time in microseconds */ + + uint32_t session_time_msw; + /* + * Upper 32 bits of the current session time in microseconds. + * The 64-bit number formed by session_time_lsw and session_time_msw + * is treated as signed. + */ + + uint32_t absolute_time_lsw; + /* + * Lower 32 bits of the 64-bit absolute time in microseconds. + * This is the time when the sample corresponding to the + * session_time_lsw is rendered to the hardware. This absolute + * time can be slightly in the future or past. + */ + + uint32_t absolute_time_msw; + /* + * Upper 32 bits of the 64-bit absolute time in microseconds. + * This is the time when the sample corresponding to the + * session_time_msw is rendered to hardware. This absolute + * time can be slightly in the future or past. The 64-bit number + * formed by absolute_time_lsw and absolute_time_msw is treated as + * unsigned. + */ + + uint32_t time_stamp_lsw; + /* Lower 32 bits of the last processed timestamp in microseconds */ + + uint32_t time_stamp_msw; + /* + * Upper 32 bits of the last processed timestamp in microseconds. + * The 64-bit number formed by time_stamp_lsw and time_stamp_lsw + * is treated as unsigned. + */ + + uint32_t flags; + /* + * Keeps track of any additional flags needed. + * @values{for bit 31} + * - 0 -- Uninitialized/invalid + * - 1 -- Valid + * All other bits are reserved; clients must set them to zero. + */ +}; + +union asm_session_mtmx_strtr_data_type { + struct asm_session_mtmx_strtr_param_session_time_v3_t session_time; +}; + +struct asm_mtmx_strtr_get_params { + struct apr_hdr hdr; + struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info; +} __packed; + +struct asm_mtmx_strtr_get_params_cmdrsp { + uint32_t err_code; + struct asm_stream_param_data_v2 param_info; + union asm_session_mtmx_strtr_data_type param_data; +} __packed; + +#define AUDPROC_MODULE_ID_RESAMPLER 0x00010719 + +enum { + LEGACY_PCM = 0, + COMPRESSED_PASSTHROUGH, + COMPRESSED_PASSTHROUGH_CONVERT, + COMPRESSED_PASSTHROUGH_DSD, +}; + +#define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770 +#define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771 + +struct adm_set_compressed_device_mute { + struct adm_cmd_set_pp_params_v5 command; + struct adm_param_data_v5 params; + u32 mute_on; +} __packed; + +#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E +#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F + +struct adm_set_compressed_device_latency { + struct adm_cmd_set_pp_params_v5 command; + struct adm_param_data_v5 params; + u32 latency; +} __packed; + +#define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6 +#define VOICEPROC_PARAM_ID_FLUENCE_SOUNDFOCUS 0x00010E37 +#define VOICEPROC_PARAM_ID_FLUENCE_SOURCETRACKING 0x00010E38 +#define MAX_SECTORS 8 +#define MAX_NOISE_SOURCE_INDICATORS 3 +#define MAX_POLAR_ACTIVITY_INDICATORS 360 + +struct sound_focus_param { + uint16_t start_angle[MAX_SECTORS]; + uint8_t enable[MAX_SECTORS]; + uint16_t gain_step; +} __packed; + +struct source_tracking_param { + uint8_t vad[MAX_SECTORS]; + uint16_t doa_speech; + uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS]; + uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS]; +} __packed; + +struct adm_param_fluence_soundfocus_t { + uint16_t start_angles[MAX_SECTORS]; + uint8_t enables[MAX_SECTORS]; + uint16_t gain_step; + uint16_t reserved; +} __packed; + +struct adm_set_fluence_soundfocus_param { + struct adm_cmd_set_pp_params_v5 params; + struct adm_param_data_v5 data; + struct adm_param_fluence_soundfocus_t soundfocus_data; +} __packed; + +struct adm_param_fluence_sourcetracking_t { + uint8_t vad[MAX_SECTORS]; + uint16_t doa_speech; + uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS]; + uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS]; +} __packed; + +#define AUDPROC_MODULE_ID_AUDIOSPHERE 0x00010916 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE 0x00010917 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH 0x00010918 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE 0x00010919 + +#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT 0x0001091A +#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT 0x0001091B +#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT 0x0001091C +#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT 0x0001091D + +#define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO 0x0001091E +#endif /*_APR_AUDIO_V2_H_ */ |