/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 and * only version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. */ #ifndef _APR_AUDIO_V2_H_ #define _APR_AUDIO_V2_H_ #include /* size of header needed for passing data out of band */ #define APR_CMD_OB_HDR_SZ 12 /* size of header needed for getting data */ #define APR_CMD_GET_HDR_SZ 16 struct param_outband { size_t size; void *kvaddr; phys_addr_t paddr; }; #define ADSP_ADM_VERSION 0x00070000 #define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322 #define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323 #define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324 #define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325 #define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D /* Enumeration for an audio Rx matrix ID.*/ #define ADM_MATRIX_ID_AUDIO_RX 0 #define ADM_MATRIX_ID_AUDIO_TX 1 #define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX 2 #define ADM_MATRIX_ID_LISTEN_TX 4 /* Enumeration for an audio Tx matrix ID.*/ #define ADM_MATRIX_ID_AUDIOX 1 #define ADM_MAX_COPPS 5 /* make sure this matches with msm_audio_calibration */ #define SP_V2_NUM_MAX_SPKR 2 /* Session map node structure. * Immediately following this structure are num_copps * entries of COPP IDs. The COPP IDs are 16 bits, so * there might be a padding 16-bit field if num_copps * is odd. */ struct adm_session_map_node_v5 { u16 session_id; /* Handle of the ASM session to be routed. Supported values: 1 * to 8. */ u16 num_copps; /* Number of COPPs to which this session is to be routed. Supported values: 0 < num_copps <= ADM_MAX_COPPS. */ } __packed; /* Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command. * Immediately following this structure are num_sessions of the session map * node payload (adm_session_map_node_v5). */ struct adm_cmd_matrix_map_routings_v5 { struct apr_hdr hdr; u32 matrix_id; /* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx * (1). Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX * macros to set this field. */ u32 num_sessions; /* Number of sessions being updated by this command (optional).*/ } __packed; /* This command allows a client to open a COPP/Voice Proc. TX module * and sets up the device session: Matrix -> COPP -> AFE on the RX * and AFE -> COPP -> Matrix on the TX. This enables PCM data to * be transferred to/from the endpoint (AFEPortID). * * @return * #ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and * COPP ID. */ #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 /* This command allows a client to open a COPP/Voice Proc the * way as ADM_CMD_DEVICE_OPEN_V5 but supports multiple endpoint2 * channels. * * @return * #ADM_CMDRSP_DEVICE_OPEN_V6 with the resulting status and * COPP ID. */ #define ADM_CMD_DEVICE_OPEN_V6 0x00010356 /* Definition for a low latency stream session. */ #define ADM_LOW_LATENCY_DEVICE_SESSION 0x2000 /* Definition for a ultra low latency stream session. */ #define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION 0x4000 /* Definition for a ultra low latency with Post Processing stream session. */ #define ADM_ULL_POST_PROCESSING_DEVICE_SESSION 0x8000 /* Definition for a legacy device session. */ #define ADM_LEGACY_DEVICE_SESSION 0 /* Indicates that endpoint_id_2 is to be ignored.*/ #define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF #define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1 #define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2 #define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3 /* Indicates that an audio COPP is to send/receive a mono PCM * stream to/from * END_POINT_ID_1. */ #define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1 /* Indicates that an audio COPP is to send/receive a * stereo PCM stream to/from END_POINT_ID_1. */ #define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2 /* Sample rate is 8000 Hz.*/ #define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000 /* Sample rate is 16000 Hz.*/ #define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000 /* Sample rate is 48000 Hz.*/ #define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000 /* Definition for a COPP live input flag bitmask.*/ #define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U) /* Definition for a COPP live shift value bitmask.*/ #define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0 /* Definition for the COPP ID bitmask.*/ #define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL) /* Definition for the COPP ID shift value.*/ #define ADM_SHIFT_COPP_ID 0 /* Definition for the service ID bitmask.*/ #define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL) /* Definition for the service ID shift value.*/ #define ADM_SHIFT_SERVICE_ID 16 /* Definition for the domain ID bitmask.*/ #define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL) /* Definition for the domain ID shift value.*/ #define ADM_SHIFT_DOMAIN_ID 24 /* ADM device open command payload of the #ADM_CMD_DEVICE_OPEN_V5 command. */ struct adm_cmd_device_open_v5 { struct apr_hdr hdr; u16 flags; /* Reserved for future use. Clients must set this field * to zero. */ u16 mode_of_operation; /* Specifies whether the COPP must be opened on the Tx or Rx * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for * supported values and interpretation. * Supported values: * - 0x1 -- Rx path COPP * - 0x2 -- Tx path live COPP * - 0x3 -- Tx path nonlive COPP * Live connections cause sample discarding in the Tx device * matrix if the destination output ports do not pull them * fast enough. Nonlive connections queue the samples * indefinitely. */ u16 endpoint_id_1; /* Logical and physical endpoint ID of the audio path. * If the ID is a voice processor Tx block, it receives near * samples. Supported values: Any pseudoport, AFE Rx port, * or AFE Tx port For a list of valid IDs, refer to * @xhyperref{Q4,[Q4]}. * Q4 = Hexagon Multimedia: AFE Interface Specification */ u16 endpoint_id_2; /* Logical and physical endpoint ID 2 for a voice processor * Tx block. * This is not applicable to audio COPP. * Supported values: * - AFE Rx port * - 0xFFFF -- Endpoint 2 is unavailable and the voice * processor Tx * block ignores this endpoint * When the voice processor Tx block is created on the audio * record path, * it can receive far-end samples from an AFE Rx port if the * voice call * is active. The ID of the AFE port is provided in this * field. * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. */ u32 topology_id; /* Audio COPP topology ID; 32-bit GUID. */ u16 dev_num_channel; /* Number of channels the audio COPP sends to/receives from * the endpoint. * Supported values: 1 to 8. * The value is ignored for the voice processor Tx block, * where channel * configuration is derived from the topology ID. */ u16 bit_width; /* Bit width (in bits) that the audio COPP sends to/receives * from the * endpoint. The value is ignored for the voice processing * Tx block, * where the PCM width is 16 bits. */ u32 sample_rate; /* Sampling rate at which the audio COPP/voice processor * Tx block * interfaces with the endpoint. * Supported values for voice processor Tx: 8000, 16000, * 48000 Hz * Supported values for audio COPP: >0 and <=192 kHz */ u8 dev_channel_mapping[8]; /* Array of channel mapping of buffers that the audio COPP * sends to the endpoint. Channel[i] mapping describes channel * I inside the buffer, where 0 < i < dev_num_channel. * This value is relevant only for an audio Rx COPP. * For the voice processor block and Tx audio block, this field * is set to zero and is ignored. */ } __packed; /* ADM device open command payload of the * #ADM_CMD_DEVICE_OPEN_V6 command. */ struct adm_cmd_device_open_v6 { struct apr_hdr hdr; u16 flags; /* Reserved for future use. Clients must set this field * to zero. */ u16 mode_of_operation; /* Specifies whether the COPP must be opened on the Tx or Rx * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for * supported values and interpretation. * Supported values: * - 0x1 -- Rx path COPP * - 0x2 -- Tx path live COPP * - 0x3 -- Tx path nonlive COPP * Live connections cause sample discarding in the Tx device * matrix if the destination output ports do not pull them * fast enough. Nonlive connections queue the samples * indefinitely. */ u16 endpoint_id_1; /* Logical and physical endpoint ID of the audio path. * If the ID is a voice processor Tx block, it receives near * samples. Supported values: Any pseudoport, AFE Rx port, * or AFE Tx port For a list of valid IDs, refer to * @xhyperref{Q4,[Q4]}. * Q4 = Hexagon Multimedia: AFE Interface Specification */ u16 endpoint_id_2; /* Logical and physical endpoint ID 2 for a voice processor * Tx block. * This is not applicable to audio COPP. * Supported values: * - AFE Rx port * - 0xFFFF -- Endpoint 2 is unavailable and the voice * processor Tx * block ignores this endpoint * When the voice processor Tx block is created on the audio * record path, * it can receive far-end samples from an AFE Rx port if the * voice call * is active. The ID of the AFE port is provided in this * field. * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. */ u32 topology_id; /* Audio COPP topology ID; 32-bit GUID. */ u16 dev_num_channel; /* Number of channels the audio COPP sends to/receives from * the endpoint. * Supported values: 1 to 8. * The value is ignored for the voice processor Tx block, * where channel * configuration is derived from the topology ID. */ u16 bit_width; /* Bit width (in bits) that the audio COPP sends to/receives * from the * endpoint. The value is ignored for the voice processing * Tx block, * where the PCM width is 16 bits. */ u32 sample_rate; /* Sampling rate at which the audio COPP/voice processor * Tx block * interfaces with the endpoint. * Supported values for voice processor Tx: 8000, 16000, * 48000 Hz * Supported values for audio COPP: >0 and <=192 kHz */ u8 dev_channel_mapping[8]; /* Array of channel mapping of buffers that the audio COPP * sends to the endpoint. Channel[i] mapping describes channel * I inside the buffer, where 0 < i < dev_num_channel. * This value is relevant only for an audio Rx COPP. * For the voice processor block and Tx audio block, this field * is set to zero and is ignored. */ u16 dev_num_channel_eid2; /* Number of channels the voice processor block sends * to/receives from the endpoint2. * Supported values: 1 to 8. * The value is ignored for audio COPP or if endpoint_id_2 is * set to 0xFFFF. */ u16 bit_width_eid2; /* Bit width (in bits) that the voice processor sends * to/receives from the endpoint2. * Supported values: 16 and 24. * The value is ignored for audio COPP or if endpoint_id_2 is * set to 0xFFFF. */ u32 sample_rate_eid2; /* Sampling rate at which the voice processor Tx block * interfaces with the endpoint2. * Supported values for Tx voice processor: >0 and <=384 kHz * The value is ignored for audio COPP or if endpoint_id_2 is * set to 0xFFFF. */ u8 dev_channel_mapping_eid2[8]; /* Array of channel mapping of buffers that the voice processor * sends to the endpoint. Channel[i] mapping describes channel * I inside the buffer, where 0 < i < dev_num_channel. * This value is relevant only for the Tx voice processor. * The values are ignored for audio COPP or if endpoint_id_2 is * set to 0xFFFF. */ } __packed; /* * This command allows the client to close a COPP and disconnect * the device session. */ #define ADM_CMD_DEVICE_CLOSE_V5 0x00010327 /* Sets one or more parameters to a COPP. */ #define ADM_CMD_SET_PP_PARAMS_V5 0x00010328 /* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command. * If the data_payload_addr_lsw and data_payload_addr_msw element * are NULL, a series of adm_param_datastructures immediately * follows, whose total size is data_payload_size bytes. */ struct adm_cmd_set_pp_params_v5 { struct apr_hdr hdr; u32 payload_addr_lsw; /* LSW of parameter data payload address.*/ u32 payload_addr_msw; /* MSW of parameter data payload address.*/ u32 mem_map_handle; /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS * command */ /* If mem_map_handle is zero implies the message is in * the payload */ u32 payload_size; /* Size in bytes of the variable payload accompanying this * message or * in shared memory. This is used for parsing the parameter * payload. */ } __packed; /* Payload format for COPP parameter data. * Immediately following this structure are param_size bytes * of parameter * data. */ struct adm_param_data_v5 { u32 module_id; /* Unique ID of the module. */ u32 param_id; /* Unique ID of the parameter. */ u16 param_size; /* Data size of the param_id/module_id combination. This value is a multiple of 4 bytes. */ u16 reserved; /* Reserved for future enhancements. * This field must be set to zero. */ } __packed; /* set customized mixing on matrix mixer */ #define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 0x00010344 struct adm_cmd_set_pspd_mtmx_strtr_params_v5 { struct apr_hdr hdr; /* LSW of parameter data payload address.*/ u32 payload_addr_lsw; /* MSW of parameter data payload address.*/ u32 payload_addr_msw; /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */ /* command. If mem_map_handle is zero implies the message is in */ /* the payload */ u32 mem_map_handle; /* Size in bytes of the variable payload accompanying this */ /* message or in shared memory. This is used for parsing the */ /* parameter payload. */ u32 payload_size; u16 direction; u16 sessionid; u16 deviceid; u16 reserved; } __packed; /* Defined specifically for in-band use, includes params */ struct adm_cmd_set_pp_params_inband_v5 { struct apr_hdr hdr; /* LSW of parameter data payload address.*/ u32 payload_addr_lsw; /* MSW of parameter data payload address.*/ u32 payload_addr_msw; /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */ /* command. If mem_map_handle is zero implies the message is in */ /* the payload */ u32 mem_map_handle; /* Size in bytes of the variable payload accompanying this */ /* message or in shared memory. This is used for parsing the */ /* parameter payload. */ u32 payload_size; /* Parameters passed for in band payload */ struct adm_param_data_v5 params; } __packed; /* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. */ #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 /* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message, * which returns the * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. */ struct adm_cmd_rsp_device_open_v5 { u32 status; /* Status message (error code).*/ u16 copp_id; /* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/ u16 reserved; /* Reserved. This field must be set to zero.*/ } __packed; /* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command. */ #define ADM_CMDRSP_DEVICE_OPEN_V6 0x00010357 /* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message, * which returns the * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command * is the exact same as ADM_CMDRSP_DEVICE_OPEN_V5. */ /* This command allows a query of one COPP parameter. */ #define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A /* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command. */ struct adm_cmd_get_pp_params_v5 { struct apr_hdr hdr; u32 data_payload_addr_lsw; /* LSW of parameter data payload address.*/ u32 data_payload_addr_msw; /* MSW of parameter data payload address.*/ /* If the mem_map_handle is non zero, * on ACK, the ParamData payloads begin at * the address specified (out-of-band). */ u32 mem_map_handle; /* Memory map handle returned * by ADM_CMD_SHARED_MEM_MAP_REGIONS command. * If the mem_map_handle is 0, it implies that * the ACK's payload will contain the ParamData (in-band). */ u32 module_id; /* Unique ID of the module. */ u32 param_id; /* Unique ID of the parameter. */ u16 param_max_size; /* Maximum data size of the parameter *ID/module ID combination. This * field is a multiple of 4 bytes. */ u16 reserved; /* Reserved for future enhancements. * This field must be set to zero. */ } __packed; /* Returns parameter values * in response to an #ADM_CMD_GET_PP_PARAMS_V5 command. */ #define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B /* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message, * which returns parameter values in response * to an #ADM_CMD_GET_PP_PARAMS_V5 command. * Immediately following this * structure is the adm_param_data_v5 * structure containing the pre/postprocessing * parameter data. For an in-band * scenario, the variable payload depends * on the size of the parameter. */ struct adm_cmd_rsp_get_pp_params_v5 { u32 status; /* Status message (error code).*/ } __packed; /* Structure for holding soft stepping volume parameters. */ /* * Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS * parameters used by the Volume Control module. */ struct audproc_softvolume_params { u32 period; u32 step; u32 rampingcurve; } __packed; /* * ID of the Media Format Converter (MFC) module. * This module supports the following parameter IDs: * #AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT * #AUDPROC_CHMIXER_PARAM_ID_COEFF */ #define AUDPROC_MODULE_ID_MFC 0x00010912 /* ID of the Output Media Format parameters used by AUDPROC_MODULE_ID_MFC. * */ #define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913 struct audproc_mfc_output_media_fmt { struct adm_cmd_set_pp_params_v5 params; struct adm_param_data_v5 data; uint32_t sampling_rate; uint16_t bits_per_sample; uint16_t num_channels; uint16_t channel_type[8]; } __packed; struct audproc_volume_ctrl_master_gain { struct adm_cmd_set_pp_params_v5 params; struct adm_param_data_v5 data; /* Linear gain in Q13 format. */ uint16_t master_gain; /* Clients must set this field to zero. */ uint16_t reserved; } __packed; struct audproc_soft_step_volume_params { struct adm_cmd_set_pp_params_v5 params; struct adm_param_data_v5 data; /* * Period in milliseconds. * Supported values: 0 to 15000 */ uint32_t period; /* * Step in microseconds. * Supported values: 0 to 15000000 */ uint32_t step; /* * Ramping curve type. * Supported values: * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LINEAR * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_EXP * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LOG */ uint32_t ramping_curve; } __packed; struct audproc_enable_param_t { struct adm_cmd_set_pp_params_inband_v5 pp_params; /* * Specifies whether the Audio processing module is enabled. * This parameter is generic/common parameter to configure or * determine the state of any audio processing module. * @values 0 : Disable 1: Enable */ uint32_t enable; }; /* * Allows a client to control the gains on various session-to-COPP paths. */ #define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C /* Indicates that the target gain in the * current adm_session_copp_gain_v5 * structure is to be applied to all * the session-to-COPP paths that exist for * the specified session. */ #define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF /* Indicates that the target gain is * to be immediately applied to the * specified session-to-COPP path, * without a ramping fashion. */ #define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000 /* Enumeration for a linear ramping curve.*/ #define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000 /* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command. * Immediately following this structure are num_gains of the * adm_session_copp_gain_v5structure. */ struct adm_cmd_matrix_ramp_gains_v5 { u32 matrix_id; /* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX * macros to set this field. */ u16 num_gains; /* Number of gains being applied. */ u16 reserved_for_align; /* Reserved. This field must be set to zero.*/ } __packed; /* Session-to-COPP path gain structure, used by the * #ADM_CMD_MATRIX_RAMP_GAINS_V5 command. * This structure specifies the target * gain (per channel) that must be applied * to a particular session-to-COPP path in * the audio matrix. The structure can * also be used to apply the gain globally * to all session-to-COPP paths that * exist for the given session. * The aDSP uses device channel mapping to * determine which channel gains to * use from this command. For example, * if the device is configured as stereo, * the aDSP uses only target_gain_ch_1 and * target_gain_ch_2, and it ignores * the others. */ struct adm_session_copp_gain_v5 { u16 session_id; /* Handle of the ASM session. * Supported values: 1 to 8. */ u16 copp_id; /* Handle of the COPP. Gain will be applied on the Session ID * COPP ID path. */ u16 ramp_duration; /* Duration (in milliseconds) of the ramp over * which target gains are * to be applied. Use * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE * to indicate that gain must be applied immediately. */ u16 step_duration; /* Duration (in milliseconds) of each step in the ramp. * This parameter is ignored if ramp_duration is equal to * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE. * Supported value: 1 */ u16 ramp_curve; /* Type of ramping curve. * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR */ u16 reserved_for_align; /* Reserved. This field must be set to zero. */ u16 target_gain_ch_1; /* Target linear gain for channel 1 in Q13 format; */ u16 target_gain_ch_2; /* Target linear gain for channel 2 in Q13 format; */ u16 target_gain_ch_3; /* Target linear gain for channel 3 in Q13 format; */ u16 target_gain_ch_4; /* Target linear gain for channel 4 in Q13 format; */ u16 target_gain_ch_5; /* Target linear gain for channel 5 in Q13 format; */ u16 target_gain_ch_6; /* Target linear gain for channel 6 in Q13 format; */ u16 target_gain_ch_7; /* Target linear gain for channel 7 in Q13 format; */ u16 target_gain_ch_8; /* Target linear gain for channel 8 in Q13 format; */ } __packed; /* Allows to set mute/unmute on various session-to-COPP paths. * For every session-to-COPP path (stream-device interconnection), * mute/unmute can be set individually on the output channels. */ #define ADM_CMD_MATRIX_MUTE_V5 0x0001032D /* Indicates that mute/unmute in the * current adm_session_copp_mute_v5structure * is to be applied to all the session-to-COPP * paths that exist for the specified session. */ #define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF /* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/ struct adm_cmd_matrix_mute_v5 { u32 matrix_id; /* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX * macros to set this field. */ u16 session_id; /* Handle of the ASM session. * Supported values: 1 to 8. */ u16 copp_id; /* Handle of the COPP. * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS * to indicate that mute/unmute must be applied to * all the COPPs connected to session_id. * Supported values: * - 0xFFFF -- Apply mute/unmute to all connected COPPs * - Other values -- Valid COPP ID */ u8 mute_flag_ch_1; /* Mute flag for channel 1 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_2; /* Mute flag for channel 2 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_3; /* Mute flag for channel 3 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_4; /* Mute flag for channel 4 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_5; /* Mute flag for channel 5 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_6; /* Mute flag for channel 6 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_7; /* Mute flag for channel 7 is set to unmute (0) or mute (1). */ u8 mute_flag_ch_8; /* Mute flag for channel 8 is set to unmute (0) or mute (1). */ u16 ramp_duration; /* Period (in milliseconds) over which the soft mute/unmute will be * applied. * Supported values: 0 (Default) to 0xFFFF * The default of 0 means mute/unmute will be applied immediately. */ u16 reserved_for_align; /* Clients must set this field to zero.*/ } __packed; #define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8) struct asm_aac_stereo_mix_coeff_selection_param_v2 { struct apr_hdr hdr; u32 param_id; u32 param_size; u32 aac_stereo_mix_coeff_flag; } __packed; /* Allows a client to connect the desired stream to * the desired AFE port through the stream router * * This command allows the client to connect specified session to * specified AFE port. This is used for compressed streams only * opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command. * * @prerequisites * Session ID and AFE Port ID must be valid. * #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED * must have been called on this session. */ #define ADM_CMD_CONNECT_AFE_PORT_V5 0x0001032E #define ADM_CMD_DISCONNECT_AFE_PORT_V5 0x0001032F /* Enumeration for the Rx stream router ID.*/ #define ADM_STRTR_ID_RX 0 /* Enumeration for the Tx stream router ID.*/ #define ADM_STRTR_IDX 1 /* Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/ struct adm_cmd_connect_afe_port_v5 { struct apr_hdr hdr; u8 mode; /* ID of the stream router (RX/TX). Use the * ADM_STRTR_ID_RX or ADM_STRTR_IDX macros * to set this field. */ u8 session_id; /* Session ID of the stream to connect */ u16 afe_port_id; /* Port ID of the AFE port to connect to.*/ u32 num_channels; /* Number of device channels * Supported values: 2(Audio Sample Packet), * 8 (HBR Audio Stream Sample Packet) */ u32 sampling_rate; /* Device sampling rate * Supported values: Any */ } __packed; /* adsp_adm_api.h */ /* Port ID. Update afe_get_port_index * when a new port is added here. */ #define PRIMARY_I2S_RX 0 #define PRIMARY_I2S_TX 1 #define SECONDARY_I2S_RX 4 #define SECONDARY_I2S_TX 5 #define MI2S_RX 6 #define MI2S_TX 7 #define HDMI_RX 8 #define RSVD_2 9 #define RSVD_3 10 #define DIGI_MIC_TX 11 #define VOICE2_PLAYBACK_TX 0x8002 #define VOICE_RECORD_RX 0x8003 #define VOICE_RECORD_TX 0x8004 #define VOICE_PLAYBACK_TX 0x8005 /* Slimbus Multi channel port id pool */ #define SLIMBUS_0_RX 0x4000 #define SLIMBUS_0_TX 0x4001 #define SLIMBUS_1_RX 0x4002 #define SLIMBUS_1_TX 0x4003 #define SLIMBUS_2_RX 0x4004 #define SLIMBUS_2_TX 0x4005 #define SLIMBUS_3_RX 0x4006 #define SLIMBUS_3_TX 0x4007 #define SLIMBUS_4_RX 0x4008 #define SLIMBUS_4_TX 0x4009 #define SLIMBUS_5_RX 0x400a #define SLIMBUS_5_TX 0x400b #define SLIMBUS_6_RX 0x400c #define SLIMBUS_6_TX 0x400d #define SLIMBUS_PORT_LAST SLIMBUS_6_TX #define INT_BT_SCO_RX 0x3000 #define INT_BT_SCO_TX 0x3001 #define INT_BT_A2DP_RX 0x3002 #define INT_FM_RX 0x3004 #define INT_FM_TX 0x3005 #define RT_PROXY_PORT_001_RX 0x2000 #define RT_PROXY_PORT_001_TX 0x2001 #define AFE_LOOPBACK_TX 0x6001 #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® * 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 MI2S_PORT_LAST AFE_PORT_ID_QUATERNARY_MI2S_TX #define AUDIO_PORT_ID_I2S_RX 0x1008 #define AFE_PORT_ID_DIGITAL_MIC_TX 0x1009 #define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A #define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B #define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C #define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D #define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E #define AFE_PORT_ID_SECONDARY_MI2S_RX_SD1 0x1010 #define AFE_PORT_ID_QUINARY_MI2S_RX 0x1016 #define AFE_PORT_ID_QUINARY_MI2S_TX 0x1017 /* ID of the senary MI2S Rx port. */ #define AFE_PORT_ID_SENARY_MI2S_RX 0x1018 /* ID of the senary MI2S Tx port. */ #define AFE_PORT_ID_SENARY_MI2S_TX 0x1019 #define AFE_PORT_ID_SPDIF_RX 0x5000 #define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000 #define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001 #define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000 #define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001 #define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002 #define AFE_PORT_ID_INTERNAL_FM_RX 0x3004 #define AFE_PORT_ID_INTERNAL_FM_TX 0x3005 /* SLIMbus Rx port on channel 0. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000 /* SLIMbus Tx port on channel 0. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001 /* SLIMbus Rx port on channel 1. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002 /* SLIMbus Tx port on channel 1. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003 /* SLIMbus Rx port on channel 2. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004 /* SLIMbus Tx port on channel 2. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005 /* SLIMbus Rx port on channel 3. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006 /* SLIMbus Tx port on channel 3. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007 /* SLIMbus Rx port on channel 4. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008 /* SLIMbus Tx port on channel 4. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009 /* SLIMbus Rx port on channel 5. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX 0x400a /* SLIMbus Tx port on channel 5. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX 0x400b /* SLIMbus Rx port on channel 6. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c /* SLIMbus Tx port on channel 6. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d /* Generic pseudoport 1. */ #define AFE_PORT_ID_PSEUDOPORT_01 0x8001 /* Generic pseudoport 2. */ #define AFE_PORT_ID_PSEUDOPORT_02 0x8002 /* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx} Primary Aux PCM Tx port ID. */ #define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B /* Pseudoport that corresponds to the voice Rx path. * For recording, the voice Rx path samples are written to this * port and consumed by the audio path. */ #define AFE_PORT_ID_VOICE_RECORD_RX 0x8003 /* Pseudoport that corresponds to the voice Tx path. * For recording, the voice Tx path samples are written to this * port and consumed by the audio path. */ #define AFE_PORT_ID_VOICE_RECORD_TX 0x8004 /* Pseudoport that corresponds to in-call voice delivery samples. * During in-call audio delivery, the audio path delivers samples * to this port from where the voice path delivers them on the * Rx path. */ #define AFE_PORT_ID_VOICE2_PLAYBACK_TX 0x8002 #define AFE_PORT_ID_VOICE_PLAYBACK_TX 0x8005 #define AFE_PORT_ID_PRIMARY_TDM_RX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x00) #define AFE_PORT_ID_PRIMARY_TDM_RX_1 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x02) #define AFE_PORT_ID_PRIMARY_TDM_RX_2 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x04) #define AFE_PORT_ID_PRIMARY_TDM_RX_3 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x06) #define AFE_PORT_ID_PRIMARY_TDM_RX_4 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x08) #define AFE_PORT_ID_PRIMARY_TDM_RX_5 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0A) #define AFE_PORT_ID_PRIMARY_TDM_RX_6 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0C) #define AFE_PORT_ID_PRIMARY_TDM_RX_7 \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0E) #define AFE_PORT_ID_PRIMARY_TDM_TX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x01) #define AFE_PORT_ID_PRIMARY_TDM_TX_1 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x02) #define AFE_PORT_ID_PRIMARY_TDM_TX_2 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x04) #define AFE_PORT_ID_PRIMARY_TDM_TX_3 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x06) #define AFE_PORT_ID_PRIMARY_TDM_TX_4 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x08) #define AFE_PORT_ID_PRIMARY_TDM_TX_5 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0A) #define AFE_PORT_ID_PRIMARY_TDM_TX_6 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0C) #define AFE_PORT_ID_PRIMARY_TDM_TX_7 \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0E) #define AFE_PORT_ID_SECONDARY_TDM_RX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x10) #define AFE_PORT_ID_SECONDARY_TDM_RX_1 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x02) #define AFE_PORT_ID_SECONDARY_TDM_RX_2 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x04) #define AFE_PORT_ID_SECONDARY_TDM_RX_3 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x06) #define AFE_PORT_ID_SECONDARY_TDM_RX_4 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x08) #define AFE_PORT_ID_SECONDARY_TDM_RX_5 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0A) #define AFE_PORT_ID_SECONDARY_TDM_RX_6 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0C) #define AFE_PORT_ID_SECONDARY_TDM_RX_7 \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0E) #define AFE_PORT_ID_SECONDARY_TDM_TX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x11) #define AFE_PORT_ID_SECONDARY_TDM_TX_1 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x02) #define AFE_PORT_ID_SECONDARY_TDM_TX_2 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x04) #define AFE_PORT_ID_SECONDARY_TDM_TX_3 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x06) #define AFE_PORT_ID_SECONDARY_TDM_TX_4 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x08) #define AFE_PORT_ID_SECONDARY_TDM_TX_5 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0A) #define AFE_PORT_ID_SECONDARY_TDM_TX_6 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0C) #define AFE_PORT_ID_SECONDARY_TDM_TX_7 \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0E) #define AFE_PORT_ID_TERTIARY_TDM_RX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x20) #define AFE_PORT_ID_TERTIARY_TDM_RX_1 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x02) #define AFE_PORT_ID_TERTIARY_TDM_RX_2 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x04) #define AFE_PORT_ID_TERTIARY_TDM_RX_3 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x06) #define AFE_PORT_ID_TERTIARY_TDM_RX_4 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x08) #define AFE_PORT_ID_TERTIARY_TDM_RX_5 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0A) #define AFE_PORT_ID_TERTIARY_TDM_RX_6 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0C) #define AFE_PORT_ID_TERTIARY_TDM_RX_7 \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0E) #define AFE_PORT_ID_TERTIARY_TDM_TX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x21) #define AFE_PORT_ID_TERTIARY_TDM_TX_1 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x02) #define AFE_PORT_ID_TERTIARY_TDM_TX_2 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x04) #define AFE_PORT_ID_TERTIARY_TDM_TX_3 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x06) #define AFE_PORT_ID_TERTIARY_TDM_TX_4 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x08) #define AFE_PORT_ID_TERTIARY_TDM_TX_5 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0A) #define AFE_PORT_ID_TERTIARY_TDM_TX_6 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0C) #define AFE_PORT_ID_TERTIARY_TDM_TX_7 \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0E) #define AFE_PORT_ID_QUATERNARY_TDM_RX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x30) #define AFE_PORT_ID_QUATERNARY_TDM_RX_1 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x02) #define AFE_PORT_ID_QUATERNARY_TDM_RX_2 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x04) #define AFE_PORT_ID_QUATERNARY_TDM_RX_3 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x06) #define AFE_PORT_ID_QUATERNARY_TDM_RX_4 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x08) #define AFE_PORT_ID_QUATERNARY_TDM_RX_5 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0A) #define AFE_PORT_ID_QUATERNARY_TDM_RX_6 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0C) #define AFE_PORT_ID_QUATERNARY_TDM_RX_7 \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0E) #define AFE_PORT_ID_QUATERNARY_TDM_TX \ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x31) #define AFE_PORT_ID_QUATERNARY_TDM_TX_1 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x02) #define AFE_PORT_ID_QUATERNARY_TDM_TX_2 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x04) #define AFE_PORT_ID_QUATERNARY_TDM_TX_3 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x06) #define AFE_PORT_ID_QUATERNARY_TDM_TX_4 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x08) #define AFE_PORT_ID_QUATERNARY_TDM_TX_5 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0A) #define AFE_PORT_ID_QUATERNARY_TDM_TX_6 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0C) #define AFE_PORT_ID_QUATERNARY_TDM_TX_7 \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0E) #define AFE_PORT_ID_INVALID 0xFFFF #define AAC_ENC_MODE_AAC_LC 0x02 #define AAC_ENC_MODE_AAC_P 0x05 #define AAC_ENC_MODE_EAAC_P 0x1D #define AFE_PSEUDOPORT_CMD_START 0x000100cf struct afe_pseudoport_start_command { struct apr_hdr hdr; u16 port_id; /* Pseudo Port 1 = 0x8000 */ /* Pseudo Port 2 = 0x8001 */ /* Pseudo Port 3 = 0x8002 */ u16 timing; /* FTRT = 0 , AVTimer = 1, */ } __packed; #define AFE_PSEUDOPORT_CMD_STOP 0x000100d0 struct afe_pseudoport_stop_command { struct apr_hdr hdr; u16 port_id; /* Pseudo Port 1 = 0x8000 */ /* Pseudo Port 2 = 0x8001 */ /* Pseudo Port 3 = 0x8002 */ u16 reserved; } __packed; #define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202 #define AFE_PARAM_ID_ENABLE 0x00010203 /* Payload of the #AFE_PARAM_ID_ENABLE * parameter, which enables or * disables any module. * The fixed size of this structure is four bytes. */ struct afe_mod_enable_param { u16 enable; /* Enables (1) or disables (0) the module. */ u16 reserved; /* This field must be set to zero. */ } __packed; /* ID of the configuration parameter used by the * #AFE_MODULE_SIDETONE_IIR_FILTER module. */ #define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204 struct afe_sidetone_iir_filter_config_params { u16 num_biquad_stages; /* Number of stages. * Supported values: Minimum of 5 and maximum of 10 */ u16 pregain; /* Pregain for the compensating filter response. * Supported values: Any number in Q13 format */ } __packed; #define AFE_MODULE_LOOPBACK 0x00010205 #define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206 /* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter, * which gets/sets loopback gain of a port to an Rx port. * The Tx port ID of the loopback is part of the set_param command. */ /* Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's * configuration/calibration settings for the AFE port. */ struct afe_port_cmd_set_param_v2 { u16 port_id; /* Port interface and direction (Rx or Tx) to start. */ u16 payload_size; /* Actual size of the payload in bytes. * This is used for parsing the parameter payload. * Supported values: > 0 */ u32 payload_address_lsw; /* LSW of 64 bit Payload address. * Address should be 32-byte, * 4kbyte aligned and must be contiguous memory. */ u32 payload_address_msw; /* MSW of 64 bit Payload address. * In case of 32-bit shared memory address, * this field must be set to zero. * In case of 36-bit shared memory address, * bit-4 to bit-31 must be set to zero. * Address should be 32-byte, 4kbyte aligned * and must be contiguous memory. */ u32 mem_map_handle; /* Memory map handle returned by * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands. * Supported Values: * - NULL -- Message. The parameter data is in-band. * - Non-NULL -- The parameter data is Out-band.Pointer to * the physical address * in shared memory of the payload data. * An optional field is available if parameter * data is in-band: * afe_param_data_v2 param_data[...]. * For detailed payload content, see the * afe_port_param_data_v2 structure. */ } __packed; #define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF struct afe_port_param_data_v2 { u32 module_id; /* ID of the module to be configured. * Supported values: Valid module ID */ u32 param_id; /* ID of the parameter corresponding to the supported parameters * for the module ID. * Supported values: Valid parameter ID */ u16 param_size; /* Actual size of the data for the * module_id/param_id pair. The size is a * multiple of four bytes. * Supported values: > 0 */ u16 reserved; /* This field must be set to zero. */ } __packed; struct afe_loopback_gain_per_path_param { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; u16 rx_port_id; /* Rx port of the loopback. */ u16 gain; /* Loopback gain per path of the port. * Supported values: Any number in Q13 format */ } __packed; /* Parameter ID used to configure and enable/disable the * loopback path. The difference with respect to the existing * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be * configured as source port in loopback path. Port-id in * AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be * Tx or Rx port. In addition, we can configure the type of * routing mode to handle different use cases. */ #define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B #define AFE_API_VERSION_LOOPBACK_CONFIG 0x1 enum afe_loopback_routing_mode { LB_MODE_DEFAULT = 1, /* Regular loopback from source to destination port */ LB_MODE_SIDETONE, /* Sidetone feed from Tx source to Rx destination port */ LB_MODE_EC_REF_VOICE_AUDIO, /* Echo canceller reference, voice + audio + DTMF */ LB_MODE_EC_REF_VOICE /* Echo canceller reference, voice alone */ } __packed; /* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG , * which enables/disables one AFE loopback. */ struct afe_loopback_cfg_v1 { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; u32 loopback_cfg_minor_version; /* Minor version used for tracking the version of the RMC module * configuration interface. * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG */ u16 dst_port_id; /* Destination Port Id. */ u16 routing_mode; /* Specifies data path type from src to dest port. * Supported values: * #LB_MODE_DEFAULT * #LB_MODE_SIDETONE * #LB_MODE_EC_REF_VOICE_AUDIO * #LB_MODE_EC_REF_VOICE_A * #LB_MODE_EC_REF_VOICE */ u16 enable; /* Specifies whether to enable (1) or * disable (0) an AFE loopback. */ u16 reserved; /* Reserved for 32-bit alignment. This field must be set to 0. */ } __packed; #define AFE_MODULE_SPEAKER_PROTECTION 0x00010209 #define AFE_PARAM_ID_SPKR_PROT_CONFIG 0x0001020a #define AFE_API_VERSION_SPKR_PROT_CONFIG 0x1 #define AFE_SPKR_PROT_EXCURSIONF_LEN 512 struct afe_spkr_prot_cfg_param_v1 { u32 spkr_prot_minor_version; /* * Minor version used for tracking the version of the * speaker protection module configuration interface. * Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG */ int16_t win_size; /* Analysis and synthesis window size (nWinSize). * Supported values: 1024, 512, 256 samples */ int16_t margin; /* Allowable margin for excursion prediction, * in L16Q15 format. This is a * control parameter to allow * for overestimation of peak excursion. */ int16_t spkr_exc_limit; /* Speaker excursion limit, in L16Q15 format.*/ int16_t spkr_resonance_freq; /* Resonance frequency of the speaker; used * to define a frequency range * for signal modification. * * Supported values: 0 to 2000 Hz */ int16_t limhresh; /* Threshold of the hard limiter; used to * prevent overshooting beyond a * signal level that was set by the limiter * prior to speaker protection. * Supported values: 0 to 32767 */ int16_t hpf_cut_off_freq; /* High pass filter cutoff frequency. * Supported values: 100, 200, 300 Hz */ int16_t hpf_enable; /* Specifies whether the high pass filter * is enabled (0) or disabled (1). */ int16_t reserved; /* This field must be set to zero. */ int32_t amp_gain; /* Amplifier gain in L32Q15 format. * This is the RMS voltage at the * loudspeaker when a 0dBFS tone * is played in the digital domain. */ int16_t excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN]; /* Array of the excursion transfer function. * The peak excursion of the * loudspeaker diaphragm is * measured in millimeters for 1 Vrms Sine * tone at all FFT bin frequencies. * Supported values: Q15 format */ } __packed; #define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER 0x000100E0 /* Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER * command, which registers a real-time port driver * with the AFE service. */ struct afe_service_cmd_register_rt_port_driver { struct apr_hdr hdr; u16 port_id; /* Port ID with which the real-time driver exchanges data * (registers for events). * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END */ u16 reserved; /* This field must be set to zero. */ } __packed; #define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER 0x000100E1 /* Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER * command, which unregisters a real-time port driver from * the AFE service. */ struct afe_service_cmd_unregister_rt_port_driver { struct apr_hdr hdr; u16 port_id; /* Port ID from which the real-time * driver unregisters for events. * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END */ u16 reserved; /* This field must be set to zero. */ } __packed; #define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105 #define AFE_EVENTYPE_RT_PROXY_PORT_START 0 #define AFE_EVENTYPE_RT_PROXY_PORT_STOP 1 #define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK 2 #define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK 3 #define AFE_EVENTYPE_RT_PROXY_PORT_INVALID 0xFFFF /* Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS * message, which sends an event from the AFE service * to a registered client. */ struct afe_event_rt_proxy_port_status { u16 port_id; /* Port ID to which the event is sent. * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END */ u16 eventype; /* Type of event. * Supported values: * - #AFE_EVENTYPE_RT_PROXY_PORT_START * - #AFE_EVENTYPE_RT_PROXY_PORT_STOP * - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK * - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK */ } __packed; #define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED struct afe_port_data_cmd_rt_proxy_port_write_v2 { struct apr_hdr hdr; u16 port_id; /* Tx (mic) proxy port ID with which the real-time * driver exchanges data. * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END */ u16 reserved; /* This field must be set to zero. */ u32 buffer_address_lsw; /* LSW Address of the buffer containing the * data from the real-time source * device on a client. */ u32 buffer_address_msw; /* MSW Address of the buffer containing the * data from the real-time source * device on a client. */ u32 mem_map_handle; /* A memory map handle encapsulating shared memory * attributes is returned if * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS * command is successful. * Supported Values: * - Any 32 bit value */ u32 available_bytes; /* Number of valid bytes available * in the buffer (including all * channels: number of bytes per * channel = availableBytesumChannels). * Supported values: > 0 * * This field must be equal to the frame * size specified in the #AFE_PORT_AUDIO_IF_CONFIG * command that was sent to configure this * port. */ } __packed; #define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 0x000100EE /* Payload of the * #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which * delivers an empty buffer to the AFE service. On * acknowledgment, data is filled in the buffer. */ struct afe_port_data_cmd_rt_proxy_port_read_v2 { struct apr_hdr hdr; u16 port_id; /* Rx proxy port ID with which the real-time * driver exchanges data. * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END * (This must be an Rx (speaker) port.) */ u16 reserved; /* This field must be set to zero. */ u32 buffer_address_lsw; /* LSW Address of the buffer containing the data sent from the AFE * service to a real-time sink device on the client. */ u32 buffer_address_msw; /* MSW Address of the buffer containing the data sent from the AFE * service to a real-time sink device on the client. */ u32 mem_map_handle; /* A memory map handle encapsulating shared memory attributes is * returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is * successful. * Supported Values: * - Any 32 bit value */ u32 available_bytes; /* Number of valid bytes available in the buffer (including all * channels). * Supported values: > 0 * This field must be equal to the frame size specified in the * #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure * this port. */ } __packed; /* This module ID is related to device configuring like I2S,PCM, * HDMI, SLIMBus etc. This module supports follwing parameter ids. * - #AFE_PARAM_ID_I2S_CONFIG * - #AFE_PARAM_ID_PCM_CONFIG * - #AFE_PARAM_ID_DIGI_MIC_CONFIG * - #AFE_PARAM_ID_HDMI_CONFIG * - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG * - #AFE_PARAM_ID_SLIMBUS_CONFIG * - #AFE_PARAM_ID_RT_PROXY_CONFIG */ #define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C #define AFE_PORT_SAMPLE_RATE_8K 8000 #define AFE_PORT_SAMPLE_RATE_16K 16000 #define AFE_PORT_SAMPLE_RATE_48K 48000 #define AFE_PORT_SAMPLE_RATE_96K 96000 #define AFE_PORT_SAMPLE_RATE_192K 192000 #define AFE_LINEAR_PCM_DATA 0x0 #define AFE_NON_LINEAR_DATA 0x1 #define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2 #define AFE_NON_LINEAR_DATA_PACKED_60958 0x3 /* This param id is used to configure I2S interface */ #define AFE_PARAM_ID_I2S_CONFIG 0x0001020D #define AFE_API_VERSION_I2S_CONFIG 0x1 /* Enumeration for setting the I2S configuration * channel_mode parameter to * serial data wire number 1-3 (SD3). */ #define AFE_PORT_I2S_SD0 0x1 #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 #define AFE_PORT_I2S_8CHS 0x8 #define AFE_PORT_I2S_MONO 0x0 #define AFE_PORT_I2S_STEREO 0x1 #define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0 #define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 /* Payload of the #AFE_PARAM_ID_I2S_CONFIG * command's (I2S configuration * parameter). */ struct afe_param_id_i2s_cfg { u32 i2s_cfg_minor_version; /* Minor version used for tracking the version of the I2S * configuration interface. * Supported values: #AFE_API_VERSION_I2S_CONFIG */ u16 bit_width; /* Bit width of the sample. * Supported values: 16, 24 */ u16 channel_mode; /* I2S lines and multichannel operation. * Supported values: * - #AFE_PORT_I2S_SD0 * - #AFE_PORT_I2S_SD1 * - #AFE_PORT_I2S_SD2 * - #AFE_PORT_I2S_SD3 * - #AFE_PORT_I2S_QUAD01 * - #AFE_PORT_I2S_QUAD23 * - #AFE_PORT_I2S_6CHS * - #AFE_PORT_I2S_8CHS */ u16 mono_stereo; /* Specifies mono or stereo. This applies only when * a single I2S line is used. * Supported values: * - #AFE_PORT_I2S_MONO * - #AFE_PORT_I2S_STEREO */ u16 ws_src; /* Word select source: internal or external. * Supported values: * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K * - #AFE_PORT_SAMPLE_RATE_48K * - #AFE_PORT_SAMPLE_RATE_96K * - #AFE_PORT_SAMPLE_RATE_192K */ u16 data_format; /* data format * Supported values: * - #LINEAR_PCM_DATA * - #NON_LINEAR_DATA * - #LINEAR_PCM_DATA_PACKED_IN_60958 * - #NON_LINEAR_DATA_PACKED_IN_60958 */ u16 reserved; /* This field must be set to zero. */ } __packed; /* * This param id is used to configure PCM interface */ #define AFE_API_VERSION_SPDIF_CONFIG 0x1 #define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1 #define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1 #define AFE_CH_STATUS_A 1 #define AFE_CH_STATUS_B 2 #define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244 #define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245 #define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246 #define AFE_PORT_CLK_ROOT_LPAPLL 0x3 #define AFE_PORT_CLK_ROOT_LPAQ6PLL 0x4 struct afe_param_id_spdif_cfg { /* Minor version used for tracking the version of the SPDIF * configuration interface. * Supported values: #AFE_API_VERSION_SPDIF_CONFIG */ u32 spdif_cfg_minor_version; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_22_05K * - #AFE_PORT_SAMPLE_RATE_32K * - #AFE_PORT_SAMPLE_RATE_44_1K * - #AFE_PORT_SAMPLE_RATE_48K * - #AFE_PORT_SAMPLE_RATE_96K * - #AFE_PORT_SAMPLE_RATE_176_4K * - #AFE_PORT_SAMPLE_RATE_192K */ u32 sample_rate; /* data format * Supported values: * - #AFE_LINEAR_PCM_DATA * - #AFE_NON_LINEAR_DATA */ u16 data_format; /* Number of channels supported by the port * - PCM - 1, Compressed Case - 2 */ u16 num_channels; /* Bit width of the sample. * Supported values: 16, 24 */ u16 bit_width; /* This field must be set to zero. */ u16 reserved; } __packed; struct afe_param_id_spdif_ch_status_cfg { u32 ch_status_cfg_minor_version; /* Minor version used for tracking the version of channel * status configuration. Current supported version is 1 */ u32 status_type; /* Indicate if the channel status is for channel A or B * Supported values: * - #AFE_CH_STATUS_A * - #AFE_CH_STATUS_B */ u8 status_bits[24]; /* Channel status - 192 bits for channel * Byte ordering as defined by IEC60958-3 */ u8 status_mask[24]; /* Channel status with mask bits 1 will be applied. * Byte ordering as defined by IEC60958-3 */ } __packed; struct afe_param_id_spdif_clk_cfg { u32 clk_cfg_minor_version; /* Minor version used for tracking the version of SPDIF * interface clock configuration. Current supported version * is 1 */ u32 clk_value; /* Specifies the clock frequency in Hz to set * Supported values: * 0 - Disable the clock * 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2 * (channels A and B) */ u32 clk_root; /* Specifies SPDIF root clk source * Supported Values: * - #AFE_PORT_CLK_ROOT_LPAPLL * - #AFE_PORT_CLK_ROOT_LPAQ6PLL */ } __packed; struct afe_spdif_clk_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_id_spdif_clk_cfg clk_cfg; } __packed; struct afe_spdif_chstatus_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_id_spdif_ch_status_cfg ch_status; } __packed; struct afe_spdif_port_config { struct afe_param_id_spdif_cfg cfg; struct afe_param_id_spdif_ch_status_cfg ch_status; } __packed; #define AFE_PARAM_ID_PCM_CONFIG 0x0001020E #define AFE_API_VERSION_PCM_CONFIG 0x1 /* Enumeration for the auxiliary PCM synchronization signal * provided by an external source. */ #define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 /* Enumeration for the auxiliary PCM synchronization signal * provided by an internal source. */ #define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1 /* Enumeration for the PCM configuration aux_mode parameter, * which configures the auxiliary PCM interface to use * short synchronization. */ #define AFE_PORT_PCM_AUX_MODE_PCM 0x0 /* * Enumeration for the PCM configuration aux_mode parameter, * which configures the auxiliary PCM interface to use long * synchronization. */ #define AFE_PORT_PCM_AUX_MODE_AUX 0x1 /* * Enumeration for setting the PCM configuration frame to 8. */ #define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0 /* * Enumeration for setting the PCM configuration frame to 16. */ #define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1 /* Enumeration for setting the PCM configuration frame to 32.*/ #define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2 /* Enumeration for setting the PCM configuration frame to 64.*/ #define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3 /* Enumeration for setting the PCM configuration frame to 128.*/ #define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4 /* Enumeration for setting the PCM configuration frame to 256.*/ #define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5 /* Enumeration for setting the PCM configuration * quantype parameter to A-law with no padding. */ #define AFE_PORT_PCM_ALAW_NOPADDING 0x0 /* Enumeration for setting the PCM configuration quantype * parameter to mu-law with no padding. */ #define AFE_PORT_PCM_MULAW_NOPADDING 0x1 /* Enumeration for setting the PCM configuration quantype * parameter to linear with no padding. */ #define AFE_PORT_PCM_LINEAR_NOPADDING 0x2 /* Enumeration for setting the PCM configuration quantype * parameter to A-law with padding. */ #define AFE_PORT_PCM_ALAW_PADDING 0x3 /* Enumeration for setting the PCM configuration quantype * parameter to mu-law with padding. */ #define AFE_PORT_PCM_MULAW_PADDING 0x4 /* Enumeration for setting the PCM configuration quantype * parameter to linear with padding. */ #define AFE_PORT_PCM_LINEAR_PADDING 0x5 /* Enumeration for disabling the PCM configuration * ctrl_data_out_enable parameter. * The PCM block is the only master. */ #define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0 /* * Enumeration for enabling the PCM configuration * ctrl_data_out_enable parameter. The PCM block shares * the signal with other masters. */ #define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1 /* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's * (PCM configuration parameter). */ struct afe_param_id_pcm_cfg { u32 pcm_cfg_minor_version; /* Minor version used for tracking the version of the AUX PCM * configuration interface. * Supported values: #AFE_API_VERSION_PCM_CONFIG */ u16 aux_mode; /* PCM synchronization setting. * Supported values: * - #AFE_PORT_PCM_AUX_MODE_PCM * - #AFE_PORT_PCM_AUX_MODE_AUX */ u16 sync_src; /* Synchronization source. * Supported values: * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL */ u16 frame_setting; /* Number of bits per frame. * Supported values: * - #AFE_PORT_PCM_BITS_PER_FRAME_8 * - #AFE_PORT_PCM_BITS_PER_FRAME_16 * - #AFE_PORT_PCM_BITS_PER_FRAME_32 * - #AFE_PORT_PCM_BITS_PER_FRAME_64 * - #AFE_PORT_PCM_BITS_PER_FRAME_128 * - #AFE_PORT_PCM_BITS_PER_FRAME_256 */ u16 quantype; /* PCM quantization type. * Supported values: * - #AFE_PORT_PCM_ALAW_NOPADDING * - #AFE_PORT_PCM_MULAW_NOPADDING * - #AFE_PORT_PCM_LINEAR_NOPADDING * - #AFE_PORT_PCM_ALAW_PADDING * - #AFE_PORT_PCM_MULAW_PADDING * - #AFE_PORT_PCM_LINEAR_PADDING */ u16 ctrl_data_out_enable; /* Specifies whether the PCM block shares the data-out * signal to the drive with other masters. * Supported values: * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE */ u16 reserved; /* This field must be set to zero. */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K */ u16 bit_width; /* Bit width of the sample. * Supported values: 16 */ u16 num_channels; /* Number of channels. * Supported values: 1 to 4 */ u16 slot_number_mapping[4]; /* Specifies the slot number for the each channel in * multi channel scenario. * Supported values: 1 to 32 */ } __packed; /* * This param id is used to configure DIGI MIC interface */ #define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F /* This version information is used to handle the new * additions to the config interface in future in backward * compatible manner. */ #define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1 /* Enumeration for setting the digital mic configuration * channel_mode parameter to left 0. */ #define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1 /*Enumeration for setting the digital mic configuration * channel_mode parameter to right 0. */ #define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2 /* Enumeration for setting the digital mic configuration * channel_mode parameter to left 1. */ #define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3 /* Enumeration for setting the digital mic configuration * channel_mode parameter to right 1. */ #define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4 /* Enumeration for setting the digital mic configuration * channel_mode parameter to stereo 0. */ #define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5 /* Enumeration for setting the digital mic configuration * channel_mode parameter to stereo 1. */ #define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6 /* Enumeration for setting the digital mic configuration * channel_mode parameter to quad. */ #define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7 /* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's * (DIGI MIC configuration * parameter). */ struct afe_param_id_digi_mic_cfg { u32 digi_mic_cfg_minor_version; /* Minor version used for tracking the version of the DIGI Mic * configuration interface. * Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG */ u16 bit_width; /* Bit width of the sample. * Supported values: 16 */ u16 channel_mode; /* Digital mic and multichannel operation. * Supported values: * - #AFE_PORT_DIGI_MIC_MODE_LEFT0 * - #AFE_PORT_DIGI_MIC_MODE_RIGHT0 * - #AFE_PORT_DIGI_MIC_MODE_LEFT1 * - #AFE_PORT_DIGI_MIC_MODE_RIGHT1 * - #AFE_PORT_DIGI_MIC_MODE_STEREO0 * - #AFE_PORT_DIGI_MIC_MODE_STEREO1 * - #AFE_PORT_DIGI_MIC_MODE_QUAD */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K * - #AFE_PORT_SAMPLE_RATE_48K */ } __packed; /* * This param id is used to configure HDMI interface */ #define AFE_PARAM_ID_HDMI_CONFIG 0x00010210 /* This version information is used to handle the new * additions to the config interface in future in backward * compatible manner. */ #define AFE_API_VERSION_HDMI_CONFIG 0x1 /* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command, * which configures a multichannel HDMI audio interface. */ struct afe_param_id_hdmi_multi_chan_audio_cfg { u32 hdmi_cfg_minor_version; /* Minor version used for tracking the version of the HDMI * configuration interface. * Supported values: #AFE_API_VERSION_HDMI_CONFIG */ u16 datatype; /* data type * Supported values: * - #LINEAR_PCM_DATA * - #NON_LINEAR_DATA * - #LINEAR_PCM_DATA_PACKED_IN_60958 * - #NON_LINEAR_DATA_PACKED_IN_60958 */ u16 channel_allocation; /* HDMI channel allocation information for programming an HDMI * frame. The default is 0 (Stereo). * * This information is defined in the HDMI standard, CEA 861-D * (refer to @xhyperref{S1,[S1]}). The number of channels is also * inferred from this parameter. */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K * - #AFE_PORT_SAMPLE_RATE_48K * - #AFE_PORT_SAMPLE_RATE_96K * - 22050, 44100, 176400 for compressed streams */ u16 bit_width; /* Bit width of the sample. * Supported values: 16, 24 */ u16 reserved; /* This field must be set to zero. */ } __packed; /* * This param id is used to configure BT or FM(RIVA) interface */ #define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211 /* This version information is used to handle the new * additions to the config interface in future in backward * compatible manner. */ #define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1 /* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG * command's BT voice/BT audio/FM configuration parameter. */ struct afe_param_id_internal_bt_fm_cfg { u32 bt_fm_cfg_minor_version; /* Minor version used for tracking the version of the BT and FM * configuration interface. * Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG */ u16 num_channels; /* Number of channels. * Supported values: 1 to 2 */ u16 bit_width; /* Bit width of the sample. * Supported values: 16 */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO) * - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO) * - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP) */ } __packed; /* This param id is used to configure SLIMBUS interface using * shared channel approach. */ #define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 /* This version information is used to handle the new * additions to the config interface in future in backward * compatible manner. */ #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 /* Enumeration for setting SLIMbus device ID 1. */ #define AFE_SLIMBUS_DEVICE_1 0x0 /* Enumeration for setting SLIMbus device ID 2. */ #define AFE_SLIMBUS_DEVICE_2 0x1 /* Enumeration for setting the SLIMbus data formats. */ #define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0 /* Enumeration for setting the maximum number of streams per * device. */ #define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8 /* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus * port configuration parameter. */ struct afe_param_id_slimbus_cfg { u32 sb_cfg_minor_version; /* Minor version used for tracking the version of the SLIMBUS * configuration interface. * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG */ u16 slimbus_dev_id; /* SLIMbus hardware device ID, which is required to handle * multiple SLIMbus hardware blocks. * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2 */ u16 bit_width; /* Bit width of the sample. * Supported values: 16, 24 */ u16 data_format; /* Data format supported by the SLIMbus hardware. The default is * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the * hardware does not perform any format conversions before the data * transfer. */ u16 num_channels; /* Number of channels. * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT */ u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; /* Mapping of shared channel IDs (128 to 255) to which the * master port is to be connected. * Shared_channel_mapping[i] represents the shared channel assigned * for audio channel i in multichannel audio data. */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K * - #AFE_PORT_SAMPLE_RATE_48K * - #AFE_PORT_SAMPLE_RATE_96K * - #AFE_PORT_SAMPLE_RATE_192K */ } __packed; /* * This param id is used to configure Real Time Proxy interface. */ #define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213 /* This version information is used to handle the new * additions to the config interface in future in backward * compatible manner. */ #define AFE_API_VERSION_RT_PROXY_CONFIG 0x1 /* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG * command (real-time proxy port configuration parameter). */ struct afe_param_id_rt_proxy_port_cfg { u32 rt_proxy_cfg_minor_version; /* Minor version used for tracking the version of rt-proxy * config interface. */ u16 bit_width; /* Bit width of the sample. * Supported values: 16 */ u16 interleaved; /* Specifies whether the data exchanged between the AFE * interface and real-time port is interleaved. * Supported values: - 0 -- Non-interleaved (samples from each * channel are contiguous in the buffer) - 1 -- Interleaved * (corresponding samples from each input channel are interleaved * within the buffer) */ u16 frame_size; /* Size of the frames that are used for PCM exchanges with this * port. * Supported values: > 0, in bytes * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320 * bytes. */ u16 jitter_allowance; /* Configures the amount of jitter that the port will allow. * Supported values: > 0 * For example, if +/-10 ms of jitter is anticipated in the timing * of sending frames to the port, and the configuration is 16 kHz * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2 * bytes/sample = 320. */ u16 low_water_mark; /* Low watermark in bytes (including all channels). * Supported values: * - 0 -- Do not send any low watermark events * - > 0 -- Low watermark for triggering an event * If the number of bytes in an internal circular buffer is lower * than this low_water_mark parameter, a LOW_WATER_MARK event is * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS * event). * Use of watermark events is optional for debugging purposes. */ u16 high_water_mark; /* High watermark in bytes (including all channels). * Supported values: * - 0 -- Do not send any high watermark events * - > 0 -- High watermark for triggering an event * If the number of bytes in an internal circular buffer exceeds * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS * event). * The use of watermark events is optional and for debugging * purposes. */ u32 sample_rate; /* Sampling rate of the port. * Supported values: * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_16K * - #AFE_PORT_SAMPLE_RATE_48K */ u16 num_channels; /* Number of channels. * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT */ u16 reserved; /* For 32 bit alignment. */ } __packed; /* This param id is used to configure the Pseudoport interface */ #define AFE_PARAM_ID_PSEUDO_PORT_CONFIG 0x00010219 /* Version information used to handle future additions to the configuration * interface (for backward compatibility). */ #define AFE_API_VERSION_PSEUDO_PORT_CONFIG 0x1 /* Enumeration for setting the timing_mode parameter to faster than real * time. */ #define AFE_PSEUDOPORT_TIMING_MODE_FTRT 0x0 /* Enumeration for setting the timing_mode parameter to real time using * timers. */ #define AFE_PSEUDOPORT_TIMING_MODE_TIMER 0x1 /* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by AFE_MODULE_AUDIO_DEV_INTERFACE. */ struct afe_param_id_pseudo_port_cfg { u32 pseud_port_cfg_minor_version; /* * Minor version used for tracking the version of the pseudoport * configuration interface. */ u16 bit_width; /* Bit width of the sample at values 16, 24 */ u16 num_channels; /* Number of channels at values 1 to 8 */ u16 data_format; /* Non-linear data format supported by the pseudoport (for future use). * At values #AFE_LINEAR_PCM_DATA */ u16 timing_mode; /* Indicates whether the pseudoport synchronizes to the clock or * operates faster than real time. * at values * - #AFE_PSEUDOPORT_TIMING_MODE_FTRT * - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend */ u32 sample_rate; /* Sample rate at which the pseudoport will run. * at values * - #AFE_PORT_SAMPLE_RATE_8K * - #AFE_PORT_SAMPLE_RATE_32K * - #AFE_PORT_SAMPLE_RATE_48K * - #AFE_PORT_SAMPLE_RATE_96K * - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend */ } __packed; #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_API_VERSION_TDM_CONFIG 1 #define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE 0 #define AFE_PORT_TDM_LONG_SYNC_MODE 1 #define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE 2 #define AFE_PORT_TDM_SYNC_SRC_EXTERNAL 0 #define AFE_PORT_TDM_SYNC_SRC_INTERNAL 1 #define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE 0 #define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE 1 #define AFE_PORT_TDM_SYNC_NORMAL 0 #define AFE_PORT_TDM_SYNC_INVERT 1 #define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE 0 #define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE 1 #define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE 2 /* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by AFE_MODULE_AUDIO_DEV_INTERFACE. */ struct afe_param_id_tdm_cfg { u32 tdm_cfg_minor_version; /**< Minor version used to track TDM configuration. @values #AFE_API_VERSION_TDM_CONFIG */ u32 num_channels; /**< Number of enabled slots for TDM frame. @values 1 to 8 */ u32 sample_rate; /**< Sampling rate of the port. @values - #AFE_PORT_SAMPLE_RATE_8K - #AFE_PORT_SAMPLE_RATE_16K - #AFE_PORT_SAMPLE_RATE_24K - #AFE_PORT_SAMPLE_RATE_32K - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend */ u32 bit_width; /**< Bit width of the sample. @values 16, 24 */ u16 data_format; /**< Data format: linear and compressed @values - #AFE_LINEAR_PCM_DATA - #AFE_NON_LINEAR_DATA @tablebulletend */ u16 sync_mode; /**< TDM synchronization setting. @values (short, long, slot) sync mode - #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE - #AFE_PORT_TDM_LONG_SYNC_MODE - #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend */ u16 sync_src; /**< Synchronization source. @values - #AFE_PORT_TDM_SYNC_SRC_EXTERNAL - #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend */ u16 nslots_per_frame; /**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32. @values 1 - 32 */ u16 ctrl_data_out_enable; /**< Specifies whether the TDM block shares the data-out signal to the drive with other masters. @values - #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE - #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend */ u16 ctrl_invert_sync_pulse; /**< Specifies whether to invert the sync or not. @values - #AFE_PORT_TDM_SYNC_NORMAL - #AFE_PORT_TDM_SYNC_INVERT @tablebulletend */ u16 ctrl_sync_data_delay; /**< Specifies the number of bit clock to delay data with respect to sync edge. @values - #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE - #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE - #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend */ u16 slot_width; /**< Slot width of the slot in a TDM frame. (slot_width >= bit_width) have to be satisfied. @values 16, 24, 32 */ u32 slot_mask; /**< Position of active slots. When that bit is set, that paricular slot is active. Number of active slots can be inferred by number of bits set in the mask. Only 8 individual bits can be enabled. Bits 0..31 corresponding to slot 0..31 @values 1 to 2^32 - 1 */ } __packed; /** ID of Time Divsion Multiplexing (TDM) module, which is used for configuring the AFE TDM. This module supports following parameter IDs: - #AFE_PORT_TDM_SLOT_CONFIG To configure the TDM interface, the client must use the #AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the respective parameter IDs as listed above. */ #define AFE_MODULE_TDM 0x0001028A /** ID of the parameter used by #AFE_MODULE_TDM to configure the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. */ #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 /** Version information used to handle future additions to slot mapping configuration (for backward compatibility). */ #define AFE_API_VERSION_SLOT_MAPPING_CONFIG 0x1 /** Data align type */ #define AFE_SLOT_MAPPING_DATA_ALIGN_MSB 0 #define AFE_SLOT_MAPPING_DATA_ALIGN_LSB 1 #define AFE_SLOT_MAPPING_OFFSET_INVALID 0xFFFF /* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG command's TDM configuration parameter. */ struct afe_param_id_slot_mapping_cfg { u32 minor_version; /**< Minor version used for tracking TDM slot configuration. @values #AFE_API_VERSION_TDM_SLOT_CONFIG */ u16 num_channel; /**< number of channel of the audio sample. @values 1, 2, 4, 6, 8 @tablebulletend */ u16 bitwidth; /**< Slot bit width for each channel @values 16, 24, 32 */ u32 data_align_type; /**< indicate how data packed from slot_offset for 32 slot bit width in case of sample bit width is 24. @values #AFE_SLOT_MAPPING_DATA_ALIGN_MSB #AFE_SLOT_MAPPING_DATA_ALIGN_LSB */ u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT]; /**< Array of the slot mapping start offset in bytes for this frame. The bytes is counted from 0. The 0 is mapped to the 1st byte in or out of the digital serial data line this sub-frame belong to. slot_offset[] setting is per-channel based. The max num of channel supported is 8. The valid offset value must always be continuly placed in from index 0. Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays. If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24, "data_align_type" is used to indicate how 24 bit sample data in aligning with 32 bit slot width per-channel. @values, in byte*/ } __packed; /** ID of the parameter used by #AFE_MODULE_TDM to configure the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. */ #define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG 0x00010298 /** Version information used to handle future additions to custom TDM header configuration (for backward compatibility). */ #define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG 0x1 #define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID 0x0 #define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT 0x1 #define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST 0x2 #define AFE_CUSTOM_TDM_HEADER_MAX_CNT 0x8 /** Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID */ struct afe_param_id_custom_tdm_header_cfg { u32 minor_version; /**< Minor version used for tracking custom TDM header configuration. @values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG */ u16 start_offset; /**< the slot mapping start offset in bytes from this sub-frame The bytes is counted from 0. The 0 is mapped to the 1st byte in or out of the digital serial data line this sub-frame belong to. @values, in byte, supported values are 0, 4, 8, */ u16 header_width; /**< the header width per-frame followed. 2 bytes for MOST/TDM case @values, in byte supported value is 2 */ u16 header_type; /**< Indicate what kind of custom TDM header it is. @values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0 #AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1 (for AAN channel per MOST) #AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2 (for entertainment channel, which will overwrite AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST) */ u16 num_frame_repeat; /**< num of header followed. @values, supported value is 8*/ u16 header[AFE_CUSTOM_TDM_HEADER_MAX_CNT]; /** < SAD header for MOST/TDM case is followed as payload as below. The size of followed SAD header in bytes is num_of_frame_repeat * header_width_per_frame which is 2 * 8 = 16 bytes here. the supported payload format is in uint16_t as below uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00 uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218 uint16_t header2; DTCP Info Info[6] - unused -> 0x0 uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0 uint16_t header4; Reserved Info[0] - Num of bytes following -> 0x7 uint16_t header5; Reserved Info[1] - Media Type -> 0x0 uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0 uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low Byte -> 0x0 */ } __packed; struct afe_slot_mapping_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_id_slot_mapping_cfg slot_mapping; } __packed; struct afe_custom_tdm_header_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_id_custom_tdm_header_cfg custom_tdm_header; } __packed; struct afe_tdm_port_config { struct afe_param_id_tdm_cfg tdm; struct afe_param_id_slot_mapping_cfg slot_mapping; struct afe_param_id_custom_tdm_header_cfg custom_tdm_header; } __packed; #define AFE_PARAM_ID_DEVICE_HW_DELAY 0x00010243 #define AFE_API_VERSION_DEVICE_HW_DELAY 0x1 struct afe_param_id_device_hw_delay_cfg { uint32_t device_hw_delay_minor_version; uint32_t delay_in_us; } __packed; #define AFE_PARAM_ID_SET_TOPOLOGY 0x0001025A #define AFE_API_VERSION_TOPOLOGY_V1 0x1 struct afe_param_id_set_topology_cfg { /* * Minor version used for tracking afe topology id configuration. * @values #AFE_API_VERSION_TOPOLOGY_V1 */ u32 minor_version; /* * Id of the topology for the afe session. * @values Any valid AFE topology ID */ u32 topology_id; } __packed; union afe_port_config { struct afe_param_id_pcm_cfg pcm; struct afe_param_id_i2s_cfg i2s; struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; struct afe_param_id_slimbus_cfg slim_sch; struct afe_param_id_rt_proxy_port_cfg rtproxy; struct afe_param_id_internal_bt_fm_cfg int_bt_fm; struct afe_param_id_pseudo_port_cfg pseudo_port; struct afe_param_id_device_hw_delay_cfg hw_delay; struct afe_param_id_spdif_cfg spdif; struct afe_param_id_set_topology_cfg topology; struct afe_param_id_tdm_cfg tdm; } __packed; struct afe_audioif_config_command_no_payload { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; } __packed; struct afe_audioif_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; union afe_port_config port; } __packed; #define AFE_PORT_CMD_DEVICE_START 0x000100E5 /* Payload of the #AFE_PORT_CMD_DEVICE_START.*/ struct afe_port_cmd_device_start { struct apr_hdr hdr; u16 port_id; /* Port interface and direction (Rx or Tx) to start. An even * number represents the Rx direction, and an odd number represents * the Tx direction. */ u16 reserved; /* Reserved for 32-bit alignment. This field must be set to 0.*/ } __packed; #define AFE_PORT_CMD_DEVICE_STOP 0x000100E6 /* Payload of the #AFE_PORT_CMD_DEVICE_STOP. */ struct afe_port_cmd_device_stop { struct apr_hdr hdr; u16 port_id; /* Port interface and direction (Rx or Tx) to start. An even * number represents the Rx direction, and an odd number represents * the Tx direction. */ u16 reserved; /* Reserved for 32-bit alignment. This field must be set to 0.*/ } __packed; #define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA /* Memory map regions command payload used by the * #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS . * This structure allows clients to map multiple shared memory * regions in a single command. Following this structure are * num_regions of afe_service_shared_map_region_payload. */ struct afe_service_cmd_shared_mem_map_regions { struct apr_hdr hdr; u16 mem_pool_id; /* Type of memory on which this memory region is mapped. * Supported values: * - #ADSP_MEMORY_MAP_EBI_POOL * - #ADSP_MEMORY_MAP_SMI_POOL * - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL * - Other values are reserved * * The memory pool ID implicitly defines the characteristics of the * memory. Characteristics may include alignment type, permissions, * etc. * * ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory * ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory * ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte * addressable, and 4 KB aligned. */ u16 num_regions; /* Number of regions to map. * Supported values: * - Any value greater than zero */ u32 property_flag; /* Configures one common property for all the regions in the * payload. * * Supported values: - 0x00000000 to 0x00000001 * * b0 - bit 0 indicates physical or virtual mapping 0 Shared memory * address provided in afe_service_shared_map_region_payloadis a * physical address. The shared memory needs to be mapped( hardware * TLB entry) and a software entry needs to be added for internal * book keeping. * * 1 Shared memory address provided in * afe_service_shared_map_region_payloadis a virtual address. The * shared memory must not be mapped (since hardware TLB entry is * already available) but a software entry needs to be added for * internal book keeping. This can be useful if two services with in * ADSP is communicating via APR. They can now directly communicate * via the Virtual address instead of Physical address. The virtual * regions must be contiguous. num_regions must be 1 in this case. * * b31-b1 - reserved bits. must be set to zero */ } __packed; /* Map region payload used by the * afe_service_shared_map_region_payloadstructure. */ struct afe_service_shared_map_region_payload { u32 shm_addr_lsw; /* least significant word of starting address in the memory * region to map. It must be contiguous memory, and it must be 4 KB * aligned. * Supported values: - Any 32 bit value */ u32 shm_addr_msw; /* most significant word of startng address in the memory region * to map. For 32 bit shared memory address, this field must be set * to zero. For 36 bit shared memory address, bit31 to bit 4 must be * set to zero * * Supported values: - For 32 bit shared memory address, this field * must be set to zero. - For 36 bit shared memory address, bit31 to * bit 4 must be set to zero - For 64 bit shared memory address, any * 32 bit value */ u32 mem_size_bytes; /* Number of bytes in the region. The aDSP will always map the * regions as virtual contiguous memory, but the memory size must be * in multiples of 4 KB to avoid gaps in the virtually contiguous * mapped memory. * * Supported values: - multiples of 4KB */ } __packed; #define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB struct afe_service_cmdrsp_shared_mem_map_regions { u32 mem_map_handle; /* A memory map handle encapsulating shared memory attributes is * returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is * successful. In the case of failure , a generic APR error response * is returned to the client. * * Supported Values: - Any 32 bit value */ } __packed; #define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC /* Memory unmap regions command payload used by the * #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS * * This structure allows clients to unmap multiple shared memory * regions in a single command. */ struct afe_service_cmd_shared_mem_unmap_regions { struct apr_hdr hdr; u32 mem_map_handle; /* memory map handle returned by * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands * * Supported Values: * - Any 32 bit value */ } __packed; #define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0 /* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command, * which queries for one post/preprocessing parameter of a * stream. */ struct afe_port_cmd_get_param_v2 { u16 port_id; /* Port interface and direction (Rx or Tx) to start. */ u16 payload_size; /* Maximum data size of the parameter ID/module ID combination. * This is a multiple of four bytes * Supported values: > 0 */ u32 payload_address_lsw; /* LSW of 64 bit Payload address. Address should be 32-byte, * 4kbyte aligned and must be contig memory. */ u32 payload_address_msw; /* MSW of 64 bit Payload address. In case of 32-bit shared * memory address, this field must be set to zero. In case of 36-bit * shared memory address, bit-4 to bit-31 must be set to zero. * Address should be 32-byte, 4kbyte aligned and must be contiguous * memory. */ u32 mem_map_handle; /* Memory map handle returned by * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands. * Supported Values: - NULL -- Message. The parameter data is * in-band. - Non-NULL -- The parameter data is Out-band.Pointer to * - the physical address in shared memory of the payload data. * For detailed payload content, see the afe_port_param_data_v2 * structure */ u32 module_id; /* ID of the module to be queried. * Supported values: Valid module ID */ u32 param_id; /* ID of the parameter to be queried. * Supported values: Valid parameter ID */ } __packed; #define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106 /* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which * responds to an #AFE_PORT_CMD_GET_PARAM_V2 command. * * Immediately following this structure is the parameters structure * (afe_port_param_data) containing the response(acknowledgment) * parameter payload. This payload is included for an in-band * scenario. For an address/shared memory-based set parameter, this * payload is not needed. */ struct afe_port_cmdrsp_get_param_v2 { u32 status; } __packed; #define AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG 0x0001028C #define AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG 0x1 /* * Payload of the AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG parameter used by * AFE_MODULE_AUDIO_DEV_INTERFACE. */ struct afe_param_id_lpass_core_shared_clk_cfg { u32 lpass_core_shared_clk_cfg_minor_version; /* * Minor version used for lpass core shared clock configuration * Supported value: AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG */ u32 enable; /* * Specifies whether the lpass core shared clock is * enabled (1) or disabled (0). */ } __packed; struct afe_lpass_core_shared_clk_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_id_lpass_core_shared_clk_cfg clk_cfg; } __packed; /* adsp_afe_service_commands.h */ #define ADSP_MEMORY_MAP_EBI_POOL 0 #define ADSP_MEMORY_MAP_SMI_POOL 1 #define ADSP_MEMORY_MAP_IMEM_POOL 2 #define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3 /* * Definition of virtual memory flag */ #define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1 /* * Definition of physical memory flag */ #define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0 #define NULL_POPP_TOPOLOGY 0x00010C68 #define NULL_COPP_TOPOLOGY 0x00010312 #define DEFAULT_COPP_TOPOLOGY 0x00010314 #define DEFAULT_POPP_TOPOLOGY 0x00010BE4 #define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY 0x0001076B #define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71 #define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72 #define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75 #define VPM_TX_DM_RFECNS_COPP_TOPOLOGY 0x00010F86 #define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_DTS_HPX 0x10015002 #define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_AUDIOSPHERE 0x10028000 /* Memory map regions command payload used by the * #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS * commands. * * This structure allows clients to map multiple shared memory * regions in a single command. Following this structure are * num_regions of avs_shared_map_region_payload. */ struct avs_cmd_shared_mem_map_regions { struct apr_hdr hdr; u16 mem_pool_id; /* Type of memory on which this memory region is mapped. * * Supported values: - #ADSP_MEMORY_MAP_EBI_POOL - * #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL * (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values * are reserved * * The memory ID implicitly defines the characteristics of the * memory. Characteristics may include alignment type, permissions, * etc. * * SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned. */ u16 num_regions; /* Number of regions to map.*/ u32 property_flag; /* Configures one common property for all the regions in the * payload. No two regions in the same memory map regions cmd can * have differnt property. Supported values: - 0x00000000 to * 0x00000001 * * b0 - bit 0 indicates physical or virtual mapping 0 shared memory * address provided in avs_shared_map_regions_payload is physical * address. The shared memory needs to be mapped( hardware TLB * entry) * * and a software entry needs to be added for internal book keeping. * * 1 Shared memory address provided in MayPayload[usRegions] is * virtual address. The shared memory must not be mapped (since * hardware TLB entry is already available) but a software entry * needs to be added for internal book keeping. This can be useful * if two services with in ADSP is communicating via APR. They can * now directly communicate via the Virtual address instead of * Physical address. The virtual regions must be contiguous. * * b31-b1 - reserved bits. must be set to zero */ } __packed; struct avs_shared_map_region_payload { u32 shm_addr_lsw; /* least significant word of shared memory address of the memory * region to map. It must be contiguous memory, and it must be 4 KB * aligned. */ u32 shm_addr_msw; /* most significant word of shared memory address of the memory * region to map. For 32 bit shared memory address, this field must * tbe set to zero. For 36 bit shared memory address, bit31 to bit 4 * must be set to zero */ u32 mem_size_bytes; /* Number of bytes in the region. * * The aDSP will always map the regions as virtual contiguous * memory, but the memory size must be in multiples of 4 KB to avoid * gaps in the virtually contiguous mapped memory. */ } __packed; struct avs_cmd_shared_mem_unmap_regions { struct apr_hdr hdr; u32 mem_map_handle; /* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS * , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands */ } __packed; /* Memory map command response payload used by the * #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS * ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS */ struct avs_cmdrsp_shared_mem_map_regions { u32 mem_map_handle; /* A memory map handle encapsulating shared memory attributes is * returned */ } __packed; /*adsp_audio_memmap_api.h*/ /* ASM related data structures */ struct asm_wma_cfg { u16 format_tag; u16 ch_cfg; u32 sample_rate; u32 avg_bytes_per_sec; u16 block_align; u16 valid_bits_per_sample; u32 ch_mask; u16 encode_opt; u16 adv_encode_opt; u32 adv_encode_opt2; u32 drc_peak_ref; u32 drc_peak_target; u32 drc_ave_ref; u32 drc_ave_target; } __packed; struct asm_wmapro_cfg { u16 format_tag; u16 ch_cfg; u32 sample_rate; u32 avg_bytes_per_sec; u16 block_align; u16 valid_bits_per_sample; u32 ch_mask; u16 encode_opt; u16 adv_encode_opt; u32 adv_encode_opt2; u32 drc_peak_ref; u32 drc_peak_target; u32 drc_ave_ref; u32 drc_ave_target; } __packed; struct asm_aac_cfg { u16 format; u16 aot; u16 ep_config; u16 section_data_resilience; u16 scalefactor_data_resilience; u16 spectral_data_resilience; u16 ch_cfg; u16 reserved; u32 sample_rate; } __packed; struct asm_amrwbplus_cfg { u32 size_bytes; u32 version; u32 num_channels; u32 amr_band_mode; u32 amr_dtx_mode; u32 amr_frame_fmt; u32 amr_lsf_idx; } __packed; struct asm_flac_cfg { u32 sample_rate; u32 ext_sample_rate; u32 min_frame_size; u32 max_frame_size; u16 stream_info_present; u16 min_blk_size; u16 max_blk_size; u16 ch_cfg; u16 sample_size; u16 md5_sum; }; struct asm_alac_cfg { u32 frame_length; u8 compatible_version; u8 bit_depth; u8 pb; u8 mb; u8 kb; u8 num_channels; u16 max_run; u32 max_frame_bytes; u32 avg_bit_rate; u32 sample_rate; u32 channel_layout_tag; }; struct asm_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_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 /* Enumeration for the AAC LATM format. */ #define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM 4 #define ASM_MEDIA_FMT_AAC_AOT_LC 2 #define ASM_MEDIA_FMT_AAC_AOT_SBR 5 #define ASM_MEDIA_FMT_AAC_AOT_PS 29 #define ASM_MEDIA_FMT_AAC_AOT_BSAC 22 struct asm_aac_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmt_blk; u16 aac_fmt_flag; /* Bitstream format option. * Supported values: * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW */ u16 audio_objype; /* Audio Object Type (AOT) present in the AAC stream. * Supported values: * - #ASM_MEDIA_FMT_AAC_AOT_LC * - #ASM_MEDIA_FMT_AAC_AOT_SBR * - #ASM_MEDIA_FMT_AAC_AOT_BSAC * - #ASM_MEDIA_FMT_AAC_AOT_PS * - Otherwise -- Not supported */ u16 channel_config; /* Number of channels present in the AAC stream. * Supported values: * - 1 -- Mono * - 2 -- Stereo * - 6 -- 5.1 content */ u16 total_size_of_PCE_bits; /* greater or equal to zero. * -In case of RAW formats and * channel config = 0 (PCE), client can send * the bit stream * containing PCE immediately following this structure * (in-band). * -This number does not include bits included for 32 bit alignment. * -If zero, then the PCE info is assumed to be available in the * audio -bit stream & not in-band. */ u32 sample_rate; /* Number of samples per second (in Hertz). * * Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000, * 44100, 48000 * * This field must be equal to the sample rate of the AAC-LC * decoder's output. - For MP4 or 3GP containers, this is indicated * by the samplingFrequencyIndex field in the AudioSpecificConfig * element. - For ADTS format, this is indicated by the * samplingFrequencyIndex in the ADTS fixed header. - For ADIF * format, this is indicated by the samplingFrequencyIndex in the * program_config_element present in the ADIF header. */ } __packed; struct asm_aac_enc_cfg_v2 { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; struct asm_enc_cfg_blk_param_v2 encblk; u32 bit_rate; /* Encoding rate in bits per second. */ u32 enc_mode; /* Encoding mode. * Supported values: * - #ASM_MEDIA_FMT_AAC_AOT_LC * - #ASM_MEDIA_FMT_AAC_AOT_SBR * - #ASM_MEDIA_FMT_AAC_AOT_PS */ u16 aac_fmt_flag; /* AAC format flag. * Supported values: * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW */ u16 channel_cfg; /* Number of channels to encode. * Supported values: * - 0 -- Native mode * - 1 -- Mono * - 2 -- Stereo * - Other values are not supported. * @note1hang The eAAC+ encoder mode supports only stereo. * Native mode indicates that encoding must be performed with the * number of channels at the input. * The number of channels must not change during encoding. */ u32 sample_rate; /* Number of samples per second. * Supported values: - 0 -- Native mode - For other values, * Native mode indicates that encoding must be performed with the * sampling rate at the input. * The sampling rate must not change during encoding. */ } __packed; #define ASM_MEDIA_FMT_G711_ALAW_FS 0x00010BF7 #define ASM_MEDIA_FMT_G711_MLAW_FS 0x00010C2E struct asm_g711_enc_cfg_v2 { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; struct asm_enc_cfg_blk_param_v2 encblk; u32 sample_rate; /* * Number of samples per second. * Supported values: 8000, 16000 Hz */ } __packed; struct asm_vorbis_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u32 bit_stream_fmt; /* Bit stream format. * Supported values: * - 0 -- Raw bitstream * - 1 -- Transcoded bitstream * * Transcoded bitstream containing the size of the frame as the first * word in each frame. */ } __packed; struct asm_flac_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u16 is_stream_info_present; /* Specifies whether stream information is present in the FLAC format * block. * * Supported values: * - 0 -- Stream information is not present in this message * - 1 -- Stream information is present in this message * * When set to 1, the FLAC bitstream was successfully parsed by the * client, and other fields in the FLAC format block can be read by the * decoder to get metadata stream information. */ u16 num_channels; /* Number of channels for decoding. * Supported values: 1 to 2 */ u16 min_blk_size; /* Minimum block size (in samples) used in the stream. It must be less * than or equal to max_blk_size. */ u16 max_blk_size; /* Maximum block size (in samples) used in the stream. If the * minimum block size equals the maximum block size, a fixed block * size stream is implied. */ u16 md5_sum[8]; /* MD5 signature array of the unencoded audio data. This allows the * decoder to determine if an error exists in the audio data, even when * the error does not result in an invalid bitstream. */ u32 sample_rate; /* Number of samples per second. * Supported values: 8000 to 48000 Hz */ u32 min_frame_size; /* Minimum frame size used in the stream. * Supported values: * - > 0 bytes * - 0 -- The value is unknown */ u32 max_frame_size; /* Maximum frame size used in the stream. * Supported values: * -- > 0 bytes * -- 0 . The value is unknown */ u16 sample_size; /* Bits per sample.Supported values: 8, 16 */ u16 reserved; /* Clients must set this field to zero */ } __packed; struct asm_alac_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u32 frame_length; u8 compatible_version; u8 bit_depth; u8 pb; u8 mb; u8 kb; u8 num_channels; u16 max_run; u32 max_frame_bytes; u32 avg_bit_rate; u32 sample_rate; u32 channel_layout_tag; } __packed; struct asm_g711_dec_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u32 sample_rate; } __packed; struct asm_ape_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u16 compatible_version; u16 compression_level; u32 format_flags; u32 blocks_per_frame; u32 final_frame_blocks; u32 total_frames; u16 bits_per_sample; u16 num_channels; u32 sample_rate; u32 seek_table_present; } __packed; #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_APTX 0x000131FF /* 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. None 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. None 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. None means no postprocessing. * * Supported values: * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL * - #ASM_STREAM_POSTPROCOPO_ID_NONE */ u32 dec_fmt_id; /* Specifies the media type of the input data. PCM indicates that * no decoding must be performed, e.g., this is an NT encoder * session. * Supported values: * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 * - #ASM_MEDIA_FMT_ADPCM * - #ASM_MEDIA_FMT_MP3 * - #ASM_MEDIA_FMT_AAC_V2 * - #ASM_MEDIA_FMT_DOLBY_AAC * - #ASM_MEDIA_FMT_AMRNB_FS * - #ASM_MEDIA_FMT_AMRWB_FS * - #ASM_MEDIA_FMT_V13K_FS * - #ASM_MEDIA_FMT_EVRC_FS * - #ASM_MEDIA_FMT_EVRCB_FS * - #ASM_MEDIA_FMT_EVRCWB_FS * - #ASM_MEDIA_FMT_SBC * - #ASM_MEDIA_FMT_WMA_V10PRO_V2 * - #ASM_MEDIA_FMT_WMA_V9_V2 * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2 * - #ASM_MEDIA_FMT_AC3 * - #ASM_MEDIA_FMT_G711_ALAW_FS * - #ASM_MEDIA_FMT_G711_MLAW_FS * - #ASM_MEDIA_FMT_G729A_FS * - #ASM_MEDIA_FMT_EXAMPLE */ u32 enc_cfg_id; /* Specifies the media type for the output of the stream. PCM * indicates that no encoding must be performed, e.g., this is an NT * decoder session. * Supported values: * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 * - #ASM_MEDIA_FMT_AAC_V2 * - #ASM_MEDIA_FMT_AMRNB_FS * - #ASM_MEDIA_FMT_AMRWB_FS * - #ASM_MEDIA_FMT_V13K_FS * - #ASM_MEDIA_FMT_EVRC_FS * - #ASM_MEDIA_FMT_EVRCB_FS * - #ASM_MEDIA_FMT_EVRCWB_FS * - #ASM_MEDIA_FMT_SBC * - #ASM_MEDIA_FMT_G711_ALAW_FS * - #ASM_MEDIA_FMT_G711_MLAW_FS * - #ASM_MEDIA_FMT_G729A_FS * - #ASM_MEDIA_FMT_EXAMPLE * - #ASM_MEDIA_FMT_WMA_V8 */ u16 bits_per_sample; /* Number of bits per sample processed by ASM modules. * Supported values: 16 and 24 bits per sample */ u16 reserved; /* Reserved for future use. This field must be set to zero.*/ } __packed; #define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E struct asm_stream_cmd_open_loopback_v2 { struct apr_hdr hdr; u32 mode_flags; /* Mode flags. * Bit 0-31: reserved; client should set these bits to 0 */ u16 src_endpointype; /* Endpoint type. 0 = Tx Matrix */ u16 sink_endpointype; /* Endpoint type. 0 = Rx Matrix */ u32 postprocopo_id; /* Postprocessor topology ID. Specifies the topology of * postprocessing algorithms. */ u16 bits_per_sample; /* The number of bits per sample processed by ASM modules * Supported values: 16 and 24 bits per sample */ u16 reserved; /* Reserved for future use. This field must be set to zero. */ } __packed; #define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA /* Bitmask for the stream's Performance mode. */ #define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \ (0x70000000UL) /* Bit shift for the stream's Performance mode. */ #define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28 /* Bitmask for the decoder converter enable flag. */ #define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL) /* Shift value for the decoder converter enable flag. */ #define ASM_SHIFT_DECODER_CONVERTER_FLAG 3 /* Converter mode is None (Default). */ #define ASM_CONVERTER_MODE_NONE 0 /* Converter mode is DDP-to-DD. */ #define ASM_DDP_DD_CONVERTER_MODE 1 /* Identifies a special converter mode where source and sink formats * are the same but postprocessing must applied. Therefore, Decode * @rarrow Re-encode is necessary. */ #define ASM_POST_PROCESS_CONVERTER_MODE 2 struct asm_stream_cmd_open_transcode_loopback_t { struct apr_hdr hdr; u32 mode_flags; /* Mode Flags specifies the performance mode in which this stream * is to be opened. * Supported values{for bits 30 to 28}(stream_perf_mode flag) * * #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward * compatibility to the original behavior * of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK * * #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using * shortened buffers in low latency POPP * - Recommendation: Do not enable high latency algorithms. They might * negate the benefits of opening a low latency stream, and they * might also suffer quality degradation from unexpected jitter. * - This Low Latency mode is supported only for PCM In and PCM Out * loopbacks. An error is returned if Low Latency mode is opened for * other transcode loopback modes. * - To configure this subfield, use * ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and * ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK. * * Supported values{for bits 6 to 3} (decoder-converter compatibility) * #ASM_CONVERTER_MODE_NONE (0x0) -- Default * #ASM_DDP_DD_CONVERTER_MODE (0x1) * #ASM_POST_PROCESS_CONVERTER_MODE (0x2) * 0x3-0xF -- Reserved for future use * - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and * ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit * All other bits are reserved; clients must set them to 0. */ u32 src_format_id; /* Specifies the media format of the input audio stream. * * Supported values * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 * - #ASM_MEDIA_FMT_DTS * - #ASM_MEDIA_FMT_EAC3_DEC * - #ASM_MEDIA_FMT_EAC3 * - #ASM_MEDIA_FMT_AC3_DEC * - #ASM_MEDIA_FMT_AC3 */ u32 sink_format_id; /* Specifies the media format of the output stream. * * Supported values * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 * - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode) * - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode) * - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode) * - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode) * - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode) */ u32 audproc_topo_id; /* Postprocessing topology ID, which specifies the topology (order of * processing) of postprocessing algorithms. * * Supported values * - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT * - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER * - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL * - #ASM_STREAM_POSTPROC_TOPO_ID_NONE * Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES. * This field is ignored for the Converter mode, in which no * postprocessing is performed. */ u16 src_endpoint_type; /* Specifies the source endpoint that provides the input samples. * * Supported values * - 0 -- Tx device matrix or stream router (gateway to the hardware * ports) * - All other values are reserved * Clients must set this field to 0. Otherwise, an error is returned. */ u16 sink_endpoint_type; /* Specifies the sink endpoint type. * * Supported values * - 0 -- Rx device matrix or stream router (gateway to the hardware * ports) * - All other values are reserved * Clients must set this field to 0. Otherwise, an error is returned. */ u16 bits_per_sample; /* Number of bits per sample processed by the ASM modules. * Supported values 16, 24 */ u16 reserved; /* This field must be set to 0. */ } __packed; #define ASM_STREAM_CMD_CLOSE 0x00010BCD #define ASM_STREAM_CMD_FLUSH 0x00010BCE #define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09 #define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1 struct asm_stream_cmd_set_pp_params_v2 { u32 data_payload_addr_lsw; /* LSW of parameter data payload address. Supported values: any. */ u32 data_payload_addr_msw; /* MSW of Parameter data payload address. Supported values: any. * - Must be set to zero for in-band data. * - In the case of 32 bit Shared memory address, msw field must be * - set to zero. * - In the case of 36 bit shared memory address, bit 31 to bit 4 of * msw * * - must be set to zero. */ u32 mem_map_handle; /* Supported Values: Any. * memory map handle returned by DSP through * ASM_CMD_SHARED_MEM_MAP_REGIONS * command. * if mmhandle is NULL, the ParamData payloads are within the * message payload (in-band). * If mmhandle is non-NULL, the ParamData payloads begin at the * address specified in the address msw and lsw (out-of-band). */ u32 data_payload_size; /* Size in bytes of the variable payload accompanying the message, or in shared memory. This field is used for parsing the parameter payload. */ } __packed; struct asm_stream_param_data_v2 { u32 module_id; /* Unique module ID. */ u32 param_id; /* Unique parameter ID. */ u16 param_size; /* Data size of the param_id/module_id combination. This is * a multiple of 4 bytes. */ u16 reserved; /* Reserved for future enhancements. This field must be set to * zero. */ } __packed; #define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2 struct asm_stream_cmd_get_pp_params_v2 { u32 data_payload_addr_lsw; /* LSW of the parameter data payload address. */ u32 data_payload_addr_msw; /* MSW of the parameter data payload address. * - Size of the shared memory, if specified, shall be large enough * to contain the whole ParamData payload, including Module ID, * Param ID, Param Size, and Param Values * - Must be set to zero for in-band data * - In the case of 32 bit Shared memory address, msw field must be * set to zero. * - In the case of 36 bit shared memory address, bit 31 to bit 4 of * msw must be set to zero. */ u32 mem_map_handle; /* Supported Values: Any. * memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS * command. * if mmhandle is NULL, the ParamData payloads in the ACK are within the * message payload (in-band). * If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the * address specified in the address msw and lsw. * (out-of-band). */ u32 module_id; /* Unique module ID. */ u32 param_id; /* Unique parameter ID. */ u16 param_max_size; /* Maximum data size of the module_id/param_id combination. This * is a multiple of 4 bytes. */ u16 reserved; /* Reserved for backward compatibility. Clients must set this * field to zero. */ } __packed; #define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10 #define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13 struct asm_bitrate_param { u32 bitrate; /* Maximum supported bitrate. Only the AAC encoder is supported.*/ } __packed; #define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3 #define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63 /* Flag to turn off both SBR and PS processing, if they are * present in the bitstream. */ #define ASM_AAC_SBR_OFF_PS_OFF (2) /* Flag to turn on SBR but turn off PS processing,if they are * present in the bitstream. */ #define ASM_AAC_SBR_ON_PS_OFF (1) /* Flag to turn on both SBR and PS processing, if they are * present in the bitstream (default behavior). */ #define ASM_AAC_SBR_ON_PS_ON (0) /* Structure for an AAC SBR PS processing flag. */ /* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ struct asm_aac_sbr_ps_flag_param { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; struct asm_enc_cfg_blk_param_v2 encblk; u32 sbr_ps_flag; /* Control parameter to enable or disable SBR/PS processing in * the AAC bitstream. Use the following macros to set this field: * - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS * processing, if they are present in the bitstream. * - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS * processing, if they are present in the bitstream. * - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing, * if they are present in the bitstream (default behavior). * - All other values are invalid. * Changes are applied to the next decoded frame. */ } __packed; #define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64 /* First single channel element in a dual mono bitstream.*/ #define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1) /* Second single channel element in a dual mono bitstream.*/ #define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2) /* Structure for AAC decoder dual mono channel mapping. */ struct asm_aac_dual_mono_mapping_param { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; u16 left_channel_sce; u16 right_channel_sce; } __packed; #define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4 struct asm_stream_cmdrsp_get_pp_params_v2 { u32 status; } __packed; #define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73 /* Enumeration for both vocals in a karaoke stream.*/ #define AC3_KARAOKE_MODE_NO_VOCAL (0) /* Enumeration for only the left vocal in a karaoke stream.*/ #define AC3_KARAOKE_MODE_LEFT_VOCAL (1) /* Enumeration for only the right vocal in a karaoke stream.*/ #define AC3_KARAOKE_MODE_RIGHT_VOCAL (2) /* Enumeration for both vocal channels in a karaoke stream.*/ #define AC3_KARAOKE_MODE_BOTH_VOCAL (3) #define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74 /* Enumeration for the Custom Analog mode.*/ #define AC3_DRC_MODE_CUSTOM_ANALOG (0) /* Enumeration for the Custom Digital mode.*/ #define AC3_DRC_MODE_CUSTOM_DIGITAL (1) /* Enumeration for the Line Out mode (light compression).*/ #define AC3_DRC_MODE_LINE_OUT (2) /* Enumeration for the RF remodulation mode (heavy compression).*/ #define AC3_DRC_MODE_RF_REMOD (3) #define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75 /* Enumeration for playing dual mono in stereo mode.*/ #define AC3_DUAL_MONO_MODE_STEREO (0) /* Enumeration for playing left mono.*/ #define AC3_DUAL_MONO_MODE_LEFT_MONO (1) /* Enumeration for playing right mono.*/ #define AC3_DUAL_MONO_MODE_RIGHT_MONO (2) /* Enumeration for mixing both dual mono channels and playing them.*/ #define AC3_DUAL_MONO_MODE_MIXED_MONO (3) #define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76 /* Enumeration for using the Downmix mode indicated in the bitstream. */ #define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0) /* Enumeration for Surround Compatible mode (preserves the * surround information). */ #define AC3_STEREO_DOWNMIX_MODE_LT_RT (1) /* Enumeration for Mono Compatible mode (if the output is to be * further downmixed to mono). */ #define AC3_STEREO_DOWNMIX_MODE_LO_RO (2) /* ID of the AC3 PCM scale factor parameter in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ #define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78 /* ID of the AC3 DRC boost scale factor parameter in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ #define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79 /* ID of the AC3 DRC cut scale factor parameter in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ #define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A /* Structure for AC3 Generic Parameter. */ /* Payload of the AC3 parameters in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ struct asm_ac3_generic_param { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; struct asm_enc_cfg_blk_param_v2 encblk; u32 generic_parameter; /* AC3 generic parameter. Select from one of the following * possible values. * * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are: * - AC3_KARAOKE_MODE_NO_VOCAL * - AC3_KARAOKE_MODE_LEFT_VOCAL * - AC3_KARAOKE_MODE_RIGHT_VOCAL * - AC3_KARAOKE_MODE_BOTH_VOCAL * * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are: * - AC3_DRC_MODE_CUSTOM_ANALOG * - AC3_DRC_MODE_CUSTOM_DIGITAL * - AC3_DRC_MODE_LINE_OUT * - AC3_DRC_MODE_RF_REMOD * * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are: * - AC3_DUAL_MONO_MODE_STEREO * - AC3_DUAL_MONO_MODE_LEFT_MONO * - AC3_DUAL_MONO_MODE_RIGHT_MONO * - AC3_DUAL_MONO_MODE_MIXED_MONO * * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are: * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT * - AC3_STEREO_DOWNMIX_MODE_LT_RT * - AC3_STEREO_DOWNMIX_MODE_LO_RO * * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are * 0 to 1 in Q31 format. * * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are * 0 to 1 in Q31 format. * * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are * 0 to 1 in Q31 format. */ } __packed; /* Enumeration for Raw mode (no downmixing), which specifies * that all channels in the bitstream are to be played out as is * without any downmixing. (Default) */ #define WMAPRO_CHANNEL_MASK_RAW (-1) /* Enumeration for setting the channel mask to 0. The 7.1 mode * (Home Theater) is assigned. */ #define WMAPRO_CHANNEL_MASK_ZERO 0x0000 /* Speaker layout mask for one channel (Home Theater, mono). * - Speaker front center */ #define WMAPRO_CHANNEL_MASK_1_C 0x0004 /* Speaker layout mask for two channels (Home Theater, stereo). * - Speaker front left * - Speaker front right */ #define WMAPRO_CHANNEL_MASK_2_L_R 0x0003 /* Speaker layout mask for three channels (Home Theater). * - Speaker front left * - Speaker front right * - Speaker front center */ #define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007 /* Speaker layout mask for two channels (stereo). * - Speaker back left * - Speaker back right */ #define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030 /* Speaker layout mask for four channels. * - Speaker front left * - Speaker front right * - Speaker back left * - Speaker back right */ #define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033 /* Speaker layout mask for four channels (Home Theater). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back center */ #define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107 /* Speaker layout mask for five channels. * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back left * - Speaker back right */ #define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037 /* Speaker layout mask for five channels (5 mode, Home Theater). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker side left * - Speaker side right */ #define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607 /* Speaker layout mask for six channels (5.1 mode). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker low frequency * - Speaker back left * - Speaker back right */ #define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F /* Speaker layout mask for six channels (5.1 mode, Home Theater). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker low frequency * - Speaker side left * - Speaker side right */ #define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F /* Speaker layout mask for six channels (5.1 mode, no LFE). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back left * - Speaker back right * - Speaker back center */ #define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137 /* Speaker layout mask for six channels (5.1 mode, Home Theater, * no LFE). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back center * - Speaker side left * - Speaker side right */ #define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707 /* Speaker layout mask for seven channels (6.1 mode). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker low frequency * - Speaker back left * - Speaker back right * - Speaker back center */ #define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F /* Speaker layout mask for seven channels (6.1 mode, Home * Theater). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker low frequency * - Speaker back center * - Speaker side left * - Speaker side right */ #define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F /* Speaker layout mask for seven channels (6.1 mode, no LFE). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back left * - Speaker back right * - Speaker front left of center * - Speaker front right of center */ #define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7 /* Speaker layout mask for seven channels (6.1 mode, Home * Theater, no LFE). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker side left * - Speaker side right * - Speaker front left of center * - Speaker front right of center */ #define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637 /* Speaker layout mask for eight channels (7.1 mode). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker back left * - Speaker back right * - Speaker low frequency * - Speaker front left of center * - Speaker front right of center */ #define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \ 0x00FF /* Speaker layout mask for eight channels (7.1 mode, Home Theater). * - Speaker front left * - Speaker front right * - Speaker front center * - Speaker side left * - Speaker side right * - Speaker low frequency * - Speaker front left of center * - Speaker front right of center * */ #define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \ 0x063F #define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82 /* Maximum number of decoder output channels.*/ #define MAX_CHAN_MAP_CHANNELS 16 /* Structure for decoder output channel mapping. */ /* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. */ struct asm_dec_out_chan_map_param { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; u32 num_channels; /* Number of decoder output channels. * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS * * A value of 0 indicates native channel mapping, which is valid * only for NT mode. This means the output of the decoder is to be * preserved as is. */ u8 channel_mapping[MAX_CHAN_MAP_CHANNELS]; } __packed; #define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84 /* Bitmask for the IEC 61937 enable flag.*/ #define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL) /* Shift value for the IEC 61937 enable flag.*/ #define ASM_SHIFT_IEC_61937_STREAM_FLAG 0 /* Bitmask for the IEC 60958 enable flag.*/ #define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL) /* Shift value for the IEC 60958 enable flag.*/ #define ASM_SHIFT_IEC_60958_STREAM_FLAG 1 /* Payload format for open write compressed 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; /*module id * data variable payload 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; /* Opcode to set BT address and license for aptx decoder */ #define APTX_DECODER_BT_ADDRESS 0x00013201 #define APTX_CLASSIC_DEC_LICENSE_ID 0x00013202 struct aptx_dec_bt_addr_cfg { uint32_t lap; uint32_t uap; uint32_t nap; } __packed; struct aptx_dec_bt_dev_addr { struct apr_hdr hdr; struct asm_stream_cmd_set_encdec_param encdec; struct aptx_dec_bt_addr_cfg bt_addr_cfg; } __packed; struct asm_aptx_dec_fmt_blk_v2 { struct apr_hdr hdr; struct asm_data_cmd_media_fmt_update_v2 fmtblk; u32 sample_rate; /* Number of samples per second. * Supported values: 44100 and 48000 Hz */ } __packed; /* Q6Core Specific */ #define AVCS_CMD_GET_FWK_VERSION (0x0001292C) #define AVCS_CMDRSP_GET_FWK_VERSION (0x0001292D) #define AVCS_SERVICE_ID_ALL (0xFFFFFFFF) #define AVCS_SERVICE_ID_AFE (0x4) struct avcs_get_fwk_version { /* * Indicates the major version of the AVS build. * This value is incremented on chipset family boundaries. */ uint32_t build_major_version; /* * Minor version of the AVS build. * This value represents the mainline to which the AVS build belongs. */ uint32_t build_minor_version; /* Indicates the AVS branch version to which the image belongs. */ uint32_t build_branch_version; /* Indicates the AVS sub-branch or customer product line information. */ uint32_t build_subbranch_version; /* Number of supported AVS services in the current build. */ uint32_t num_services; }; struct avs_svc_api_info { /* * APRV2 service IDs for the individual static services. * * @values * - APRV2_IDS_SERVICE_ID_ADSP_CORE_V * - APRV2_IDS_SERVICE_ID_ADSP_AFE_V * - APRV2_IDS_SERVICE_ID_ADSP_ASM_V * - APRV2_IDS_SERVICE_ID_ADSP_ADM_V * - APRV2_IDS_SERVICE_ID_ADSP_MVM_V * - APRV2_IDS_SERVICE_ID_ADSP_CVS_V * - APRV2_IDS_SERVICE_ID_ADSP_CVP_V * - APRV2_IDS_SERVICE_ID_ADSP_LSM_V */ uint32_t service_id; /* * Indicates the API version of the service. * * Each new API update that warrants a change on the HLOS side triggers * an increment in the version. */ uint32_t api_version; /* * Indicates the API increments on a sub-branch (not on the mainline). * * API branch version numbers can increment independently on different * sub-branches. */ uint32_t api_branch_version; }; struct avcs_fwk_ver_info { struct avcs_get_fwk_version avcs_build; struct avs_svc_api_info services[0]; }; /* LSM Specific */ #define VW_FEAT_DIM (39) #define APRV2_IDS_SERVICE_ID_ADSP_LSM_V (0xD) #define APRV2_IDS_DOMAIN_ID_ADSP_V (0x4) #define APRV2_IDS_DOMAIN_ID_APPS_V (0x5) #define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS (0x00012A7F) #define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS (0x00012A80) #define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS (0x00012A81) #define LSM_SESSION_CMD_OPEN_TX (0x00012A82) #define LSM_SESSION_CMD_CLOSE_TX (0x00012A88) #define LSM_SESSION_CMD_SET_PARAMS (0x00012A83) #define LSM_SESSION_CMD_SET_PARAMS_V2 (0x00012A8F) #define LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x00012A84) #define LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x00012A85) #define LSM_SESSION_CMD_START (0x00012A86) #define LSM_SESSION_CMD_STOP (0x00012A87) #define LSM_SESSION_CMD_EOB (0x00012A89) #define LSM_SESSION_CMD_READ (0x00012A8A) #define LSM_SESSION_CMD_OPEN_TX_V2 (0x00012A8B) #define LSM_CMD_ADD_TOPOLOGIES (0x00012A8C) #define LSM_SESSION_EVENT_DETECTION_STATUS (0x00012B00) #define LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x00012B01) #define LSM_DATA_EVENT_READ_DONE (0x00012B02) #define LSM_DATA_EVENT_STATUS (0x00012B03) #define LSM_SESSION_EVENT_DETECTION_STATUS_V3 (0x00012B04) #define LSM_MODULE_ID_VOICE_WAKEUP (0x00012C00) #define LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01) #define LSM_PARAM_ID_OPERATION_MODE (0x00012C02) #define LSM_PARAM_ID_GAIN (0x00012C03) #define LSM_PARAM_ID_CONNECT_TO_PORT (0x00012C04) #define LSM_PARAM_ID_FEATURE_COMPENSATION_DATA (0x00012C07) #define LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS (0x00012C07) #define LSM_MODULE_ID_LAB (0x00012C08) #define LSM_PARAM_ID_LAB_ENABLE (0x00012C09) #define LSM_PARAM_ID_LAB_CONFIG (0x00012C0A) #define LSM_MODULE_ID_FRAMEWORK (0x00012C0E) #define LSM_PARAM_ID_SWMAD_CFG (0x00012C18) #define LSM_PARAM_ID_SWMAD_MODEL (0x00012C19) #define LSM_PARAM_ID_SWMAD_ENABLE (0x00012C1A) #define LSM_PARAM_ID_POLLING_ENABLE (0x00012C1B) #define LSM_PARAM_ID_MEDIA_FMT (0x00012C1E) #define LSM_PARAM_ID_FWK_MODE_CONFIG (0x00012C27) /* HW MAD specific */ #define AFE_MODULE_HW_MAD (0x00010230) #define AFE_PARAM_ID_HW_MAD_CFG (0x00010231) #define AFE_PARAM_ID_HW_MAD_CTRL (0x00010232) #define AFE_PARAM_ID_SLIMBUS_SLAVE_PORT_CFG (0x00010233) /* SW MAD specific */ #define AFE_MODULE_SW_MAD (0x0001022D) #define AFE_PARAM_ID_SW_MAD_CFG (0x0001022E) #define AFE_PARAM_ID_SVM_MODEL (0x0001022F) /* Commands/Params to pass the codec/slimbus data to DSP */ #define AFE_SVC_CMD_SET_PARAM (0x000100f3) #define AFE_MODULE_CDC_DEV_CFG (0x00010234) #define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG (0x00010235) #define AFE_PARAM_ID_CDC_REG_CFG (0x00010236) #define AFE_PARAM_ID_CDC_REG_CFG_INIT (0x00010237) #define AFE_PARAM_ID_CDC_REG_PAGE_CFG (0x00010296) #define AFE_MAX_CDC_REGISTERS_TO_CONFIG (20) /* AANC Port Config Specific */ #define AFE_PARAM_ID_AANC_PORT_CONFIG (0x00010215) #define AFE_API_VERSION_AANC_PORT_CONFIG (0x1) #define AANC_TX_MIC_UNUSED (0) #define AANC_TX_VOICE_MIC (1) #define AANC_TX_ERROR_MIC (2) #define AANC_TX_NOISE_MIC (3) #define AFE_PORT_MAX_CHANNEL_CNT (8) #define AFE_MODULE_AANC (0x00010214) #define AFE_PARAM_ID_CDC_AANC_VERSION (0x0001023A) #define AFE_API_VERSION_CDC_AANC_VERSION (0x1) #define AANC_HW_BLOCK_VERSION_1 (1) #define AANC_HW_BLOCK_VERSION_2 (2) /*Clip bank selection*/ #define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1 #define AFE_CLIP_MAX_BANKS 4 #define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242 struct afe_param_aanc_port_cfg { /* Minor version used for tracking the version of the module's * source port configuration. */ uint32_t aanc_port_cfg_minor_version; /* Sampling rate of the source Tx port. 8k - 192k*/ uint32_t tx_port_sample_rate; /* Channel mapping for the Tx port signal carrying Noise (X), * Error (E), and Voice (V) signals. */ uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT]; /* Number of channels on the source Tx port. */ uint16_t tx_port_num_channels; /* Port ID of the Rx path reference signal. */ uint16_t rx_path_ref_port_id; /* Sampling rate of the reference port. 8k - 192k*/ uint32_t ref_port_sample_rate; } __packed; struct afe_param_id_cdc_aanc_version { /* Minor version used for tracking the version of the module's * hw version */ uint32_t cdc_aanc_minor_version; /* HW version. */ uint32_t aanc_hw_version; } __packed; struct afe_param_id_clip_bank_sel { /* Minor version used for tracking the version of the module's * hw version */ uint32_t minor_version; /* Number of banks to be read */ uint32_t num_banks; uint32_t bank_map[AFE_CLIP_MAX_BANKS]; } __packed; /* ERROR CODES */ /* Success. The operation completed with no errors. */ #define ADSP_EOK 0x00000000 /* General failure. */ #define ADSP_EFAILED 0x00000001 /* Bad operation parameter. */ #define ADSP_EBADPARAM 0x00000002 /* Unsupported routine or operation. */ #define ADSP_EUNSUPPORTED 0x00000003 /* Unsupported version. */ #define ADSP_EVERSION 0x00000004 /* Unexpected problem encountered. */ #define ADSP_EUNEXPECTED 0x00000005 /* Unhandled problem occurred. */ #define ADSP_EPANIC 0x00000006 /* Unable to allocate resource. */ #define ADSP_ENORESOURCE 0x00000007 /* Invalid handle. */ #define ADSP_EHANDLE 0x00000008 /* Operation is already processed. */ #define ADSP_EALREADY 0x00000009 /* Operation is not ready to be processed. */ #define ADSP_ENOTREADY 0x0000000A /* Operation is pending completion. */ #define ADSP_EPENDING 0x0000000B /* Operation could not be accepted or processed. */ #define ADSP_EBUSY 0x0000000C /* Operation aborted due to an error. */ #define ADSP_EABORTED 0x0000000D /* Operation preempted by a higher priority. */ #define ADSP_EPREEMPTED 0x0000000E /* Operation requests intervention to complete. */ #define ADSP_ECONTINUE 0x0000000F /* Operation requests immediate intervention to complete. */ #define ADSP_EIMMEDIATE 0x00000010 /* Operation is not implemented. */ #define ADSP_ENOTIMPL 0x00000011 /* Operation needs more data or resources. */ #define ADSP_ENEEDMORE 0x00000012 /* Operation does not have memory. */ #define ADSP_ENOMEMORY 0x00000014 /* Item does not exist. */ #define ADSP_ENOTEXIST 0x00000015 /* Max count for adsp error code sent to HLOS*/ #define ADSP_ERR_MAX (ADSP_ENOTEXIST + 1) /* Operation is finished. */ #define ADSP_ETERMINATED 0x00011174 /*bharath, adsp_error_codes.h */ /* LPASS clock for I2S Interface */ /* Supported OSR clock values */ #define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ 0xBB8000 #define Q6AFE_LPASS_OSR_CLK_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 Primary PCM IBIT */ #define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200 /* Clock ID for Primary PCM EBIT */ #define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201 /* Clock ID for Secondary PCM IBIT */ #define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202 /* Clock ID for Secondary PCM EBIT */ #define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203 /* Clock ID for Tertiary PCM IBIT */ #define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204 /* Clock ID for Tertiary PCM EBIT */ #define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205 /* Clock ID for Quartery PCM IBIT */ #define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206 /* Clock ID for Quartery PCM EBIT */ #define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207 /** Clock ID for Primary TDM IBIT */ #define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200 /** Clock ID for Primary TDM EBIT */ #define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201 /** Clock ID for Secondary TDM IBIT */ #define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202 /** Clock ID for Secondary TDM EBIT */ #define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203 /** Clock ID for Tertiary TDM IBIT */ #define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204 /** Clock ID for Tertiary TDM EBIT */ #define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205 /** Clock ID for Quartery TDM IBIT */ #define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206 /** Clock ID for Quartery TDM EBIT */ #define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207 /* Clock ID for MCLK1 */ #define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300 /* Clock ID for MCLK2 */ #define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301 /* Clock ID for MCLK3 */ #define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302 /* Clock ID for Internal Digital Codec Core */ #define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303 /* Clock ID for AHB HDMI input */ #define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT 0x400 /* Clock ID for SPDIF core */ #define Q6AFE_LPASS_CLK_ID_SPDIF_CORE 0x500 /* Clock attribute for invalid use (reserved for internal usage) */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0 /* Clock attribute for no couple case */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1 /* Clock attribute for dividend couple case */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2 /* Clock attribute for divisor couple case */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3 /* Clock attribute for invert and no couple case */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4 /* Clock set API version */ #define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1 struct afe_clk_set { /* * Minor version used for tracking clock set. * @values #AFE_API_VERSION_CLOCK_SET */ uint32_t clk_set_minor_version; /* * Clock ID * @values * - 0x100 to 0x10A - MSM8996 * - 0x200 to 0x207 - MSM8996 * - 0x300 to 0x302 - MSM8996 @tablebulletend */ uint32_t clk_id; /* * Clock frequency (in Hertz) to be set. * @values * - >= 0 for clock frequency to set @tablebulletend */ uint32_t clk_freq_in_hz; /* Use to specific divider for two clocks if needed. * Set to Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO for no divider * relation clocks * @values * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR @tablebulletend */ uint16_t clk_attri; /* * Specifies the root clock source. * Currently, only Q6AFE_LPASS_CLK_ROOT_DEFAULT is valid * @values * - 0 @tablebulletend */ uint16_t clk_root; /* * for enable and disable clock. * "clk_freq_in_hz", "clk_attri", and "clk_root" * are ignored in disable clock case. * @values  * - 0 -- Disabled * - 1 -- Enabled @tablebulletend */ uint32_t enable; }; struct afe_clk_cfg { /* Minor version used for tracking the version of the I2S * configuration interface. * Supported values: #AFE_API_VERSION_I2S_CONFIG */ u32 i2s_cfg_minor_version; /* clk value 1 in MHz. */ u32 clk_val1; /* clk value 2 in MHz. */ u32 clk_val2; /* clk_src * #Q6AFE_LPASS_CLK_SRC_EXTERNAL * #Q6AFE_LPASS_CLK_SRC_INTERNAL */ u16 clk_src; /* clk_root -0 for default */ u16 clk_root; /* clk_set_mode * #Q6AFE_LPASS_MODE_BOTH_INVALID * #Q6AFE_LPASS_MODE_CLK1_VALID * #Q6AFE_LPASS_MODE_CLK2_VALID * #Q6AFE_LPASS_MODE_BOTH_VALID */ u16 clk_set_mode; /* This param id is used to configure I2S clk */ u16 reserved; } __packed; /* This param id is used to configure I2S clk */ #define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238 #define AFE_MODULE_CLOCK_SET 0x0001028F #define AFE_PARAM_ID_CLOCK_SET 0x00010290 struct afe_lpass_clk_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_clk_cfg clk_cfg; } __packed; enum afe_lpass_digital_clk_src { Q6AFE_LPASS_DIGITAL_ROOT_INVALID, Q6AFE_LPASS_DIGITAL_ROOT_PRI_MI2S_OSR, Q6AFE_LPASS_DIGITAL_ROOT_SEC_MI2S_OSR, Q6AFE_LPASS_DIGITAL_ROOT_TER_MI2S_OSR, Q6AFE_LPASS_DIGITAL_ROOT_QUAD_MI2S_OSR, Q6AFE_LPASS_DIGITAL_ROOT_CDC_ROOT_CLK, } __packed; /* This param id is used to configure internal clk */ #define AFE_PARAM_ID_INTERNAL_DIGIATL_CDC_CLK_CONFIG 0x00010239 struct afe_digital_clk_cfg { /* Minor version used for tracking the version of the I2S * configuration interface. * Supported values: #AFE_API_VERSION_I2S_CONFIG */ u32 i2s_cfg_minor_version; /* clk value in MHz. */ u32 clk_val; /* INVALID * PRI_MI2S_OSR * SEC_MI2S_OSR * TER_MI2S_OSR * QUAD_MI2S_OSR * DIGT_CDC_ROOT */ u16 clk_root; /* This field must be set to zero. */ u16 reserved; } __packed; struct afe_lpass_digital_clk_config_command { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_digital_clk_cfg clk_cfg; } __packed; /* * Opcode for AFE to start DTMF. */ #define AFE_PORTS_CMD_DTMF_CTL 0x00010102 /** DTMF payload.*/ struct afe_dtmf_generation_command { struct apr_hdr hdr; /* * Duration of the DTMF tone in ms. * -1 -> continuous, * 0 -> disable */ int64_t duration_in_ms; /* * The DTMF high tone frequency. */ uint16_t high_freq; /* * The DTMF low tone frequency. */ uint16_t low_freq; /* * The DTMF volume setting */ uint16_t gain; /* * The number of ports to enable/disable on. */ uint16_t num_ports; /* * The Destination ports - array . * For DTMF on multiple ports, portIds needs to * be populated numPorts times. */ uint16_t port_ids; /* * variable for 32 bit alignment of APR packet. */ uint16_t reserved; } __packed; enum afe_config_type { AFE_SLIMBUS_SLAVE_PORT_CONFIG, AFE_SLIMBUS_SLAVE_CONFIG, AFE_CDC_REGISTERS_CONFIG, AFE_AANC_VERSION, AFE_CDC_CLIP_REGISTERS_CONFIG, AFE_CLIP_BANK_SEL, AFE_CDC_REGISTER_PAGE_CONFIG, AFE_MAX_CONFIG_TYPES, }; struct afe_param_slimbus_slave_port_cfg { uint32_t minor_version; uint16_t slimbus_dev_id; uint16_t slave_dev_pgd_la; uint16_t slave_dev_intfdev_la; uint16_t bit_width; uint16_t data_format; uint16_t num_channels; uint16_t slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; } __packed; struct afe_param_cdc_slimbus_slave_cfg { uint32_t minor_version; uint32_t device_enum_addr_lsw; uint32_t device_enum_addr_msw; uint16_t tx_slave_port_offset; uint16_t rx_slave_port_offset; } __packed; struct afe_param_cdc_reg_cfg { uint32_t minor_version; uint32_t reg_logical_addr; uint32_t reg_field_type; uint32_t reg_field_bit_mask; uint16_t reg_bit_width; uint16_t reg_offset_scale; } __packed; #define AFE_API_VERSION_CDC_REG_PAGE_CFG 1 enum { AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_0 = 0, AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_1, AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_2, AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_3, }; struct afe_param_cdc_reg_page_cfg { uint32_t minor_version; uint32_t enable; uint32_t proc_id; } __packed; struct afe_param_cdc_reg_cfg_data { uint32_t num_registers; struct afe_param_cdc_reg_cfg *reg_data; } __packed; struct afe_svc_cmd_set_param { uint32_t payload_size; uint32_t payload_address_lsw; uint32_t payload_address_msw; uint32_t mem_map_handle; } __packed; struct afe_svc_param_data { uint32_t module_id; uint32_t param_id; uint16_t param_size; uint16_t reserved; } __packed; struct afe_param_hw_mad_ctrl { uint32_t minor_version; uint16_t mad_type; uint16_t mad_enable; } __packed; struct afe_cmd_hw_mad_ctrl { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_hw_mad_ctrl payload; } __packed; struct afe_cmd_hw_mad_slimbus_slave_port_cfg { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; struct afe_param_slimbus_slave_port_cfg sb_port_cfg; } __packed; struct afe_cmd_sw_mad_enable { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; } __packed; struct afe_param_cdc_reg_cfg_payload { struct afe_svc_param_data common; struct afe_param_cdc_reg_cfg reg_cfg; } __packed; struct afe_lpass_clk_config_command_v2 { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_svc_param_data pdata; struct afe_clk_set clk_cfg; } __packed; /* * reg_data's size can be up to AFE_MAX_CDC_REGISTERS_TO_CONFIG */ struct afe_svc_cmd_cdc_reg_cfg { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_param_cdc_reg_cfg_payload reg_data[0]; } __packed; struct afe_svc_cmd_init_cdc_reg_cfg { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 init; } __packed; struct afe_svc_cmd_sb_slave_cfg { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 pdata; struct afe_param_cdc_slimbus_slave_cfg sb_slave_cfg; } __packed; struct afe_svc_cmd_cdc_reg_page_cfg { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 pdata; struct afe_param_cdc_reg_page_cfg cdc_reg_page_cfg; } __packed; struct afe_svc_cmd_cdc_aanc_version { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 pdata; struct afe_param_id_cdc_aanc_version version; } __packed; struct afe_port_cmd_set_aanc_param { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; struct afe_port_param_data_v2 pdata; union { struct afe_param_aanc_port_cfg aanc_port_cfg; struct afe_mod_enable_param mod_enable; } __packed data; } __packed; struct afe_port_cmd_set_aanc_acdb_table { struct apr_hdr hdr; struct afe_port_cmd_set_param_v2 param; } __packed; /* Dolby DAP topology */ #define DOLBY_ADM_COPP_TOPOLOGY_ID 0x0001033B #define DS2_ADM_COPP_TOPOLOGY_ID 0x1301033B /* RMS value from DSP */ #define RMS_MODULEID_APPI_PASSTHRU 0x10009011 #define RMS_PARAM_FIRST_SAMPLE 0x10009012 #define RMS_PAYLOAD_LEN 4 /* Customized mixing in matix mixer */ #define MTMX_MODULE_ID_DEFAULT_CHMIXER 0x00010341 #define DEFAULT_CHMIXER_PARAM_ID_COEFF 0x00010342 #define CUSTOM_STEREO_PAYLOAD_SIZE 9 #define CUSTOM_STEREO_CMD_PARAM_SIZE 24 #define CUSTOM_STEREO_NUM_OUT_CH 0x0002 #define CUSTOM_STEREO_NUM_IN_CH 0x0002 #define CUSTOM_STEREO_INDEX_PARAM 0x0002 #define Q14_GAIN_ZERO_POINT_FIVE 0x2000 #define Q14_GAIN_UNITY 0x4000 struct afe_svc_cmd_set_clip_bank_selection { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 pdata; struct afe_param_id_clip_bank_sel bank_sel; } __packed; /* Ultrasound supported formats */ #define US_POINT_EPOS_FORMAT_V2 0x0001272D #define US_RAW_FORMAT_V2 0x0001272C #define US_PROX_FORMAT_V4 0x0001273B #define US_RAW_SYNC_FORMAT 0x0001272F #define US_GES_SYNC_FORMAT 0x00012730 #define AFE_MODULE_GROUP_DEVICE 0x00010254 #define AFE_PARAM_ID_GROUP_DEVICE_CFG 0x00010255 #define AFE_PARAM_ID_GROUP_DEVICE_ENABLE 0x00010256 #define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX 0x1102 /* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_CFG * parameter, which configures max of 8 AFE ports * into a group. * The fixed size of this structure is sixteen bytes. */ struct afe_group_device_group_cfg { u32 minor_version; u16 group_id; u16 num_channels; u16 port_id[8]; } __packed; #define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX \ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x100) #define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX \ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x100) #define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX \ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x100) #define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX \ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x100) #define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX \ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x100) #define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX \ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x100) #define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX \ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x100) #define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX \ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x100) /** ID of the parameter used by #AFE_MODULE_GROUP_DEVICE to configure the group device. #AFE_SVC_CMD_SET_PARAM can use this parameter ID. Requirements: - Configure the group before the member ports in the group are configured and started. - Enable the group only after it is configured. - Stop all member ports in the group before disabling the group. */ #define AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG 0x0001029E /** Version information used to handle future additions to AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG processing (for backward compatibility). */ #define AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG 0x1 /** Number of AFE ports in group device */ #define AFE_GROUP_DEVICE_NUM_PORTS 8 /* Payload of the AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG parameter ID used by AFE_MODULE_GROUP_DEVICE. */ struct afe_param_id_group_device_tdm_cfg { u32 group_device_cfg_minor_version; /**< Minor version used to track group device configuration. @values #AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG */ u16 group_id; /**< ID for the group device. @values - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX */ u16 reserved; /** 0 */ u16 port_id[AFE_GROUP_DEVICE_NUM_PORTS]; /**< Array of member port IDs of this group. @values - #AFE_PORT_ID_PRIMARY_TDM_RX - #AFE_PORT_ID_PRIMARY_TDM_RX_1 - #AFE_PORT_ID_PRIMARY_TDM_RX_2 - #AFE_PORT_ID_PRIMARY_TDM_RX_3 - #AFE_PORT_ID_PRIMARY_TDM_RX_4 - #AFE_PORT_ID_PRIMARY_TDM_RX_5 - #AFE_PORT_ID_PRIMARY_TDM_RX_6 - #AFE_PORT_ID_PRIMARY_TDM_RX_7 - #AFE_PORT_ID_PRIMARY_TDM_TX - #AFE_PORT_ID_PRIMARY_TDM_TX_1 - #AFE_PORT_ID_PRIMARY_TDM_TX_2 - #AFE_PORT_ID_PRIMARY_TDM_TX_3 - #AFE_PORT_ID_PRIMARY_TDM_TX_4 - #AFE_PORT_ID_PRIMARY_TDM_TX_5 - #AFE_PORT_ID_PRIMARY_TDM_TX_6 - #AFE_PORT_ID_PRIMARY_TDM_TX_7 - #AFE_PORT_ID_SECONDARY_TDM_RX - #AFE_PORT_ID_SECONDARY_TDM_RX_1 - #AFE_PORT_ID_SECONDARY_TDM_RX_2 - #AFE_PORT_ID_SECONDARY_TDM_RX_3 - #AFE_PORT_ID_SECONDARY_TDM_RX_4 - #AFE_PORT_ID_SECONDARY_TDM_RX_5 - #AFE_PORT_ID_SECONDARY_TDM_RX_6 - #AFE_PORT_ID_SECONDARY_TDM_RX_7 - #AFE_PORT_ID_SECONDARY_TDM_TX - #AFE_PORT_ID_SECONDARY_TDM_TX_1 - #AFE_PORT_ID_SECONDARY_TDM_TX_2 - #AFE_PORT_ID_SECONDARY_TDM_TX_3 - #AFE_PORT_ID_SECONDARY_TDM_TX_4 - #AFE_PORT_ID_SECONDARY_TDM_TX_5 - #AFE_PORT_ID_SECONDARY_TDM_TX_6 - #AFE_PORT_ID_SECONDARY_TDM_TX_7 - #AFE_PORT_ID_TERTIARY_TDM_RX - #AFE_PORT_ID_TERTIARY_TDM_RX_1 - #AFE_PORT_ID_TERTIARY_TDM_RX_2 - #AFE_PORT_ID_TERTIARY_TDM_RX_3 - #AFE_PORT_ID_TERTIARY_TDM_RX_4 - #AFE_PORT_ID_TERTIARY_TDM_RX_5 - #AFE_PORT_ID_TERTIARY_TDM_RX_6 - #AFE_PORT_ID_TERTIARY_TDM_RX_7 - #AFE_PORT_ID_TERTIARY_TDM_TX - #AFE_PORT_ID_TERTIARY_TDM_TX_1 - #AFE_PORT_ID_TERTIARY_TDM_TX_2 - #AFE_PORT_ID_TERTIARY_TDM_TX_3 - #AFE_PORT_ID_TERTIARY_TDM_TX_4 - #AFE_PORT_ID_TERTIARY_TDM_TX_5 - #AFE_PORT_ID_TERTIARY_TDM_TX_6 - #AFE_PORT_ID_TERTIARY_TDM_TX_7 - #AFE_PORT_ID_QUATERNARY_TDM_RX - #AFE_PORT_ID_QUATERNARY_TDM_RX_1 - #AFE_PORT_ID_QUATERNARY_TDM_RX_2 - #AFE_PORT_ID_QUATERNARY_TDM_RX_3 - #AFE_PORT_ID_QUATERNARY_TDM_RX_4 - #AFE_PORT_ID_QUATERNARY_TDM_RX_5 - #AFE_PORT_ID_QUATERNARY_TDM_RX_6 - #AFE_PORT_ID_QUATERNARY_TDM_RX_7 - #AFE_PORT_ID_QUATERNARY_TDM_TX - #AFE_PORT_ID_QUATERNARY_TDM_TX_1 - #AFE_PORT_ID_QUATERNARY_TDM_TX_2 - #AFE_PORT_ID_QUATERNARY_TDM_TX_3 - #AFE_PORT_ID_QUATERNARY_TDM_TX_4 - #AFE_PORT_ID_QUATERNARY_TDM_TX_5 - #AFE_PORT_ID_QUATERNARY_TDM_TX_6 - #AFE_PORT_ID_QUATERNARY_TDM_TX_7 @tablebulletend */ u32 num_channels; /**< Number of enabled slots for TDM frame. @values 1 to 8 */ u32 sample_rate; /**< Sampling rate of the port. @values - #AFE_PORT_SAMPLE_RATE_8K - #AFE_PORT_SAMPLE_RATE_16K - #AFE_PORT_SAMPLE_RATE_24K - #AFE_PORT_SAMPLE_RATE_32K - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend */ u32 bit_width; /**< Bit width of the sample. @values 16, 24, (32) */ u16 nslots_per_frame; /**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32. @values 1 - 32 */ u16 slot_width; /**< Slot width of the slot in a TDM frame. (slot_width >= bit_width) have to be satisfied. @values 16, 24, 32 */ u32 slot_mask; /**< Position of active slots. When that bit is set, that paricular slot is active. Number of active slots can be inferred by number of bits set in the mask. Only 8 individual bits can be enabled. Bits 0..31 corresponding to slot 0..31 @values 1 to 2^32 -1 */ } __packed; /* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_ENABLE * parameter, which enables or * disables any module. * The fixed size of this structure is four bytes. */ struct afe_group_device_enable { u16 group_id; /* valid value is AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX */ u16 enable; /* Enables (1) or disables (0) the module. */ } __packed; union afe_port_group_config { struct afe_group_device_group_cfg group_cfg; struct afe_group_device_enable group_enable; struct afe_param_id_group_device_tdm_cfg tdm_cfg; } __packed; struct afe_port_group_create { struct apr_hdr hdr; struct afe_svc_cmd_set_param param; struct afe_port_param_data_v2 pdata; union afe_port_group_config data; } __packed; /* Command for Matrix or Stream Router */ #define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2 0x00010DCE /* Module for AVSYNC */ #define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC 0x00010DC6 /* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the * render window start value. This parameter is supported only for a Set * command (not a Get command) in the Rx direction * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). * Render window start is a value (session time minus timestamp, or ST-TS) * below which frames are held, and after which frames are immediately * rendered. */ #define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1 /* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the * render window end value. This parameter is supported only for a Set * command (not a Get command) in the Rx direction * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value * (session time minus timestamp) above which frames are dropped, and below * which frames are immediately rendered. */ #define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 0x00010DD2 /* Generic payload of the window parameters in the * #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module. * This payload is supported only for a Set command * (not a Get command) on the Rx path. */ struct asm_session_mtmx_strtr_param_window_v2_t { u32 window_lsw; /* Lower 32 bits of the render window start value. */ u32 window_msw; /* Upper 32 bits of the render window start value. * The 64-bit number formed by window_lsw and window_msw specifies a * signed 64-bit window value in microseconds. The sign extension is * necessary. This value is used by the following parameter IDs: * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2 * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2 * The value depends on which parameter ID is used. * The aDSP honors the windows at a granularity of 1 ms. */ }; struct asm_session_cmd_set_mtmx_strstr_params_v2 { uint32_t data_payload_addr_lsw; /* Lower 32 bits of the 64-bit data payload address. */ uint32_t data_payload_addr_msw; /* Upper 32 bits of the 64-bit data payload address. * If the address is not sent (NULL), the message is in the payload. * If the address is sent (non-NULL), the parameter data payloads * begin at the specified address. */ uint32_t mem_map_handle; /* Unique identifier for an address. This memory map handle is returned * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. * values * - NULL -- Parameter data payloads are within the message payload * (in-band). * - Non-NULL -- Parameter data payloads begin at the address specified * in the data_payload_addr_lsw and data_payload_addr_msw fields * (out-of-band). */ uint32_t data_payload_size; /* Actual size of the variable payload accompanying the message, or in * shared memory. This field is used for parsing the parameter payload. * values > 0 bytes */ uint32_t direction; /* Direction of the entity (matrix mixer or stream router) on which * the parameter is to be set. * values * - 0 -- Rx (for Rx stream router or Rx matrix mixer) * - 1 -- Tx (for Tx stream router or Tx matrix mixer) */ }; struct asm_mtmx_strtr_params { struct apr_hdr hdr; struct asm_session_cmd_set_mtmx_strstr_params_v2 param; struct asm_stream_param_data_v2 data; u32 window_lsw; u32 window_msw; } __packed; #define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF #define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0 #define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B #define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL) struct asm_session_cmd_get_mtmx_strstr_params_v2 { uint32_t data_payload_addr_lsw; /* Lower 32 bits of the 64-bit data payload address. */ uint32_t data_payload_addr_msw; /* * Upper 32 bits of the 64-bit data payload address. * If the address is not sent (NULL), the message is in the payload. * If the address is sent (non-NULL), the parameter data payloads * begin at the specified address. */ uint32_t mem_map_handle; /* * Unique identifier for an address. This memory map handle is returned * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. * values * - NULL -- Parameter data payloads are within the message payload * (in-band). * - Non-NULL -- Parameter data payloads begin at the address specified * in the data_payload_addr_lsw and data_payload_addr_msw fields * (out-of-band). */ uint32_t direction; /* * Direction of the entity (matrix mixer or stream router) on which * the parameter is to be set. * values * - 0 -- Rx (for Rx stream router or Rx matrix mixer) * - 1 -- Tx (for Tx stream router or Tx matrix mixer) */ uint32_t module_id; /* Unique module ID. */ uint32_t param_id; /* Unique parameter ID. */ uint32_t param_max_size; }; struct asm_session_mtmx_strtr_param_session_time_v3_t { uint32_t session_time_lsw; /* Lower 32 bits of the current session time in microseconds */ uint32_t session_time_msw; /* * Upper 32 bits of the current session time in microseconds. * The 64-bit number formed by session_time_lsw and session_time_msw * is treated as signed. */ uint32_t absolute_time_lsw; /* * Lower 32 bits of the 64-bit absolute time in microseconds. * This is the time when the sample corresponding to the * session_time_lsw is rendered to the hardware. This absolute * time can be slightly in the future or past. */ uint32_t absolute_time_msw; /* * Upper 32 bits of the 64-bit absolute time in microseconds. * This is the time when the sample corresponding to the * session_time_msw is rendered to hardware. This absolute * time can be slightly in the future or past. The 64-bit number * formed by absolute_time_lsw and absolute_time_msw is treated as * unsigned. */ uint32_t time_stamp_lsw; /* Lower 32 bits of the last processed timestamp in microseconds */ uint32_t time_stamp_msw; /* * Upper 32 bits of the last processed timestamp in microseconds. * The 64-bit number formed by time_stamp_lsw and time_stamp_lsw * is treated as unsigned. */ uint32_t flags; /* * Keeps track of any additional flags needed. * @values{for bit 31} * - 0 -- Uninitialized/invalid * - 1 -- Valid * All other bits are reserved; clients must set them to zero. */ }; union asm_session_mtmx_strtr_data_type { struct asm_session_mtmx_strtr_param_session_time_v3_t session_time; }; struct asm_mtmx_strtr_get_params { struct apr_hdr hdr; struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info; } __packed; struct asm_mtmx_strtr_get_params_cmdrsp { uint32_t err_code; struct asm_stream_param_data_v2 param_info; union asm_session_mtmx_strtr_data_type param_data; } __packed; #define AUDPROC_MODULE_ID_RESAMPLER 0x00010719 enum { LEGACY_PCM = 0, COMPRESSED_PASSTHROUGH, COMPRESSED_PASSTHROUGH_CONVERT, LISTEN, }; #define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770 #define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771 struct adm_set_compressed_device_mute { struct adm_cmd_set_pp_params_v5 command; struct adm_param_data_v5 params; u32 mute_on; } __packed; #define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E #define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F struct adm_set_compressed_device_latency { struct adm_cmd_set_pp_params_v5 command; struct adm_param_data_v5 params; u32 latency; } __packed; #define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6 #define VOICEPROC_PARAM_ID_FLUENCE_SOUNDFOCUS 0x00010E37 #define VOICEPROC_PARAM_ID_FLUENCE_SOURCETRACKING 0x00010E38 #define MAX_SECTORS 8 #define MAX_NOISE_SOURCE_INDICATORS 3 #define MAX_POLAR_ACTIVITY_INDICATORS 360 struct sound_focus_param { uint16_t start_angle[MAX_SECTORS]; uint8_t enable[MAX_SECTORS]; uint16_t gain_step; } __packed; struct source_tracking_param { uint8_t vad[MAX_SECTORS]; uint16_t doa_speech; uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS]; uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS]; } __packed; struct adm_param_fluence_soundfocus_t { uint16_t start_angles[MAX_SECTORS]; uint8_t enables[MAX_SECTORS]; uint16_t gain_step; uint16_t reserved; } __packed; struct adm_set_fluence_soundfocus_param { struct adm_cmd_set_pp_params_v5 params; struct adm_param_data_v5 data; struct adm_param_fluence_soundfocus_t soundfocus_data; } __packed; struct adm_param_fluence_sourcetracking_t { uint8_t vad[MAX_SECTORS]; uint16_t doa_speech; uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS]; uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS]; } __packed; #define AUDPROC_MODULE_ID_AUDIOSPHERE 0x00010916 #define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE 0x00010917 #define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH 0x00010918 #define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE 0x00010919 #define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT 0x0001091A #define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT 0x0001091B #define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT 0x0001091C #define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT 0x0001091D #define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO 0x0001091E #define AUDPROC_MODULE_ID_VOICE_TX_SECNS 0x10027059 #define AUDPROC_PARAM_IDX_SEC_PRIMARY_MIC_CH 0x10014444 struct admx_sec_primary_mic_ch { uint16_t version; /*version number*/ uint16_t reserved; uint16_t sec_primary_mic_ch; /*