diff options
Diffstat (limited to 'include/sound')
31 files changed, 17249 insertions, 16 deletions
diff --git a/include/sound/adsp_err.h b/include/sound/adsp_err.h new file mode 100644 index 000000000000..43be91d6ba8f --- /dev/null +++ b/include/sound/adsp_err.h @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __ADSP_ERR__ +#define __ADSP_ERR__ + +int adsp_err_get_lnx_err_code(u32 adsp_error); + +char *adsp_err_get_err_str(u32 adsp_error); + +#endif diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h new file mode 100644 index 000000000000..f312284024a9 --- /dev/null +++ b/include/sound/apr_audio-v2.h @@ -0,0 +1,11742 @@ +/* Copyright (c) 2012-2018, 2020, The Linux Foundation. All rights reserved. +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 and +* only version 2 as published by the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +*/ + + +#ifndef _APR_AUDIO_V2_H_ +#define _APR_AUDIO_V2_H_ + +#include <linux/qdsp6v2/apr.h> +#include <linux/msm_audio.h> + +/* size of header needed for passing data out of band */ +#define APR_CMD_OB_HDR_SZ 12 + +/* size of header needed for getting data */ +#define APR_CMD_GET_HDR_SZ 16 + +struct param_outband { + size_t size; + void *kvaddr; + phys_addr_t paddr; +}; + +/* --------- Common Structures and Definitions------------- */ +/* Instance ID Definitions */ +#define INSTANCE_ID_0 0x0000 + +struct mem_mapping_hdr { + /* + * LSW of parameter data payload address. Supported values: any. + * - Must be set to zero for in-band data. + */ + u32 data_payload_addr_lsw; + + /* + * MSW of Parameter data payload address. Supported values: any. + * - Must be set to zero for in-band data. + * - In the case of 32 bit Shared memory address, msw field must be + * set to zero. + * - In the case of 36 bit shared memory address, bit 31 to bit 4 of + * msw must be set to zero. + */ + u32 data_payload_addr_msw; + + /* + * Memory map handle returned by DSP through + * ASM_CMD_SHARED_MEM_MAP_REGIONS command. + * Supported Values: Any. + * If 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 mem_map_handle; + +} __packed; + +/* + * Payload format for parameter data. + * Immediately following these structures are param_size bytes of parameter + * data. + */ +struct param_hdr_v1 { + /* Valid ID of the module. */ + uint32_t module_id; + + /* Valid ID of the parameter. */ + uint32_t param_id; + + /* The size of the parameter specified by the module/param ID combo */ + uint16_t param_size; + + /* This field must be set to zero. */ + uint16_t reserved; +} __packed; + +struct param_hdr_v2 { + /* Valid ID of the module. */ + uint32_t module_id; + + /* Valid ID of the parameter. */ + uint32_t param_id; + + /* The size of the parameter specified by the module/param ID combo */ + uint32_t param_size; +} __packed; + +struct param_hdr_v3 { + /* Valid ID of the module. */ + uint32_t module_id; + + /* Instance of the module. */ + uint16_t instance_id; + + /* This field must be set to zero. */ + uint16_t reserved; + + /* Valid ID of the parameter. */ + uint32_t param_id; + + /* The size of the parameter specified by the module/param ID combo */ + uint32_t param_size; +} __packed; + +/* A union of all param_hdr versions for versitility and max size */ +union param_hdrs { + struct param_hdr_v1 v1; + struct param_hdr_v2 v2; + struct param_hdr_v3 v3; +}; + +struct module_instance_info { + /* Module ID. */ + u32 module_id; + + /* Instance of the module */ + u16 instance_id; + + /* Reserved. This field must be set to zero. */ + u16 reserved; +} __packed; +/* -------------------------------------------------------- */ + +/* Begin service specific definitions and structures */ + +#define ADSP_ADM_VERSION 0x00070000 +#define ADSP_ASM_API_VERSION_V2 2 +#define ADSP_ADM_API_VERSION_V3 3 +#define ADSP_AFE_API_VERSION_V3 3 + +#define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322 +#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323 +#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324 + +#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325 +#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D +/* Enumeration for an audio Rx matrix ID.*/ +#define ADM_MATRIX_ID_AUDIO_RX 0 + +#define ADM_MATRIX_ID_AUDIO_TX 1 + +#define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX 2 + +#define ADM_MATRIX_ID_COMPRESSED_AUDIO_TX 3 + +#define ADM_MATRIX_ID_LISTEN_TX 4 +/* Enumeration for an audio Tx matrix ID.*/ +#define ADM_MATRIX_ID_AUDIOX 1 + +#define ADM_MAX_COPPS 5 + +/* make sure this matches with msm_audio_calibration */ +#define SP_V2_NUM_MAX_SPKR 2 + +/* Session map node structure. +* Immediately following this structure are num_copps +* entries of COPP IDs. The COPP IDs are 16 bits, so +* there might be a padding 16-bit field if num_copps +* is odd. +*/ +struct adm_session_map_node_v5 { + u16 session_id; +/* Handle of the ASM session to be routed. Supported values: 1 +* to 8. +*/ + + + u16 num_copps; + /* Number of COPPs to which this session is to be routed. + Supported values: 0 < num_copps <= ADM_MAX_COPPS. + */ +} __packed; + +/* Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command. +* Immediately following this structure are num_sessions of the session map +* node payload (adm_session_map_node_v5). +*/ + +struct adm_cmd_matrix_map_routings_v5 { + struct apr_hdr hdr; + + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx +* (1). Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX +* macros to set this field. +*/ + u32 num_sessions; + /* Number of sessions being updated by this command (optional).*/ +} __packed; + +/* This command allows a client to open a COPP/Voice Proc. TX module +* and sets up the device session: Matrix -> COPP -> AFE on the RX +* and AFE -> COPP -> Matrix on the TX. This enables PCM data to +* be transferred to/from the endpoint (AFEPortID). +* +* @return +* #ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and +* COPP ID. +*/ +#define ADM_CMD_DEVICE_OPEN_V5 0x00010326 + +/* This command allows a client to open a COPP/Voice Proc the +* way as ADM_CMD_DEVICE_OPEN_V5 but supports multiple endpoint2 +* channels. +* +* @return +* #ADM_CMDRSP_DEVICE_OPEN_V6 with the resulting status and +* COPP ID. +*/ +#define ADM_CMD_DEVICE_OPEN_V6 0x00010356 + +/* This command allows a client to open a COPP/Voice Proc the +* way as ADM_CMD_DEVICE_OPEN_V8 but supports any number channel +* of configuration. +* +* @return +* #ADM_CMDRSP_DEVICE_OPEN_V8 with the resulting status and +* COPP ID. +*/ +#define ADM_CMD_DEVICE_OPEN_V8 0x0001036A + +/* Definition for a low latency stream session. */ +#define ADM_LOW_LATENCY_DEVICE_SESSION 0x2000 + +/* Definition for a ultra low latency stream session. */ +#define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION 0x4000 + +/* Definition for a ultra low latency with Post Processing stream session. */ +#define ADM_ULL_POST_PROCESSING_DEVICE_SESSION 0x8000 + +/* Definition for a legacy device session. */ +#define ADM_LEGACY_DEVICE_SESSION 0 + +/* Indicates that endpoint_id_2 is to be ignored.*/ +#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1 + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2 + +#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3 + +/* Indicates that an audio COPP is to send/receive a mono PCM + * stream to/from + * END_POINT_ID_1. + */ +#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1 + +/* Indicates that an audio COPP is to send/receive a + * stereo PCM stream to/from END_POINT_ID_1. + */ +#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2 + +/* Sample rate is 8000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000 + +/* Sample rate is 16000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000 + +/* Sample rate is 48000 Hz.*/ +#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000 + +/* Definition for a COPP live input flag bitmask.*/ +#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U) + +/* Definition for a COPP live shift value bitmask.*/ +#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0 + +/* Definition for the COPP ID bitmask.*/ +#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL) + +/* Definition for the COPP ID shift value.*/ +#define ADM_SHIFT_COPP_ID 0 + +/* Definition for the service ID bitmask.*/ +#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL) + +/* Definition for the service ID shift value.*/ +#define ADM_SHIFT_SERVICE_ID 16 + +/* Definition for the domain ID bitmask.*/ +#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL) + +/* Definition for the domain ID shift value.*/ +#define ADM_SHIFT_DOMAIN_ID 24 + +/* ADM device open command payload of the + #ADM_CMD_DEVICE_OPEN_V5 command. +*/ +struct adm_cmd_device_open_v5 { + struct apr_hdr hdr; + u16 flags; +/* Reserved for future use. Clients must set this field + * to zero. + */ + + u16 mode_of_operation; +/* Specifies whether the COPP must be opened on the Tx or Rx + * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for + * supported values and interpretation. + * Supported values: + * - 0x1 -- Rx path COPP + * - 0x2 -- Tx path live COPP + * - 0x3 -- Tx path nonlive COPP + * Live connections cause sample discarding in the Tx device + * matrix if the destination output ports do not pull them + * fast enough. Nonlive connections queue the samples + * indefinitely. + */ + + u16 endpoint_id_1; +/* Logical and physical endpoint ID of the audio path. + * If the ID is a voice processor Tx block, it receives near + * samples. Supported values: Any pseudoport, AFE Rx port, + * or AFE Tx port For a list of valid IDs, refer to + * @xhyperref{Q4,[Q4]}. + * Q4 = Hexagon Multimedia: AFE Interface Specification + */ + + u16 endpoint_id_2; +/* Logical and physical endpoint ID 2 for a voice processor + * Tx block. + * This is not applicable to audio COPP. + * Supported values: + * - AFE Rx port + * - 0xFFFF -- Endpoint 2 is unavailable and the voice + * processor Tx + * block ignores this endpoint + * When the voice processor Tx block is created on the audio + * record path, + * it can receive far-end samples from an AFE Rx port if the + * voice call + * is active. The ID of the AFE port is provided in this + * field. + * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. + */ + + u32 topology_id; + /* Audio COPP topology ID; 32-bit GUID. */ + + u16 dev_num_channel; +/* Number of channels the audio COPP sends to/receives from + * the endpoint. + * Supported values: 1 to 8. + * The value is ignored for the voice processor Tx block, + * where channel + * configuration is derived from the topology ID. + */ + + u16 bit_width; +/* Bit width (in bits) that the audio COPP sends to/receives + * from the + * endpoint. The value is ignored for the voice processing + * Tx block, + * where the PCM width is 16 bits. + */ + + u32 sample_rate; +/* Sampling rate at which the audio COPP/voice processor + * Tx block + * interfaces with the endpoint. + * Supported values for voice processor Tx: 8000, 16000, + * 48000 Hz + * Supported values for audio COPP: >0 and <=192 kHz + */ + + u8 dev_channel_mapping[8]; +/* Array of channel mapping of buffers that the audio COPP + * sends to the endpoint. Channel[i] mapping describes channel + * I inside the buffer, where 0 < i < dev_num_channel. + * This value is relevant only for an audio Rx COPP. + * For the voice processor block and Tx audio block, this field + * is set to zero and is ignored. + */ +} __packed; + +/* ADM device open command payload of the + * #ADM_CMD_DEVICE_OPEN_V6 command. + */ +struct adm_cmd_device_open_v6 { + struct apr_hdr hdr; + u16 flags; +/* Reserved for future use. Clients must set this field + * to zero. + */ + + u16 mode_of_operation; +/* Specifies whether the COPP must be opened on the Tx or Rx + * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for + * supported values and interpretation. + * Supported values: + * - 0x1 -- Rx path COPP + * - 0x2 -- Tx path live COPP + * - 0x3 -- Tx path nonlive COPP + * Live connections cause sample discarding in the Tx device + * matrix if the destination output ports do not pull them + * fast enough. Nonlive connections queue the samples + * indefinitely. + */ + + u16 endpoint_id_1; +/* Logical and physical endpoint ID of the audio path. + * If the ID is a voice processor Tx block, it receives near + * samples. Supported values: Any pseudoport, AFE Rx port, + * or AFE Tx port For a list of valid IDs, refer to + * @xhyperref{Q4,[Q4]}. + * Q4 = Hexagon Multimedia: AFE Interface Specification + */ + + u16 endpoint_id_2; +/* Logical and physical endpoint ID 2 for a voice processor + * Tx block. + * This is not applicable to audio COPP. + * Supported values: + * - AFE Rx port + * - 0xFFFF -- Endpoint 2 is unavailable and the voice + * processor Tx + * block ignores this endpoint + * When the voice processor Tx block is created on the audio + * record path, + * it can receive far-end samples from an AFE Rx port if the + * voice call + * is active. The ID of the AFE port is provided in this + * field. + * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. + */ + + u32 topology_id; +/* Audio COPP topology ID; 32-bit GUID. */ + + u16 dev_num_channel; +/* Number of channels the audio COPP sends to/receives from + * the endpoint. + * Supported values: 1 to 8. + * The value is ignored for the voice processor Tx block, + * where channel + * configuration is derived from the topology ID. + */ + + u16 bit_width; +/* Bit width (in bits) that the audio COPP sends to/receives + * from the + * endpoint. The value is ignored for the voice processing + * Tx block, + * where the PCM width is 16 bits. + */ + + u32 sample_rate; +/* Sampling rate at which the audio COPP/voice processor + * Tx block + * interfaces with the endpoint. + * Supported values for voice processor Tx: 8000, 16000, + * 48000 Hz + * Supported values for audio COPP: >0 and <=192 kHz + */ + + u8 dev_channel_mapping[8]; +/* Array of channel mapping of buffers that the audio COPP + * sends to the endpoint. Channel[i] mapping describes channel + * I inside the buffer, where 0 < i < dev_num_channel. + * This value is relevant only for an audio Rx COPP. + * For the voice processor block and Tx audio block, this field + * is set to zero and is ignored. + */ + + u16 dev_num_channel_eid2; +/* Number of channels the voice processor block sends + * to/receives from the endpoint2. + * Supported values: 1 to 8. + * The value is ignored for audio COPP or if endpoint_id_2 is + * set to 0xFFFF. + */ + + u16 bit_width_eid2; +/* Bit width (in bits) that the voice processor sends + * to/receives from the endpoint2. + * Supported values: 16 and 24. + * The value is ignored for audio COPP or if endpoint_id_2 is + * set to 0xFFFF. + */ + + u32 sample_rate_eid2; +/* Sampling rate at which the voice processor Tx block + * interfaces with the endpoint2. + * Supported values for Tx voice processor: >0 and <=384 kHz + * The value is ignored for audio COPP or if endpoint_id_2 is + * set to 0xFFFF. + */ + + u8 dev_channel_mapping_eid2[8]; +/* Array of channel mapping of buffers that the voice processor + * sends to the endpoint. Channel[i] mapping describes channel + * I inside the buffer, where 0 < i < dev_num_channel. + * This value is relevant only for the Tx voice processor. + * The values are ignored for audio COPP or if endpoint_id_2 is + * set to 0xFFFF. + */ +} __packed; + +/* ADM device open endpoint payload the +* #ADM_CMD_DEVICE_OPEN_V8 command. +*/ +struct adm_device_endpoint_payload { + u16 dev_num_channel; + /* Number of channels the audio COPP sends to/receives from + * the endpoint. + * Supported values: 1 to 32. + * The value is ignored for the voice processor Tx block, + * where channel + * configuration is derived from the topology ID. + */ + + u16 bit_width; + /* Bit width (in bits) that the audio COPP sends to/receives + * from the + * endpoint. The value is ignored for the voice processing + * Tx block, + * where the PCM width is 16 bits. + */ + + u32 sample_rate; + /* Sampling rate at which the audio COPP/voice processor + * Tx block + * interfaces with the endpoint. + * Supported values for voice processor Tx: 8000, 16000, + * 48000 Hz + * Supported values for audio COPP: >0 and <=192 kHz + */ + + u8 dev_channel_mapping[32]; +} __packed; + +/* ADM device open command payload of the +* #ADM_CMD_DEVICE_OPEN_V8 command. +*/ +struct adm_cmd_device_open_v8 { + struct apr_hdr hdr; + u16 flags; +/* Bit width Native mode enabled : 11th bit of flag parameter +* If 11th bit of flag is set then that means matrix mixer will be +* running in native mode for bit width for this device session. +* +* Channel Native mode enabled : 12th bit of flag parameter +* If 12th bit of flag is set then that means matrix mixer will be +* running in native mode for channel configuration for this device session. +* All other bits are reserved; clients must set them to 0. +**/ + u16 mode_of_operation; +/* Specifies whether the COPP must be opened on the Tx or Rx + * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for + * supported values and interpretation. + * Supported values: + * - 0x1 -- Rx path COPP + * - 0x2 -- Tx path live COPP + * - 0x3 -- Tx path nonlive COPP + * Live connections cause sample discarding in the Tx device + * matrix if the destination output ports do not pull them + * fast enough. Nonlive connections queue the samples + * indefinitely. + */ + u32 topology_id; + /* Audio COPP topology ID; 32-bit GUID. */ + + + u16 endpoint_id_1; +/* Logical and physical endpoint ID of the audio path. + * If the ID is a voice processor Tx block, it receives near + * samples. Supported values: Any pseudoport, AFE Rx port, + * or AFE Tx port For a list of valid IDs, refer to + * @xhyperref{Q4,[Q4]}. + * Q4 = Hexagon Multimedia: AFE Interface Specification + */ + + u16 endpoint_id_2; +/* Logical and physical endpoint ID 2 for a voice processor + * Tx block. + * This is not applicable to audio COPP. + * Supported values: + * - AFE Rx port + * - 0xFFFF -- Endpoint 2 is unavailable and the voice + * processor Tx + * block ignores this endpoint + * When the voice processor Tx block is created on the audio + * record path, + * it can receive far-end samples from an AFE Rx port if the + * voice call + * is active. The ID of the AFE port is provided in this + * field. + * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}. + */ + +/* + * Logical and physical endpoint ID of the audio path. + * This indicated afe rx port in ADM loopback use cases. + * In all other use cases this should be set to 0xffff + */ + u16 endpoint_id_3; + u16 reserved; +} __packed; + +/* + * This command allows the client to close a COPP and disconnect + * the device session. + */ +#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327 + +/* Sets one or more parameters to a COPP. +*/ +#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328 +#define ADM_CMD_SET_PP_PARAMS_V6 0x0001035D + +/* + * Structure of the ADM Set PP Params command. Parameter data must be + * pre-packed with correct header for either V2 or V3 when sent in-band. + * Use q6core_pack_pp_params to pack the header and data correctly depending on + * Instance ID support. + */ +struct adm_cmd_set_pp_params { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* The memory mapping header to be used when sending out of band */ + struct mem_mapping_hdr mem_hdr; + + /* Size in bytes of the variable payload accompanying this + * message or + * in shared memory. This is used for parsing the parameter + * payload. + */ + u32 payload_size; + + /* Parameter data for in band payload. This should be structured as the + * parameter header immediately followed by the parameter data. Multiple + * parameters can be set in one command by repeating the header followed + * by the data for as many parameters as need to be set. + * Use q6core_pack_pp_params to pack the header and data correctly + * depending on Instance ID support. + */ + u8 param_data[0]; +} __packed; + +/* ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 command is used to set + * calibration data to the ADSP Matrix Mixer the payload is + * of struct adm_cmd_set_mtmx_params_v1. + * + * ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1 can be used to get + * the calibration data from the ADSP Matrix Mixer and + * ADM_CMDRSP_GET_MTMX_STRTR_DEV_PARAMS_V1 is the response + * ioctl to ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1. + */ +#define ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 0x00010367 +#define ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1 0x00010368 +#define ADM_CMDRSP_GET_MTMX_STRTR_DEV_PARAMS_V1 0x00010369 + +/* Payload of the #define ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 command. + * If the data_payload_addr_lsw and data_payload_addr_msw element + * are NULL, a series of struct param_hdr_v3 structures immediately + * follows, whose total size is payload_size bytes. + */ +struct adm_cmd_set_mtmx_params_v1 { + 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 it 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; + + /* COPP ID/Device ID */ + u16 copp_id; + + /* For alignment, must be set to 0 */ + u16 reserved; +} __packed; + +struct enable_param_v6 { + /* + * 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. + */ + struct param_hdr_v3 param; + + /* @values 0 : Disable 1: Enable */ + uint32_t enable; +} __packed; + +/* Defined in ADSP as VOICE_MODULE_TX_STREAM_LIMITER but + * used for RX stream limiter on matrix input to ADM. + */ +#define ADM_MTMX_MODULE_STREAM_LIMITER 0x00010F15 + +#define ASM_STREAM_CMD_REGISTER_PP_EVENTS 0x00013213 +#define ASM_STREAM_PP_EVENT 0x00013214 +#define ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE 0x13333 +#define ASM_IEC_61937_MEDIA_FMT_EVENT 0x13334 + +#define DSP_STREAM_CMD "ADSP Stream Cmd" +#define DSP_STREAM_CALLBACK "ADSP Stream Callback Event" +#define DSP_STREAM_CALLBACK_QUEUE_SIZE 1024 + +struct dsp_stream_callback_list { + struct list_head list; + struct msm_adsp_event_data event; +}; + +struct dsp_stream_callback_prtd { + uint16_t event_count; + struct list_head event_queue; + spinlock_t prtd_spin_lock; +}; + +/* set customized mixing on matrix mixer */ +#define ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 0x00010344 +struct adm_cmd_set_pspd_mtmx_strtr_params_v5 { + struct apr_hdr hdr; + /* LSW of parameter data payload address.*/ + u32 payload_addr_lsw; + /* MSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */ + /* command. If mem_map_handle is zero implies the message is in */ + /* the payload */ + u32 mem_map_handle; + /* Size in bytes of the variable payload accompanying this */ + /* message or in shared memory. This is used for parsing the */ + /* parameter payload. */ + u32 payload_size; + u16 direction; + u16 sessionid; + u16 deviceid; + u16 reserved; +} __packed; + +/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. + */ +#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 + +/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message, + * which returns the + * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command. + */ +struct adm_cmd_rsp_device_open_v5 { + u32 status; + /* Status message (error code).*/ + + u16 copp_id; + /* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/ + + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command. + */ +#define ADM_CMDRSP_DEVICE_OPEN_V6 0x00010357 + +/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V8 command. + */ +#define ADM_CMDRSP_DEVICE_OPEN_V8 0x0001036B + +/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message, + * which returns the + * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command + * is the exact same as ADM_CMDRSP_DEVICE_OPEN_V5. + */ + +/* This command allows a query of one COPP parameter. +*/ +#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A +#define ADM_CMD_GET_PP_PARAMS_V6 0x0001035E + +/* + * Structure of the ADM Get PP Params command. Parameter header must be + * packed correctly for either V2 or V3. Use q6core_pack_pp_params to pack the + * header correctly depending on Instance ID support. + */ +struct adm_cmd_get_pp_params { + struct apr_hdr apr_hdr; + + /* The memory mapping header to be used when requesting outband */ + struct mem_mapping_hdr mem_hdr; + + /* Parameter header for in band payload. */ + union param_hdrs param_hdr; +} __packed; + +/* Returns parameter values + * in response to an #ADM_CMD_GET_PP_PARAMS_V5 command. + */ +#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B + +/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message, + * which returns parameter values in response + * to an #ADM_CMD_GET_PP_PARAMS_V5 command. + * Immediately following this + * structure is the param_hdr_v1 + * structure containing the pre/postprocessing + * parameter data. For an in-band + * scenario, the variable payload depends + * on the size of the parameter. +*/ +struct adm_cmd_rsp_get_pp_params_v5 { + /* Status message (error code).*/ + u32 status; + + /* The header that identifies the subsequent parameter data */ + struct param_hdr_v1 param_hdr; + + /* The parameter data returned */ + u32 param_data[0]; +} __packed; + +/* + * Returns parameter values in response to an #ADM_CMD_GET_PP_PARAMS_V5/6 + * command. + */ +#define ADM_CMDRSP_GET_PP_PARAMS_V6 0x0001035F + +/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V6 message, + * which returns parameter values in response + * to an #ADM_CMD_GET_PP_PARAMS_V6 command. + * Immediately following this + * structure is the param_hdr_v3 + * structure containing the pre/postprocessing + * parameter data. For an in-band + * scenario, the variable payload depends + * on the size of the parameter. +*/ +struct adm_cmd_rsp_get_pp_params_v6 { + /* Status message (error code).*/ + u32 status; + + /* The header that identifies the subsequent parameter data */ + struct param_hdr_v3 param_hdr; + + /* The parameter data returned */ + u32 param_data[0]; +} __packed; + +/* Structure for holding soft stepping volume parameters. */ + +/* + * Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * parameters used by the Volume Control module. + */ + +struct audproc_softvolume_params { + u32 period; + u32 step; + u32 rampingcurve; +} __packed; + +/* + * ID of the Media Format Converter (MFC) module. + * This module supports the following parameter IDs: + * #AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT + * #AUDPROC_CHMIXER_PARAM_ID_COEFF + */ +#define AUDPROC_MODULE_ID_MFC 0x00010912 + +/* ID of the Output Media Format parameters used by AUDPROC_MODULE_ID_MFC. + * + */ +#define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913 + +/* ID of the Channel Mixer module, which is used to configure + * channel-mixer related parameters. + * This module supports the AUDPROC_CHMIXER_PARAM_ID_COEFF parameter ID. + */ +#define AUDPROC_MODULE_ID_CHMIXER 0x00010341 + +/* ID of the Coefficient parameter used by AUDPROC_MODULE_ID_CHMIXER to + *configure the channel mixer weighting coefficients. + */ +#define AUDPROC_CHMIXER_PARAM_ID_COEFF 0x00010342 + +/* Payload of the per-session, per-device parameter data of the + * #ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 command or + * #ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V6 command. + * Immediately following this structure are param_size bytes of parameter + * data. The structure and size depend on the module_id/param_id pair. + */ +struct adm_pspd_param_data_t { + uint32_t module_id; + uint32_t param_id; + uint16_t param_size; + uint16_t reserved; +} __packed; + +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; + +struct audproc_mfc_param_media_fmt { + 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 { + /* Linear gain in Q13 format. */ + uint16_t master_gain; + /* Clients must set this field to zero. */ + uint16_t reserved; +} __packed; + +struct audproc_soft_step_volume_params { +/* + * Period in milliseconds. + * Supported values: 0 to 15000 + */ + uint32_t period; +/* + * Step in microseconds. + * Supported values: 0 to 15000000 + */ + uint32_t step; +/* + * Ramping curve type. + * Supported values: + * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LINEAR + * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_EXP + * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LOG + */ + uint32_t ramping_curve; +} __packed; + +struct audproc_enable_param_t { + /* + * Specifies whether the Audio processing module is enabled. + * This parameter is generic/common parameter to configure or + * determine the state of any audio processing module. + + * @values 0 : Disable 1: Enable + */ + uint32_t enable; +}; + +/* + * Allows a client to control the gains on various session-to-COPP paths. + */ +#define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C + +/* + * Allows a client to control the gains on various session-to-COPP paths. + * Maximum support 32 channels + */ +#define ADM_CMD_MATRIX_RAMP_GAINS_V7 0x0001036C + +/* Indicates that the target gain in the + * current adm_session_copp_gain_v5 + * structure is to be applied to all + * the session-to-COPP paths that exist for + * the specified session. + */ +#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF + +/* Indicates that the target gain is + * to be immediately applied to the + * specified session-to-COPP path, + * without a ramping fashion. + */ +#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000 + +/* Enumeration for a linear ramping curve.*/ +#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000 + +/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command. + * Immediately following this structure are num_gains of the + * adm_session_copp_gain_v5structure. + */ +struct adm_cmd_matrix_ramp_gains_v5 { + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). + * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX + * macros to set this field. +*/ + + u16 num_gains; + /* Number of gains being applied. */ + + u16 reserved_for_align; + /* Reserved. This field must be set to zero.*/ +} __packed; + +/* Session-to-COPP path gain structure, used by the + * #ADM_CMD_MATRIX_RAMP_GAINS_V5 command. + * This structure specifies the target + * gain (per channel) that must be applied + * to a particular session-to-COPP path in + * the audio matrix. The structure can + * also be used to apply the gain globally + * to all session-to-COPP paths that + * exist for the given session. + * The aDSP uses device channel mapping to + * determine which channel gains to + * use from this command. For example, + * if the device is configured as stereo, + * the aDSP uses only target_gain_ch_1 and + * target_gain_ch_2, and it ignores + * the others. + */ +struct adm_session_copp_gain_v5 { + u16 session_id; +/* Handle of the ASM session. + * Supported values: 1 to 8. + */ + + u16 copp_id; +/* Handle of the COPP. Gain will be applied on the Session ID + * COPP ID path. + */ + + u16 ramp_duration; +/* Duration (in milliseconds) of the ramp over + * which target gains are + * to be applied. Use + * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE + * to indicate that gain must be applied immediately. + */ + + u16 step_duration; +/* Duration (in milliseconds) of each step in the ramp. + * This parameter is ignored if ramp_duration is equal to + * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE. + * Supported value: 1 + */ + + u16 ramp_curve; +/* Type of ramping curve. + * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR + */ + + u16 reserved_for_align; + /* Reserved. This field must be set to zero. */ + + u16 target_gain_ch_1; + /* Target linear gain for channel 1 in Q13 format; */ + + u16 target_gain_ch_2; + /* Target linear gain for channel 2 in Q13 format; */ + + u16 target_gain_ch_3; + /* Target linear gain for channel 3 in Q13 format; */ + + u16 target_gain_ch_4; + /* Target linear gain for channel 4 in Q13 format; */ + + u16 target_gain_ch_5; + /* Target linear gain for channel 5 in Q13 format; */ + + u16 target_gain_ch_6; + /* Target linear gain for channel 6 in Q13 format; */ + + u16 target_gain_ch_7; + /* Target linear gain for channel 7 in Q13 format; */ + + u16 target_gain_ch_8; + /* Target linear gain for channel 8 in Q13 format; */ +} __packed; + +/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V7 command. + * Immediately following this structure are num_gains of the + * adm_session_copp_gain_v5structure. + */ +struct adm_cmd_matrix_ramp_gains_v7 { + struct apr_hdr hdr; + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). + * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX + * macros to set this field. +*/ + + u16 num_gains; + /* Number of gains being applied. */ + + u16 reserved_for_align; + /* Reserved. This field must be set to zero.*/ +} __packed; + +/* Session-to-COPP path gain structure, used by the + * #ADM_CMD_MATRIX_RAMP_GAINS_V7 command. + * This structure specifies the target + * gain (per channel) that must be applied + * to a particular session-to-COPP path in + * the audio matrix. The structure can + * also be used to apply the gain globally + * to all session-to-COPP paths that + * exist for the given session. + * The aDSP uses device channel mapping to + * determine which channel gains to + * use from this command. For example, + * if the device is configured as stereo, + * the aDSP uses only target_gain_ch_1 and + * target_gain_ch_2, and it ignores + * the others. + */ +struct adm_session_copp_gain_v7 { + u16 session_id; +/* Handle of the ASM session. + * Supported values: 1 to 8. + */ + + u16 copp_id; +/* Handle of the COPP. Gain will be applied on the Session ID + * COPP ID path. + */ + + u16 ramp_duration; +/* Duration (in milliseconds) of the ramp over + * which target gains are + * to be applied. Use + * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE + * to indicate that gain must be applied immediately. + */ + + u16 step_duration; +/* Duration (in milliseconds) of each step in the ramp. + * This parameter is ignored if ramp_duration is equal to + * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE. + * Supported value: 1 + */ + + u16 ramp_curve; +/* Type of ramping curve. + * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR + */ + + u16 stream_type; +/* Type of stream_type. + * Supported value: #STREAM_TYPE_ASM STREAM_TYPE_LSM + */ + u16 num_channels; +/* Number of channels on which gain needs to be applied. + * Supported value: 1 to 32. + */ + u16 reserved_for_align; + /* Reserved. This field must be set to zero. */ +} __packed; + +/* Allows to set mute/unmute on various session-to-COPP paths. + * For every session-to-COPP path (stream-device interconnection), + * mute/unmute can be set individually on the output channels. + */ +#define ADM_CMD_MATRIX_MUTE_V5 0x0001032D + +/* Allows to set mute/unmute on various session-to-COPP paths. + * For every session-to-COPP path (stream-device interconnection), + * mute/unmute can be set individually on the output channels. + */ +#define ADM_CMD_MATRIX_MUTE_V7 0x0001036D + +/* Indicates that mute/unmute in the + * current adm_session_copp_mute_v5structure + * is to be applied to all the session-to-COPP + * paths that exist for the specified session. + */ +#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF + +/* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/ +struct adm_cmd_matrix_mute_v5 { + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). + * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX + * macros to set this field. + */ + + u16 session_id; +/* Handle of the ASM session. + * Supported values: 1 to 8. + */ + + u16 copp_id; +/* Handle of the COPP. + * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS + * to indicate that mute/unmute must be applied to + * all the COPPs connected to session_id. + * Supported values: + * - 0xFFFF -- Apply mute/unmute to all connected COPPs + * - Other values -- Valid COPP ID + */ + + u8 mute_flag_ch_1; + /* Mute flag for channel 1 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_2; + /* Mute flag for channel 2 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_3; + /* Mute flag for channel 3 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_4; + /* Mute flag for channel 4 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_5; + /* Mute flag for channel 5 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_6; + /* Mute flag for channel 6 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_7; + /* Mute flag for channel 7 is set to unmute (0) or mute (1). */ + + u8 mute_flag_ch_8; + /* Mute flag for channel 8 is set to unmute (0) or mute (1). */ + + u16 ramp_duration; +/* Period (in milliseconds) over which the soft mute/unmute will be + * applied. + * Supported values: 0 (Default) to 0xFFFF + * The default of 0 means mute/unmute will be applied immediately. + */ + + u16 reserved_for_align; + /* Clients must set this field to zero.*/ +} __packed; + +/* Payload of the #ADM_CMD_MATRIX_MUTE_V7 command*/ +struct adm_cmd_matrix_mute_v7 { + struct apr_hdr hdr; + u32 matrix_id; +/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1). + * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX + * macros to set this field. + */ + + u16 session_id; +/* Handle of the ASM session. + * Supported values: 1 to . + */ + + u16 copp_id; +/* Handle of the COPP. + * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS + * to indicate that mute/unmute must be applied to + * all the COPPs connected to session_id. + * Supported values: + * - 0xFFFF -- Apply mute/unmute to all connected COPPs + * - Other values -- Valid COPP ID + */ + + u16 ramp_duration; +/* Duration (in milliseconds) of the ramp over + * which target gains are + * to be applied. Use + * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE + * to indicate that gain must be applied immediately. + */ + + u16 stream_type; +/* Specify whether the stream type is connectedon the ASM or LSM + * Supported value: 1 + */ + u16 num_channels; +/* Number of channels on which gain needs to be applied + * Supported value: 1 to 32 + */ +} __packed; + +#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8) + +struct asm_aac_stereo_mix_coeff_selection_param_v2 { + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + u32 aac_stereo_mix_coeff_flag; +} __packed; + +/* Allows a client to connect the desired stream to + * the desired AFE port through the stream router + * + * This command allows the client to connect specified session to + * specified AFE port. This is used for compressed streams only + * opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or + * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command. + * + * @prerequisites + * Session ID and AFE Port ID must be valid. + * #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or + * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED + * must have been called on this session. + */ + +#define ADM_CMD_CONNECT_AFE_PORT_V5 0x0001032E +#define ADM_CMD_DISCONNECT_AFE_PORT_V5 0x0001032F +/* Enumeration for the Rx stream router ID.*/ +#define ADM_STRTR_ID_RX 0 +/* Enumeration for the Tx stream router ID.*/ +#define ADM_STRTR_IDX 1 + +/* Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/ +struct adm_cmd_connect_afe_port_v5 { + struct apr_hdr hdr; + u8 mode; +/* ID of the stream router (RX/TX). Use the + * ADM_STRTR_ID_RX or ADM_STRTR_IDX macros + * to set this field. + */ + + u8 session_id; + /* Session ID of the stream to connect */ + + u16 afe_port_id; + /* Port ID of the AFE port to connect to.*/ + u32 num_channels; +/* Number of device channels + * Supported values: 2(Audio Sample Packet), + * 8 (HBR Audio Stream Sample Packet) + */ + + u32 sampling_rate; +/* Device sampling rate +* Supported values: Any +*/ +} __packed; + + +/* adsp_adm_api.h */ + + +/* Port ID. Update afe_get_port_index + * when a new port is added here. */ +#define PRIMARY_I2S_RX 0 +#define PRIMARY_I2S_TX 1 +#define SECONDARY_I2S_RX 4 +#define SECONDARY_I2S_TX 5 +#define MI2S_RX 6 +#define MI2S_TX 7 +#define HDMI_RX 8 +#define RSVD_2 9 +#define RSVD_3 10 +#define DIGI_MIC_TX 11 +#define VOICE2_PLAYBACK_TX 0x8002 +#define VOICE_RECORD_RX 0x8003 +#define VOICE_RECORD_TX 0x8004 +#define VOICE_PLAYBACK_TX 0x8005 + +/* Slimbus Multi channel port id pool */ +#define SLIMBUS_0_RX 0x4000 +#define SLIMBUS_0_TX 0x4001 +#define SLIMBUS_1_RX 0x4002 +#define SLIMBUS_1_TX 0x4003 +#define SLIMBUS_2_RX 0x4004 +#define SLIMBUS_2_TX 0x4005 +#define SLIMBUS_3_RX 0x4006 +#define SLIMBUS_3_TX 0x4007 +#define SLIMBUS_4_RX 0x4008 +#define SLIMBUS_4_TX 0x4009 +#define SLIMBUS_5_RX 0x400a +#define SLIMBUS_5_TX 0x400b +#define SLIMBUS_6_RX 0x400c +#define SLIMBUS_6_TX 0x400d +#define SLIMBUS_7_RX 0x400e +#define SLIMBUS_7_TX 0x400f +#define SLIMBUS_8_RX 0x4010 +#define SLIMBUS_8_TX 0x4011 +#define SLIMBUS_PORT_LAST SLIMBUS_8_TX +#define INT_BT_SCO_RX 0x3000 +#define INT_BT_SCO_TX 0x3001 +#define INT_BT_A2DP_RX 0x3002 +#define INT_FM_RX 0x3004 +#define INT_FM_TX 0x3005 +#define RT_PROXY_PORT_001_RX 0x2000 +#define RT_PROXY_PORT_001_TX 0x2001 +#define DISPLAY_PORT_RX 0x6020 + +#define AFE_PORT_INVALID 0xFFFF +#define SLIMBUS_INVALID AFE_PORT_INVALID + +#define AFE_PORT_CMD_START 0x000100ca + +#define AFE_EVENT_RTPORT_START 0 +#define AFE_EVENT_RTPORT_STOP 1 +#define AFE_EVENT_RTPORT_LOW_WM 2 +#define AFE_EVENT_RTPORT_HI_WM 3 + +#define ADSP_AFE_VERSION 0x00200000 + +/* Size of the range of port IDs for the audio interface. */ +#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE 0xF + +/* Size of the range of port IDs for internal BT-FM ports. */ +#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE 0x6 + +/* Size of the range of port IDs for SLIMbus<sup>® + * </sup> multichannel + * ports. + */ +#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE 0xA + +/* Size of the range of port IDs for real-time proxy ports. */ +#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE 0x4 + +/* Size of the range of port IDs for pseudoports. */ +#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE 0x5 + +/* Start of the range of port IDs for the audio interface. */ +#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START 0x1000 + +/* End of the range of port IDs for the audio interface. */ +#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \ + (AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\ + AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1) + +/* Start of the range of port IDs for real-time proxy ports. */ +#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_START 0x2000 + +/* End of the range of port IDs for real-time proxy ports. */ +#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \ + (AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\ + AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1) + +/* Start of the range of port IDs for internal BT-FM devices. */ +#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START 0x3000 + +/* End of the range of port IDs for internal BT-FM devices. */ +#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \ + (AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\ + AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1) + +/* Start of the range of port IDs for SLIMbus devices. */ +#define AFE_PORT_ID_SLIMBUS_RANGE_START 0x4000 + +/* End of the range of port IDs for SLIMbus devices. */ +#define AFE_PORT_ID_SLIMBUS_RANGE_END \ + (AFE_PORT_ID_SLIMBUS_RANGE_START +\ + AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1) + +/* Start of the range of port IDs for pseudoports. */ +#define AFE_PORT_ID_PSEUDOPORT_RANGE_START 0x8001 + +/* End of the range of port IDs for pseudoports. */ +#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \ + (AFE_PORT_ID_PSEUDOPORT_RANGE_START +\ + AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1) + +/* Start of the range of port IDs for TDM devices. */ +#define AFE_PORT_ID_TDM_PORT_RANGE_START 0x9000 + +/* End of the range of port IDs for TDM devices. */ +#define AFE_PORT_ID_TDM_PORT_RANGE_END \ + (AFE_PORT_ID_TDM_PORT_RANGE_START+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_RX_1 0x1040 +#define AFE_PORT_ID_SECONDARY_MI2S_RX_2 0x1042 +#define AFE_PORT_ID_SECONDARY_MI2S_RX_3 0x1044 +#define AFE_PORT_ID_SECONDARY_MI2S_RX_4 0x1046 +#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003 +#define AFE_PORT_ID_SECONDARY_MI2S_TX_1 0x1041 +#define AFE_PORT_ID_SECONDARY_MI2S_TX_2 0x1043 +#define AFE_PORT_ID_SECONDARY_MI2S_TX_3 0x1045 +#define AFE_PORT_ID_SECONDARY_MI2S_TX_4 0x1047 +#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004 +#define AFE_PORT_ID_TERTIARY_MI2S_RX_1 0x1048 +#define AFE_PORT_ID_TERTIARY_MI2S_RX_2 0x104A +#define AFE_PORT_ID_TERTIARY_MI2S_RX_3 0x104C +#define AFE_PORT_ID_TERTIARY_MI2S_RX_4 0x104E +#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005 +#define AFE_PORT_ID_TERTIARY_MI2S_TX_1 0x1049 +#define AFE_PORT_ID_TERTIARY_MI2S_TX_2 0x104B +#define AFE_PORT_ID_TERTIARY_MI2S_TX_3 0x104D +#define AFE_PORT_ID_TERTIARY_MI2S_TX_4 0x104F +#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006 +#define AFE_PORT_ID_QUATERNARY_MI2S_RX_1 0x1020 +#define AFE_PORT_ID_QUATERNARY_MI2S_RX_2 0x1022 +#define AFE_PORT_ID_QUATERNARY_MI2S_RX_3 0x1024 +#define AFE_PORT_ID_QUATERNARY_MI2S_RX_4 0x1026 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX_1 0x1021 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX_2 0x1023 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX_3 0x1025 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX_4 0x1027 +#define AUDIO_PORT_ID_I2S_RX 0x1008 +#define AFE_PORT_ID_DIGITAL_MIC_TX 0x1009 +#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A +#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B +#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C +#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D +#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E +#define AFE_PORT_ID_SECONDARY_MI2S_RX_SD1 0x1010 +#define AFE_PORT_ID_TERTIARY_PCM_RX 0x1012 +#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013 +#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014 +#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015 +#define AFE_PORT_ID_QUINARY_MI2S_RX 0x1016 +#define AFE_PORT_ID_QUINARY_MI2S_TX 0x1017 +/* ID of the senary MI2S Rx port. */ +#define AFE_PORT_ID_SENARY_MI2S_RX 0x1018 +/* ID of the senary MI2S Tx port. */ +#define AFE_PORT_ID_SENARY_MI2S_TX 0x1019 +/* ID of the Internal 0 MI2S Rx port */ +#define AFE_PORT_ID_INT0_MI2S_RX 0x102E +/* ID of the Internal 0 MI2S Tx port */ +#define AFE_PORT_ID_INT0_MI2S_TX 0x102F +/* ID of the Internal 1 MI2S Rx port */ +#define AFE_PORT_ID_INT1_MI2S_RX 0x1030 +/* ID of the Internal 1 MI2S Tx port */ +#define AFE_PORT_ID_INT1_MI2S_TX 0x1031 +/* ID of the Internal 2 MI2S Rx port */ +#define AFE_PORT_ID_INT2_MI2S_RX 0x1032 +/* ID of the Internal 2 MI2S Tx port */ +#define AFE_PORT_ID_INT2_MI2S_TX 0x1033 +/* ID of the Internal 3 MI2S Rx port */ +#define AFE_PORT_ID_INT3_MI2S_RX 0x1034 +/* ID of the Internal 3 MI2S Tx port */ +#define AFE_PORT_ID_INT3_MI2S_TX 0x1035 +/* ID of the Internal 4 MI2S Rx port */ +#define AFE_PORT_ID_INT4_MI2S_RX 0x1036 +/* ID of the Internal 4 MI2S Tx port */ +#define AFE_PORT_ID_INT4_MI2S_TX 0x1037 +/* ID of the Internal 5 MI2S Rx port */ +#define AFE_PORT_ID_INT5_MI2S_RX 0x1038 +/* ID of the Internal 5 MI2S Tx port */ +#define AFE_PORT_ID_INT5_MI2S_TX 0x1039 +/* ID of the Internal 6 MI2S Rx port */ +#define AFE_PORT_ID_INT6_MI2S_RX 0x103A +/* ID of the Internal 6 MI2S Tx port */ +#define AFE_PORT_ID_INT6_MI2S_TX 0x103B +#define AFE_PORT_ID_SPDIF_RX 0x5000 +#define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000 +#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001 +#define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000 +#define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001 +#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002 +#define AFE_PORT_ID_INTERNAL_FM_RX 0x3004 +#define AFE_PORT_ID_INTERNAL_FM_TX 0x3005 +/* SLIMbus Rx port on channel 0. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000 +/* SLIMbus Tx port on channel 0. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001 +/* SLIMbus Rx port on channel 1. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002 +/* SLIMbus Tx port on channel 1. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003 +/* SLIMbus Rx port on channel 2. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004 +/* SLIMbus Tx port on channel 2. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005 +/* SLIMbus Rx port on channel 3. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006 +/* SLIMbus Tx port on channel 3. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007 +/* SLIMbus Rx port on channel 4. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008 +/* SLIMbus Tx port on channel 4. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009 +/* SLIMbus Rx port on channel 5. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX 0x400a +/* SLIMbus Tx port on channel 5. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX 0x400b +/* SLIMbus Rx port on channel 6. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c +/* SLIMbus Tx port on channel 6. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d +/* SLIMbus Rx port on channel 7. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_RX 0x400e +/* SLIMbus Tx port on channel 7. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_TX 0x400f +/* SLIMbus Rx port on channel 8. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_RX 0x4010 +/* SLIMbus Tx port on channel 8. */ +#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_TX 0x4011 +/* AFE Rx port for audio over Display port */ +#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020 +/*USB AFE port */ +#define AFE_PORT_ID_USB_RX 0x7000 +#define AFE_PORT_ID_USB_TX 0x7001 + +/* Generic pseudoport 1. */ +#define AFE_PORT_ID_PSEUDOPORT_01 0x8001 +/* Generic pseudoport 2. */ +#define AFE_PORT_ID_PSEUDOPORT_02 0x8002 + +/* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx} + Primary Aux PCM Tx port ID. +*/ +#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B +/* Pseudoport that corresponds to the voice Rx path. + * For recording, the voice Rx path samples are written to this + * port and consumed by the audio path. + */ + +#define AFE_PORT_ID_VOICE_RECORD_RX 0x8003 + +/* Pseudoport that corresponds to the voice Tx path. + * For recording, the voice Tx path samples are written to this + * port and consumed by the audio path. + */ + +#define AFE_PORT_ID_VOICE_RECORD_TX 0x8004 +/* Pseudoport that corresponds to in-call voice delivery samples. + * During in-call audio delivery, the audio path delivers samples + * to this port from where the voice path delivers them on the + * Rx path. + */ +#define AFE_PORT_ID_VOICE2_PLAYBACK_TX 0x8002 +#define AFE_PORT_ID_VOICE_PLAYBACK_TX 0x8005 +/* + * Proxyport used for voice call data processing. + * In cases like call-screening feature, where user can communicate + * with caller with the help of "call screen" mode, and without + * connecting the call with any HW input/output devices in the phon, + * voice call can use Pseudo port to start voice data processing. + */ +#define RT_PROXY_PORT_002_TX 0x2003 +#define RT_PROXY_PORT_002_RX 0x2002 + +#define AFE_PORT_ID_PRIMARY_TDM_RX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x00) +#define AFE_PORT_ID_PRIMARY_TDM_RX_1 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x02) +#define AFE_PORT_ID_PRIMARY_TDM_RX_2 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x04) +#define AFE_PORT_ID_PRIMARY_TDM_RX_3 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x06) +#define AFE_PORT_ID_PRIMARY_TDM_RX_4 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x08) +#define AFE_PORT_ID_PRIMARY_TDM_RX_5 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0A) +#define AFE_PORT_ID_PRIMARY_TDM_RX_6 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0C) +#define AFE_PORT_ID_PRIMARY_TDM_RX_7 \ + (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0E) + +#define AFE_PORT_ID_PRIMARY_TDM_TX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x01) +#define AFE_PORT_ID_PRIMARY_TDM_TX_1 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x02) +#define AFE_PORT_ID_PRIMARY_TDM_TX_2 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x04) +#define AFE_PORT_ID_PRIMARY_TDM_TX_3 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x06) +#define AFE_PORT_ID_PRIMARY_TDM_TX_4 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x08) +#define AFE_PORT_ID_PRIMARY_TDM_TX_5 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0A) +#define AFE_PORT_ID_PRIMARY_TDM_TX_6 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0C) +#define AFE_PORT_ID_PRIMARY_TDM_TX_7 \ + (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0E) + +#define AFE_PORT_ID_SECONDARY_TDM_RX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x10) +#define AFE_PORT_ID_SECONDARY_TDM_RX_1 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x02) +#define AFE_PORT_ID_SECONDARY_TDM_RX_2 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x04) +#define AFE_PORT_ID_SECONDARY_TDM_RX_3 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x06) +#define AFE_PORT_ID_SECONDARY_TDM_RX_4 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x08) +#define AFE_PORT_ID_SECONDARY_TDM_RX_5 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0A) +#define AFE_PORT_ID_SECONDARY_TDM_RX_6 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0C) +#define AFE_PORT_ID_SECONDARY_TDM_RX_7 \ + (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0E) + +#define AFE_PORT_ID_SECONDARY_TDM_TX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x11) +#define AFE_PORT_ID_SECONDARY_TDM_TX_1 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x02) +#define AFE_PORT_ID_SECONDARY_TDM_TX_2 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x04) +#define AFE_PORT_ID_SECONDARY_TDM_TX_3 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x06) +#define AFE_PORT_ID_SECONDARY_TDM_TX_4 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x08) +#define AFE_PORT_ID_SECONDARY_TDM_TX_5 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0A) +#define AFE_PORT_ID_SECONDARY_TDM_TX_6 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0C) +#define AFE_PORT_ID_SECONDARY_TDM_TX_7 \ + (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0E) + +#define AFE_PORT_ID_TERTIARY_TDM_RX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x20) +#define AFE_PORT_ID_TERTIARY_TDM_RX_1 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x02) +#define AFE_PORT_ID_TERTIARY_TDM_RX_2 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x04) +#define AFE_PORT_ID_TERTIARY_TDM_RX_3 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x06) +#define AFE_PORT_ID_TERTIARY_TDM_RX_4 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x08) +#define AFE_PORT_ID_TERTIARY_TDM_RX_5 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0A) +#define AFE_PORT_ID_TERTIARY_TDM_RX_6 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0C) +#define AFE_PORT_ID_TERTIARY_TDM_RX_7 \ + (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0E) + +#define AFE_PORT_ID_TERTIARY_TDM_TX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x21) +#define AFE_PORT_ID_TERTIARY_TDM_TX_1 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x02) +#define AFE_PORT_ID_TERTIARY_TDM_TX_2 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x04) +#define AFE_PORT_ID_TERTIARY_TDM_TX_3 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x06) +#define AFE_PORT_ID_TERTIARY_TDM_TX_4 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x08) +#define AFE_PORT_ID_TERTIARY_TDM_TX_5 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0A) +#define AFE_PORT_ID_TERTIARY_TDM_TX_6 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0C) +#define AFE_PORT_ID_TERTIARY_TDM_TX_7 \ + (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0E) + +#define AFE_PORT_ID_QUATERNARY_TDM_RX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x30) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_1 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x02) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_2 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x04) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_3 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x06) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_4 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x08) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_5 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0A) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_6 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0C) +#define AFE_PORT_ID_QUATERNARY_TDM_RX_7 \ + (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0E) + +#define AFE_PORT_ID_QUATERNARY_TDM_TX \ + (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x31) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_1 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x02) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_2 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x04) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_3 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x06) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_4 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x08) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_5 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0A) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_6 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0C) +#define AFE_PORT_ID_QUATERNARY_TDM_TX_7 \ + (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0E) + +#define AFE_PORT_ID_INVALID 0xFFFF + +#define AAC_ENC_MODE_AAC_LC 0x02 +#define AAC_ENC_MODE_AAC_P 0x05 +#define AAC_ENC_MODE_EAAC_P 0x1D + +#define AFE_PSEUDOPORT_CMD_START 0x000100cf +struct afe_pseudoport_start_command { + struct apr_hdr hdr; + u16 port_id; /* Pseudo Port 1 = 0x8000 */ + /* Pseudo Port 2 = 0x8001 */ + /* Pseudo Port 3 = 0x8002 */ + u16 timing; /* FTRT = 0 , AVTimer = 1, */ +} __packed; + +#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0 +struct afe_pseudoport_stop_command { + struct apr_hdr hdr; + u16 port_id; /* Pseudo Port 1 = 0x8000 */ + /* Pseudo Port 2 = 0x8001 */ + /* Pseudo Port 3 = 0x8002 */ + u16 reserved; +} __packed; + + +#define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202 +#define AFE_PARAM_ID_ENABLE 0x00010203 + +/* Payload of the #AFE_PARAM_ID_ENABLE + * parameter, which enables or + * disables any module. + * The fixed size of this structure is four bytes. + */ + +struct afe_mod_enable_param { + u16 enable; + /* Enables (1) or disables (0) the module. */ + + u16 reserved; + /* This field must be set to zero. + */ +} __packed; + +/* ID of the configuration parameter used by the + * #AFE_MODULE_SIDETONE_IIR_FILTER module. + */ +#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204 +#define MAX_SIDETONE_IIR_DATA_SIZE 224 +#define MAX_NO_IIR_FILTER_STAGE 10 + +struct afe_sidetone_iir_filter_config_params { + u16 num_biquad_stages; +/* Number of stages. + * Supported values: Minimum of 5 and maximum of 10 + */ + + u16 pregain; +/* Pregain for the compensating filter response. + * Supported values: Any number in Q13 format + */ + uint8_t iir_config[MAX_SIDETONE_IIR_DATA_SIZE]; +} __packed; + +#define AFE_MODULE_LOOPBACK 0x00010205 +#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206 + +/* Used by RTAC */ +struct afe_rtac_user_data_set_v2 { + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + + /* Actual size of the payload in bytes. + * This is used for parsing the parameter payload. + * Supported values: > 0 + */ + u16 payload_size; + + /* The header detailing the memory mapping for out of band. */ + struct mem_mapping_hdr mem_hdr; + + /* The parameter header for the parameter data to set */ + struct param_hdr_v1 param_hdr; + + /* The parameter data to be filled when sent inband */ + u32 *param_data; +} __packed; + +struct afe_rtac_user_data_set_v3 { + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + /* Reserved for future enhancements. Must be 0. */ + u16 reserved; + + /* The header detailing the memory mapping for out of band. */ + struct mem_mapping_hdr mem_hdr; + + /* The size of the parameter header and parameter data */ + u32 payload_size; + + /* The parameter header for the parameter data to set */ + struct param_hdr_v3 param_hdr; + + /* The parameter data to be filled when sent inband */ + u32 *param_data; +} __packed; + +struct afe_rtac_user_data_get_v2 { + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + + /* Actual size of the payload in bytes. + * This is used for parsing the parameter payload. + * Supported values: > 0 + */ + u16 payload_size; + + /* The header detailing the memory mapping for out of band. */ + struct mem_mapping_hdr mem_hdr; + + /* The module ID of the parameter to get */ + u32 module_id; + + /* The parameter ID of the parameter to get */ + u32 param_id; + + /* The parameter data to be filled when sent inband */ + struct param_hdr_v1 param_hdr; +} __packed; + +struct afe_rtac_user_data_get_v3 { + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + /* Reserved for future enhancements. Must be 0. */ + u16 reserved; + + /* The header detailing the memory mapping for out of band. */ + struct mem_mapping_hdr mem_hdr; + + /* The parameter data to be filled when sent inband */ + struct param_hdr_v3 param_hdr; +} __packed; +#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF +struct afe_port_cmd_set_param_v2 { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + + /* + * Actual size of the payload in bytes. + * This is used for parsing the parameter payload. + * Supported values: > 0 + */ + u16 payload_size; + + /* The header detailing the memory mapping for out of band. */ + struct mem_mapping_hdr mem_hdr; + + /* The parameter data to be filled when sent inband */ + u8 param_data[0]; +} __packed; + +#define AFE_PORT_CMD_SET_PARAM_V3 0x000100FA +struct afe_port_cmd_set_param_v3 { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* Port ID of the AFE port to configure. Port interface and direction + * (Rx or Tx) to configure. An even number represents the Rx direction, + * and an odd number represents the Tx direction. + */ + u16 port_id; + + /* Reserved. This field must be set to zero. */ + u16 reserved; + + /* The memory mapping header to be used when sending outband */ + struct mem_mapping_hdr mem_hdr; + + /* The total size of the payload, including param_hdr_v3 */ + u32 payload_size; + + /* + * The parameter data to be filled when sent inband. + * Must include param_hdr packed correctly. + */ + u8 param_data[0]; +} __packed; + +/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter, + * which gets/sets loopback gain of a port to an Rx port. + * The Tx port ID of the loopback is part of the set_param command. + */ + +struct afe_loopback_gain_per_path_param { + u16 rx_port_id; +/* Rx port of the loopback. */ + +u16 gain; +/* Loopback gain per path of the port. + * Supported values: Any number in Q13 format + */ +} __packed; + +/* Parameter ID used to configure and enable/disable the + * loopback path. The difference with respect to the existing + * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be + * configured as source port in loopback path. Port-id in + * AFE_PORT_CMD_SET_PARAM cmd is the source port 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 { + u32 loopback_cfg_minor_version; +/* Minor version used for tracking the version of the RMC module + * configuration interface. + * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG + */ + u16 dst_port_id; + /* Destination Port Id. */ + u16 routing_mode; +/* Specifies data path type from src to dest port. + * Supported values: + * #LB_MODE_DEFAULT + * #LB_MODE_SIDETONE + * #LB_MODE_EC_REF_VOICE_AUDIO + * #LB_MODE_EC_REF_VOICE_A + * #LB_MODE_EC_REF_VOICE + */ + + u16 enable; +/* Specifies whether to enable (1) or + * disable (0) an AFE loopback. + */ + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0. + */ + +} __packed; + +struct afe_loopback_sidetone_gain { + u16 rx_port_id; + u16 gain; +} __packed; + +struct loopback_cfg_data { + u32 loopback_cfg_minor_version; +/* Minor version used for tracking the version of the RMC module + * configuration interface. + * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG + */ + u16 dst_port_id; + /* Destination Port Id. */ + u16 routing_mode; +/* Specifies data path type from src to dest port. + * Supported values: + * #LB_MODE_DEFAULT + * #LB_MODE_SIDETONE + * #LB_MODE_EC_REF_VOICE_AUDIO + * #LB_MODE_EC_REF_VOICE_A + * #LB_MODE_EC_REF_VOICE + */ + + u16 enable; +/* Specifies whether to enable (1) or + * disable (0) an AFE loopback. + */ + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0. + */ +} __packed; + +struct afe_st_loopback_cfg_v1 { + struct apr_hdr hdr; + struct mem_mapping_hdr mem_hdr; + struct param_hdr_v1 gain_pdata; + struct afe_loopback_sidetone_gain gain_data; + struct param_hdr_v1 cfg_pdata; + struct loopback_cfg_data cfg_data; +} __packed; + +struct afe_loopback_iir_cfg_v2 { + struct apr_hdr hdr; + struct mem_mapping_hdr param; + struct param_hdr_v1 st_iir_enable_pdata; + struct afe_mod_enable_param st_iir_mode_enable_data; + struct param_hdr_v1 st_iir_filter_config_pdata; + struct afe_sidetone_iir_filter_config_params st_iir_filter_config_data; +} __packed; +#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_176P4K 176400 +#define AFE_PORT_SAMPLE_RATE_192K 192000 +#define AFE_PORT_SAMPLE_RATE_352P8K 352800 +#define AFE_LINEAR_PCM_DATA 0x0 +#define AFE_NON_LINEAR_DATA 0x1 +#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2 +#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3 +#define AFE_GENERIC_COMPRESSED 0x8 + +/* This param id is used to configure I2S interface */ +#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D +#define AFE_API_VERSION_I2S_CONFIG 0x1 +/* Enumeration for setting the I2S configuration + * channel_mode parameter to + * serial data wire number 1-3 (SD3). + */ +#define AFE_PORT_I2S_SD0 0x1 +#define AFE_PORT_I2S_SD1 0x2 +#define AFE_PORT_I2S_SD2 0x3 +#define AFE_PORT_I2S_SD3 0x4 +#define AFE_PORT_I2S_QUAD01 0x5 +#define AFE_PORT_I2S_QUAD23 0x6 +#define AFE_PORT_I2S_6CHS 0x7 +#define AFE_PORT_I2S_8CHS 0x8 +#define AFE_PORT_I2S_MONO 0x0 +#define AFE_PORT_I2S_STEREO 0x1 +#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0 +#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 + +/* Payload of the #AFE_PARAM_ID_I2S_CONFIG + * command's (I2S configuration + * parameter). + */ +struct afe_param_id_i2s_cfg { + u32 i2s_cfg_minor_version; +/* Minor version used for tracking the version of the I2S + * configuration interface. + * Supported values: #AFE_API_VERSION_I2S_CONFIG + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + + u16 channel_mode; +/* I2S lines and multichannel operation. + * Supported values: + * - #AFE_PORT_I2S_SD0 + * - #AFE_PORT_I2S_SD1 + * - #AFE_PORT_I2S_SD2 + * - #AFE_PORT_I2S_SD3 + * - #AFE_PORT_I2S_QUAD01 + * - #AFE_PORT_I2S_QUAD23 + * - #AFE_PORT_I2S_6CHS + * - #AFE_PORT_I2S_8CHS + */ + + u16 mono_stereo; +/* Specifies mono or stereo. This applies only when + * a single I2S line is used. + * Supported values: + * - #AFE_PORT_I2S_MONO + * - #AFE_PORT_I2S_STEREO + */ + + u16 ws_src; +/* Word select source: internal or external. + * Supported values: + * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL + * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K + */ + + u16 data_format; +/* data format + * Supported values: + * - #LINEAR_PCM_DATA + * - #NON_LINEAR_DATA + * - #LINEAR_PCM_DATA_PACKED_IN_60958 + * - #NON_LINEAR_DATA_PACKED_IN_60958 + */ + u16 reserved; + /* This field must be set to zero. */ +} __packed; + +/* + * This param id is used to configure PCM interface + */ + +#define AFE_API_VERSION_SPDIF_CONFIG 0x1 +#define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1 +#define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1 +#define AFE_CH_STATUS_A 1 +#define AFE_CH_STATUS_B 2 + +#define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244 +#define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245 +#define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246 + +#define AFE_PORT_CLK_ROOT_LPAPLL 0x3 +#define AFE_PORT_CLK_ROOT_LPAQ6PLL 0x4 + +struct afe_param_id_spdif_cfg { +/* Minor version used for tracking the version of the SPDIF + * configuration interface. + * Supported values: #AFE_API_VERSION_SPDIF_CONFIG + */ + u32 spdif_cfg_minor_version; + +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_22_05K + * - #AFE_PORT_SAMPLE_RATE_32K + * - #AFE_PORT_SAMPLE_RATE_44_1K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_176_4K + * - #AFE_PORT_SAMPLE_RATE_192K + */ + u32 sample_rate; + +/* data format + * Supported values: + * - #AFE_LINEAR_PCM_DATA + * - #AFE_NON_LINEAR_DATA + */ + u16 data_format; +/* Number of channels supported by the port + * - PCM - 1, Compressed Case - 2 + */ + u16 num_channels; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 bit_width; +/* This field must be set to zero. */ + u16 reserved; +} __packed; + +struct afe_param_id_spdif_ch_status_cfg { + u32 ch_status_cfg_minor_version; +/* Minor version used for tracking the version of channel + * status configuration. Current supported version is 1 + */ + + u32 status_type; +/* Indicate if the channel status is for channel A or B + * Supported values: + * - #AFE_CH_STATUS_A + * - #AFE_CH_STATUS_B + */ + + u8 status_bits[24]; +/* Channel status - 192 bits for channel + * Byte ordering as defined by IEC60958-3 + */ + + u8 status_mask[24]; +/* Channel status with mask bits 1 will be applied. + * Byte ordering as defined by IEC60958-3 + */ +} __packed; + +struct afe_param_id_spdif_clk_cfg { + u32 clk_cfg_minor_version; +/* Minor version used for tracking the version of SPDIF + * interface clock configuration. Current supported version + * is 1 + */ + + u32 clk_value; +/* Specifies the clock frequency in Hz to set + * Supported values: + * 0 - Disable the clock + * 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2 + * (channels A and B) + */ + + u32 clk_root; +/* Specifies SPDIF root clk source + * Supported Values: + * - #AFE_PORT_CLK_ROOT_LPAPLL + * - #AFE_PORT_CLK_ROOT_LPAQ6PLL + */ +} __packed; + +struct afe_spdif_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 + +#define AFE_PORT_MAX_AUDIO_CHAN_CNT_V2 0x20 + +/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus + * port configuration parameter. + */ + +struct afe_param_id_slimbus_cfg { + u32 sb_cfg_minor_version; +/* Minor version used for tracking the version of the SLIMBUS + * configuration interface. + * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG + */ + + u16 slimbus_dev_id; +/* SLIMbus hardware device ID, which is required to handle + * multiple SLIMbus hardware blocks. + * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2 + */ + + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + + u16 data_format; +/* Data format supported by the SLIMbus hardware. The default is + * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the + * hardware does not perform any format conversions before the data + * transfer. + */ + + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT + */ + + u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; +/* Mapping of shared channel IDs (128 to 255) to which the + * master port is to be connected. + * Shared_channel_mapping[i] represents the shared channel assigned + * for audio channel i in multichannel audio data. + */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K + */ +} __packed; + + +/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS to configure + * USB audio device parameter. It should be used with + * AFE_MODULE_AUDIO_DEV_INTERFACE + */ +#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5 + + +/* ID of the parameter used to set the endianness value for the + * USB audio device. It should be used with + * AFE_MODULE_AUDIO_DEV_INTERFACE + */ +#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA + +/* Minor version used for tracking USB audio configuration */ +#define AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG 0x1 + +/* Payload of the AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS parameter used by + * AFE_MODULE_AUDIO_DEV_INTERFACE. + */ +struct afe_param_id_usb_audio_dev_params { +/* Minor version used for tracking USB audio device parameter. + * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG + */ + u32 cfg_minor_version; +/* Token of actual end USB aduio device */ + u32 dev_token; +} __packed; + +struct afe_param_id_usb_audio_dev_lpcm_fmt { +/* Minor version used for tracking USB audio device parameter. + * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG + */ + u32 cfg_minor_version; +/* Endianness of actual end USB audio device */ + u32 endian; +} __packed; + +/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_CONFIG to configure + * USB audio interface. It should be used with AFE_MODULE_AUDIO_DEV_INTERFACE +*/ +#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4 + +/* Payload of the AFE_PARAM_ID_USB_AUDIO_CONFIG parameter used by + * AFE_MODULE_AUDIO_DEV_INTERFACE. + */ +struct afe_param_id_usb_audio_cfg { +/* Minor version used for tracking USB audio device configuration. + * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG + */ + u32 cfg_minor_version; +/* Sampling rate of the port. + * Supported values: + * - AFE_PORT_SAMPLE_RATE_8K + * - AFE_PORT_SAMPLE_RATE_11025 + * - AFE_PORT_SAMPLE_RATE_12K + * - AFE_PORT_SAMPLE_RATE_16K + * - AFE_PORT_SAMPLE_RATE_22050 + * - AFE_PORT_SAMPLE_RATE_24K + * - AFE_PORT_SAMPLE_RATE_32K + * - AFE_PORT_SAMPLE_RATE_44P1K + * - AFE_PORT_SAMPLE_RATE_48K + * - AFE_PORT_SAMPLE_RATE_96K + * - AFE_PORT_SAMPLE_RATE_192K + */ + u32 sample_rate; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 bit_width; +/* Number of channels. + * Supported values: 1 and 2 + */ + u16 num_channels; +/* Data format supported by the USB. The supported value is + * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM). + */ + u16 data_format; +/* this field must be 0 */ + u16 reserved; +/* device token of actual end USB aduio device */ + u32 dev_token; +/* endianness of this interface */ + u32 endian; +} __packed; + +/* +* This param id is used to configure Real Time Proxy interface. +*/ +#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213 + +/* This version information is used to handle the new +* additions to the config interface in future in backward +* compatible manner. +*/ +#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1 + +/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG + * command (real-time proxy port configuration parameter). + */ +struct afe_param_id_rt_proxy_port_cfg { + u32 rt_proxy_cfg_minor_version; +/* Minor version used for tracking the version of rt-proxy + * config interface. + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 interleaved; +/* Specifies whether the data exchanged between the AFE + * interface and real-time port is interleaved. + * Supported values: - 0 -- Non-interleaved (samples from each + * channel are contiguous in the buffer) - 1 -- Interleaved + * (corresponding samples from each input channel are interleaved + * within the buffer) + */ + + + u16 frame_size; + /* Size of the frames that are used for PCM exchanges with this + * port. + * Supported values: > 0, in bytes + * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples + * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320 + * bytes. + */ + u16 jitter_allowance; +/* Configures the amount of jitter that the port will allow. + * Supported values: > 0 + * For example, if +/-10 ms of jitter is anticipated in the timing + * of sending frames to the port, and the configuration is 16 kHz + * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2 + * bytes/sample = 320. + */ + + u16 low_water_mark; +/* Low watermark in bytes (including all channels). + * Supported values: + * - 0 -- Do not send any low watermark events + * - > 0 -- Low watermark for triggering an event + * If the number of bytes in an internal circular buffer is lower + * than this low_water_mark parameter, a LOW_WATER_MARK event is + * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS + * event). + * Use of watermark events is optional for debugging purposes. + */ + + u16 high_water_mark; +/* High watermark in bytes (including all channels). + * Supported values: + * - 0 -- Do not send any high watermark events + * - > 0 -- High watermark for triggering an event + * If the number of bytes in an internal circular buffer exceeds + * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event + * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS + * event). + * The use of watermark events is optional and for debugging + * purposes. + */ + + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + * - #AFE_PORT_SAMPLE_RATE_48K + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT + */ + + u16 reserved; + /* For 32 bit alignment. */ +} __packed; + + +/* This param id is used to configure the Pseudoport interface */ + +#define AFE_PARAM_ID_PSEUDO_PORT_CONFIG 0x00010219 + +/* Version information used to handle future additions to the configuration + * interface (for backward compatibility). + */ +#define AFE_API_VERSION_PSEUDO_PORT_CONFIG 0x1 + +/* Enumeration for setting the timing_mode parameter to faster than real + * time. + */ +#define AFE_PSEUDOPORT_TIMING_MODE_FTRT 0x0 + +/* Enumeration for setting the timing_mode parameter to real time using + * timers. + */ +#define AFE_PSEUDOPORT_TIMING_MODE_TIMER 0x1 + +/* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by + AFE_MODULE_AUDIO_DEV_INTERFACE. +*/ +struct afe_param_id_pseudo_port_cfg { + u32 pseud_port_cfg_minor_version; + /* + * Minor version used for tracking the version of the pseudoport + * configuration interface. + */ + + u16 bit_width; + /* Bit width of the sample at values 16, 24 */ + + u16 num_channels; + /* Number of channels at values 1 to 8 */ + + u16 data_format; + /* Non-linear data format supported by the pseudoport (for future use). + * At values #AFE_LINEAR_PCM_DATA + */ + + u16 timing_mode; + /* Indicates whether the pseudoport synchronizes to the clock or + * operates faster than real time. + * at values + * - #AFE_PSEUDOPORT_TIMING_MODE_FTRT + * - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend + */ + + u32 sample_rate; + /* Sample rate at which the pseudoport will run. + * at values + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_32K + * - #AFE_PORT_SAMPLE_RATE_48K + * - #AFE_PORT_SAMPLE_RATE_96K + * - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend + */ +} __packed; + +#define AFE_PARAM_ID_TDM_CONFIG 0x0001029D + +#define AFE_API_VERSION_TDM_CONFIG 1 + +#define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE 0 +#define AFE_PORT_TDM_LONG_SYNC_MODE 1 +#define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE 2 + +#define AFE_PORT_TDM_SYNC_SRC_EXTERNAL 0 +#define AFE_PORT_TDM_SYNC_SRC_INTERNAL 1 + +#define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE 0 +#define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE 1 + +#define AFE_PORT_TDM_SYNC_NORMAL 0 +#define AFE_PORT_TDM_SYNC_INVERT 1 + +#define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE 0 +#define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE 1 +#define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE 2 + +/* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by + AFE_MODULE_AUDIO_DEV_INTERFACE. +*/ +struct afe_param_id_tdm_cfg { + u32 tdm_cfg_minor_version; + /**< Minor version used to track TDM configuration. + @values #AFE_API_VERSION_TDM_CONFIG */ + + u32 num_channels; + /**< Number of enabled slots for TDM frame. + @values 1 to 8 */ + + u32 sample_rate; + /**< Sampling rate of the port. + @values + - #AFE_PORT_SAMPLE_RATE_8K + - #AFE_PORT_SAMPLE_RATE_16K + - #AFE_PORT_SAMPLE_RATE_24K + - #AFE_PORT_SAMPLE_RATE_32K + - #AFE_PORT_SAMPLE_RATE_48K + - #AFE_PORT_SAMPLE_RATE_176P4K + - #AFE_PORT_SAMPLE_RATE_352P8K @tablebulletend + */ + + u32 bit_width; + /**< Bit width of the sample. + * @values 16, 24, 32 + */ + + u16 data_format; + /**< Data format: linear ,compressed, generic compresssed + @values + - #AFE_LINEAR_PCM_DATA + - #AFE_NON_LINEAR_DATA + - #AFE_GENERIC_COMPRESSED + */ + + u16 sync_mode; + /**< TDM synchronization setting. + @values (short, long, slot) sync mode + - #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE + - #AFE_PORT_TDM_LONG_SYNC_MODE + - #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend + */ + + u16 sync_src; + /**< Synchronization source. + @values + - #AFE_PORT_TDM_SYNC_SRC_EXTERNAL + - #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend */ + + u16 nslots_per_frame; + /**< Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32. + @values 1 - 32 */ + + u16 ctrl_data_out_enable; + /**< Specifies whether the TDM block shares the data-out signal to the + drive with other masters. + @values + - #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE + - #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend */ + + u16 ctrl_invert_sync_pulse; + /**< Specifies whether to invert the sync or not. + @values + - #AFE_PORT_TDM_SYNC_NORMAL + - #AFE_PORT_TDM_SYNC_INVERT @tablebulletend */ + + u16 ctrl_sync_data_delay; + /**< Specifies the number of bit clock to delay data with respect to + sync edge. + @values + - #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE + - #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE + - #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend */ + + u16 slot_width; + /**< Slot width of the slot in a TDM frame. (slot_width >= bit_width) + have to be satisfied. + @values 16, 24, 32 */ + + u32 slot_mask; + /**< Position of active slots. When that bit is set, + that paricular slot is active. + Number of active slots can be inferred by number of + bits set in the mask. Only 8 individual bits can be enabled. + Bits 0..31 corresponding to slot 0..31 + @values 1 to 2^32 - 1 */ +} __packed; + +/** ID of Time Divsion Multiplexing (TDM) module, + which is used for configuring the AFE TDM. + + This module supports following parameter IDs: + - #AFE_PORT_TDM_SLOT_CONFIG + + To configure the TDM interface, the client must use the + #AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the + respective parameter IDs as listed above. +*/ + +#define AFE_MODULE_TDM 0x0001028A + +/** ID of the parameter used by #AFE_MODULE_TDM to configure + the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. +*/ +#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 + +/** Version information used to handle future additions to slot mapping + configuration (for backward compatibility). +*/ +#define AFE_API_VERSION_SLOT_MAPPING_CONFIG 0x1 + +/** Version information used to handle future additions to slot mapping +* configuration support 32 channels. +*/ +#define AFE_API_VERSION_SLOT_MAPPING_CONFIG_V2 0x2 + +/** Data align type */ +#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB 0 +#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB 1 + +#define AFE_SLOT_MAPPING_OFFSET_INVALID 0xFFFF + +/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG + command's TDM configuration parameter. +*/ +struct afe_param_id_slot_mapping_cfg { + u32 minor_version; + /**< Minor version used for tracking TDM slot configuration. + @values #AFE_API_VERSION_TDM_SLOT_CONFIG */ + + u16 num_channel; + /**< number of channel of the audio sample. + @values 1, 2, 4, 6, 8 @tablebulletend */ + + u16 bitwidth; + /**< Slot bit width for each channel + @values 16, 24, 32 */ + + u32 data_align_type; + /**< indicate how data packed from slot_offset for 32 slot bit width + in case of sample bit width is 24. + @values + #AFE_SLOT_MAPPING_DATA_ALIGN_MSB + #AFE_SLOT_MAPPING_DATA_ALIGN_LSB */ + + u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT]; + /**< Array of the slot mapping start offset in bytes for this frame. + The bytes is counted from 0. The 0 is mapped to the 1st byte + in or out of the digital serial data line this sub-frame belong to. + slot_offset[] setting is per-channel based. + The max num of channel supported is 8. + The valid offset value must always be continuly placed in from index 0. + Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays. + If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24, + "data_align_type" is used to indicate how 24 bit sample data in aligning + with 32 bit slot width per-channel. + @values, in byte*/ +} __packed; + +/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG_V2 +* command's TDM configuration parameter. +*/ +struct afe_param_id_slot_mapping_cfg_v2 { + u32 minor_version; + /**< Minor version used for tracking TDM slot configuration. + * @values #AFE_API_VERSION_TDM_SLOT_CONFIG + */ + + u16 num_channel; + /**< number of channel of the audio sample. + * @values 1, 2, 4, 6, 8, 16, 32 @tablebulletend + */ + + u16 bitwidth; + /**< Slot bit width for each channel + * @values 16, 24, 32 + */ + + u32 data_align_type; + /**< indicate how data packed from slot_offset for 32 slot bit width + * in case of sample bit width is 24. + * @values + * #AFE_SLOT_MAPPING_DATA_ALIGN_MSB + * #AFE_SLOT_MAPPING_DATA_ALIGN_LSB + */ + + u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT_V2]; + /**< Array of the slot mapping start offset in bytes for this frame. + * The bytes is counted from 0. The 0 is mapped to the 1st byte + * in or out of the digital serial data line this sub-frame belong to. + * slot_offset[] setting is per-channel based. + * The max num of channel supported is 8. + * The valid offset value must always be continuly placed in + * from index 0. + * Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays. + * If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24, + * "data_align_type" is used to indicate how 24 bit sample data in + * aligning with 32 bit slot width per-channel. + * @values, in byte + */ +} __packed; + +/** ID of the parameter used by #AFE_MODULE_TDM to configure + the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID. +*/ +#define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG 0x00010298 + +/** Version information used to handle future additions to custom TDM header + configuration (for backward compatibility). +*/ +#define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG 0x1 + +#define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID 0x0 +#define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT 0x1 +#define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST 0x2 + +#define AFE_CUSTOM_TDM_HEADER_MAX_CNT 0x8 + +/** Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID +*/ +struct afe_param_id_custom_tdm_header_cfg { + u32 minor_version; + /**< Minor version used for tracking custom TDM header configuration. + @values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG */ + + u16 start_offset; + /**< the slot mapping start offset in bytes from this sub-frame + The bytes is counted from 0. The 0 is mapped to the 1st byte in or out of + the digital serial data line this sub-frame belong to. + @values, in byte, + supported values are 0, 4, 8, */ + + u16 header_width; + /**< the header width per-frame followed. + 2 bytes for MOST/TDM case + @values, in byte + supported value is 2 */ + + u16 header_type; + /**< Indicate what kind of custom TDM header it is. + @values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0 + #AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1 (for AAN channel per MOST) + #AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2 + (for entertainment channel, which will overwrite + AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST) */ + + u16 num_frame_repeat; + /**< num of header followed. + @values, supported value is 8*/ + u16 header[AFE_CUSTOM_TDM_HEADER_MAX_CNT]; + /** < SAD header for MOST/TDM case is followed as payload as below. + The size of followed SAD header in bytes is num_of_frame_repeat * header_width_per_frame + which is 2 * 8 = 16 bytes here. + the supported payload format is in uint16_t as below + uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00 + uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218 + uint16_t header2; DTCP Info Info[6] - unused -> 0x0 + uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0 + uint16_t header4; Reserved Info[0] - Num of bytes following -> 0x7 + uint16_t header5; Reserved Info[1] - Media Type -> 0x0 + uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0 + uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low Byte -> 0x0 */ +} __packed; + +struct afe_tdm_port_config { + struct afe_param_id_tdm_cfg tdm; + struct afe_param_id_slot_mapping_cfg slot_mapping; + struct afe_param_id_slot_mapping_cfg_v2 slot_mapping_v2; + struct afe_param_id_custom_tdm_header_cfg custom_tdm_header; +} __packed; + +#define AFE_PARAM_ID_DEVICE_HW_DELAY 0x00010243 +#define AFE_API_VERSION_DEVICE_HW_DELAY 0x1 + +struct afe_param_id_device_hw_delay_cfg { + uint32_t device_hw_delay_minor_version; + uint32_t delay_in_us; +} __packed; + +#define AFE_PARAM_ID_SET_TOPOLOGY 0x0001025A +#define AFE_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; + +/* + * This command is used by client to request the LPASS resources. + * Currently this command supports only LPAIF DMA resources. + * Allocated resources will be in control of remote client until + * they get released. + * + * If all the requested resources are available then response status in + * AFE_CMDRSP_REQUEST_LPASS_RESOURCES payload will + * be updated with ADSP_EOK, otherwise it will be ADSP_EFAILED. + * + * This command is variable payload size command, and size depends + * on the type of resource requested. + * + * For example, if client requests AFE_LPAIF_DMA_RESOURCE_ID + * resources, afe_cmd_request_lpass_resources structure will + * be followed with the afe_cmd_request_lpass_dma_resources + * structure. + * + * AFE_CMDRSP_REQUEST_LPASS_RESOURCES is the response for + * this command, which returns the allocated resources. + * + * @apr_hdr_fields + * Opcode -- AFE_CMD_REQUEST_LPASS_RESOURCES + * + * @return + * #AFE_CMDRSP_REQUEST_LPASS_RESOURCES + */ +#define AFE_CMD_REQUEST_LPASS_RESOURCES 0x00010109 + +/* Macro for requesting LPAIF DMA resources */ +#define AFE_LPAIF_DMA_RESOURCE_ID 0x00000001 + +struct afe_cmd_request_lpass_resources { + /* + * LPASS Resource ID + * @values: + * - AFE_LPAIF_DMA_RESOURCE_ID + */ + u32 resource_id; +} __packed; + +/* + * AFE_CMD_REQUEST_LPASS_RESOURCES uses this structure when + * client is requesting LPAIF DMA resources. + * + * Number of read DMA channels and write DMA channels varies from chipset to + * chipset. HLOS needs to make sure that when it requests LPASS DMA + * resources, it should not impact the concurrencies which + * are mandatory for a given chipset. + */ + +/* Macro for AFE LPAIF default DMA data type */ +#define AFE_LPAIF_DEFAULT_DMA_TYPE 0x0 + +struct afe_cmd_request_lpass_dma_resources { + /* + * LPASS DMA Type + * @values: + * - AFE_LPAIF_DEFAULT_DMA_TYPE + */ + u8 dma_type; + /* + * Number of read DMA channels required + * @values: >=0 + * - 0 indicates channels are not requested + */ + u8 num_read_dma_channels; + /* + * Number of write DMA channels required + * @values: >=0 + * - 0 indicates channels are not requested + */ + u8 num_write_dma_channels; + /* + * Reserved field for 4 byte alignment + * @values: 0 + */ + u8 reserved; +} __packed; + +struct afe_request_lpass_dma_resources_command { + struct apr_hdr hdr; + struct afe_cmd_request_lpass_resources resources; + struct afe_cmd_request_lpass_dma_resources dma_resources; +} __packed; + +/* + * This is the response for the command AFE_CMD_REQUEST_LPASS_RESOURCES. + * Payload of this command is variable. + * + * Resources allocated successfully or not, are determined by the "status" + * in the payload. If status is ADSP_EOK, then resources are + * allocated successfully and allocated resource information + * follows. + * + * For example, if the response resource id is AFE_LPAIF_DMA_RESOURCE_ID, + * afe_cmdrsp_request_lpass_dma_resources structure will + * follow after afe_cmdrsp_request_lpass_resources. + * + * If status is ADSP_EFAILED, this indicates requested resources + * are not allocated successfully. In this case the payload following + * this structure is invalid. + * @apr_hdr_fields + * Opcode -- AFE_CMDRSP_REQUEST_LPASS_RESOURCES +*/ +#define AFE_CMDRSP_REQUEST_LPASS_RESOURCES 0x0001010A + +struct afe_cmdrsp_request_lpass_resources { + /* + * ADSP_EOK if all requested resources are allocated. + * ADSP_EFAILED if resource allocation is failed. + */ + u32 status; + /* + * Returned LPASS DMA resource ID + * @values: + * - AFE_LPAIF_DMA_RESOURCE_ID + */ + u32 resource_id; +} __packed; + +/* + * This command will be sent as a payload for + * AFE_CMDRSP_REQUEST_LPASS_RESOURCES, when the LPAIF DMA resources + * were requested. Payload of this command is variable, which + * follows after the afe_cmdrsp_request_lpass_dma_resources structure. + * The size in bytes following this structure is sum of + * num_read_dma_channels and num_write_dma_channels. + * + * If the resource allocation is successful, then the payload contains + * the valid DMA channel indices. + * + * For example, if number of requested DMA read channels is 2, and they + * are successfully allocated, the variable payload contains + * valid DMA channel index values in first two bytes array. + * + * In the failure case this payload can be ignored, and all the values will be + * initialized with zeros. + * + * An example payload of the command response is below: + * <struct afe_cmdrsp_request_lpass_resources> + * <struct afe_cmdrsp_request_lpass_dma_resources> + * read DMA index value for each byte. + * write DMA index value for each byte. + * padded zeros, if sum of num_read_dma_channels and num_write_dma_channels + * are not multiples of 4. +*/ + +struct afe_cmdrsp_request_lpass_dma_resources { + /* + * LPASS DMA Type + * @values: + * - AFE_LPAIF_DEFAULT_DMA_TYPE + */ + u8 dma_type; + /* + * Returned number of read DMA channels allocated + * @values: >=0 + */ + u8 num_read_dma_channels; + /* + * Returned number of write DMA channels allocated + * @values: >=0 + */ + u8 num_write_dma_channels; + /* + * Reserved field for 4 byte alignment + * @values: 0 + */ + u8 reserved; +} __packed; + +/* + * This command is for releasing resources which are allocated as + * part of AFE_CMD_REQUEST_LPASS_RESOURCES. + * + * Payload of this command is variable, which follows + * after the afe_cmd_release_lpass_resources structure. + * + * If release resource is AFE_LPAIF_DMA_RESOURCE_ID + * afe_cmd_release_lpass_dma_resources structure will be + * followed after afe_cmd_release_lpass_resources. + * + * + * @apr_hdr_fields + * Opcode -- AFE_CMD_RELEASE_LPASS_RESOURCES + + * @return + * #APRv2 IBASIC RSP Result +*/ +#define AFE_CMD_RELEASE_LPASS_RESOURCES 0x0001010B + +struct afe_cmd_release_lpass_resources { + /* + * LPASS DMA resource ID + * @values: + * - AFE_LPAIF_DMA_RESOURCE_ID + */ + u32 resource_id; +} __packed; + +/* + * This payload to be appended as part of AFE_CMD_RELEASE_LPASS_RESOURCES + * when resource id AFE_LPAIF_DMA_RESOURCE_ID is used. + * + * Payload of this command is variable, which will be followed after the + * afe_cmd_release_lpass_dma_resources structure. + * The variable payload's size in bytes is sum of + * num_read_dma_channels and num_write_dma_channels. + * Variable payload data contains the valid DMA channel indices which are + * allocated as part of AFE_CMD_REQUEST_LPASS_RESOURCES. + * + * For example, if number of DMA read channels released are 2, + * the variable payload contains valid DMA channel + * index values in first two bytes of variable payload. + * Client needs to fill the same DMA channel indices were returned + * as part of AFE_CMD_RELEASE_LPASS_RESOURCES, otherwise + * ADSP will return the error. + * + * An example payload of the release command is below: + * <struct afe_cmd_release_lpass_resources> + * <struct afe_cmd_release_lpass_dma_resources> + * read DMA index value for each byte. + * write DMA index value for each byte. +*/ + +struct afe_cmd_release_lpass_dma_resources { + /* + * LPASS DMA Type + * @values: + * - AFE_LPAIF_DEFAULT_DMA_TYPE + */ + u8 dma_type; + /* + * Number of read DMA channels to be released + * @values: >=0 + * - 0 indicates channels are not released + */ + u8 num_read_dma_channels; + /* + * Number of write DMA channels to be released + * @values: >=0 + * - 0 indicates channels are not released + */ + u8 num_write_dma_channels; + /* + * Reserved field for 4 byte alignment + * @values: 0 + */ + u8 reserved; +} __packed; + +struct afe_release_lpass_dma_resources_command { + struct apr_hdr hdr; + struct afe_cmd_release_lpass_resources resources; + struct afe_cmd_release_lpass_dma_resources dma_resources; +} __packed; + +/* + * Generic encoder module ID. + * This module supports the following parameter IDs: + * #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time) + * #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time) + * #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time) + * #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time) + * Opcode - AVS_MODULE_ID_ENCODER + * AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID. + */ +#define AFE_MODULE_ID_ENCODER 0x00013229 + +/* Macro for defining the packetizer ID: COP. */ +#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A + +/* + * Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module. + * This parameter cannot be set runtime. + */ +#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E + +/* + * Encoder config block parameter for the #AVS_MODULE_ID_ENCODER module. + * This parameter may be set runtime. + */ +#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C + +/* + * Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module. + * This parameter cannot be set runtime. + */ +#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID 0x0001322B + +/* + * Data format to send compressed data + * is transmitted/received over Slimbus lines. + */ +#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED 0x3 + +/* + * ID for AFE port module. This will be used to define port properties. + * This module supports following parameter IDs: + * #AFE_PARAM_ID_PORT_MEDIA_TYPE + * To configure the port property, the client must use the + * #AFE_PORT_CMD_SET_PARAM_V2 command, + * and fill the module ID with the respective parameter IDs as listed above. + * @apr_hdr_fields + * Opcode -- AFE_MODULE_PORT + */ +#define AFE_MODULE_PORT 0x000102a6 + +/* + * ID of the parameter used by #AFE_MODULE_PORT to set the port media type. + * parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command. + */ +#define AFE_PARAM_ID_PORT_MEDIA_TYPE 0x000102a7 + +/* + * Macros for defining the "data_format" field in the + * #AFE_PARAM_ID_PORT_MEDIA_TYPE + */ +#define AFE_PORT_DATA_FORMAT_PCM 0x0 +#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED 0x1 + +/* + * Macro for defining the "minor_version" field in the + * #AFE_PARAM_ID_PORT_MEDIA_TYPE + */ +#define AFE_API_VERSION_PORT_MEDIA_TYPE 0x1 + +#define ASM_MEDIA_FMT_NONE 0x0 + +/* + * Media format ID for SBC encode configuration. + * @par SBC encode configuration (asm_sbc_enc_cfg_t) + * @table{weak__asm__sbc__enc__cfg__t} + */ +#define ASM_MEDIA_FMT_SBC 0x00010BF2 + +/* SBC channel Mono mode.*/ +#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO 1 + +/* SBC channel Stereo mode. */ +#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO 2 + +/* SBC channel Dual Mono mode. */ +#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO 8 + +/* SBC channel Joint Stereo mode. */ +#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9 + +/* SBC bit allocation method = loudness. */ +#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0 + +/* SBC bit allocation method = SNR. */ +#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1 + + +/* + * Payload of the SBC encoder configuration parameters in the + * #ASM_MEDIA_FMT_SBC media format. + */ +struct asm_sbc_enc_cfg_t { + /* + * Number of subbands. + * @values 4, 8 + */ + uint32_t num_subbands; + + /* + * Size of the encoded block in samples. + * @values 4, 8, 12, 16 + */ + uint32_t blk_len; + + /* + * Mode used to allocate bits between channels. + * @values + * 0 (Native mode) + * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO + * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO + * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO + * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO + * Native mode indicates that encoding must be performed with the number + * of channels at the input. + * If postprocessing outputs one-channel data, Mono mode is used. If + * postprocessing outputs two-channel data, Stereo mode is used. + * The number of channels must not change during encoding. + */ + uint32_t channel_mode; + + /* + * Encoder bit allocation method. + * @values + * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS + * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend + */ + uint32_t alloc_method; + + /* + * Number of encoded bits per second. + * @values + * Mono channel -- Maximum of 320 kbps + * Stereo channel -- Maximum of 512 kbps @tablebulletend + */ + uint32_t bit_rate; + + /* + * Number of samples per second. + * @values 0 (Native mode), 16000, 32000, 44100, 48000 Hz + * Native mode indicates that encoding must be performed with the + * sampling rate at the input. + * The sampling rate must not change during encoding. + */ + uint32_t sample_rate; +}; + +#define ASM_MEDIA_FMT_AAC_AOT_LC 2 +#define ASM_MEDIA_FMT_AAC_AOT_SBR 5 +#define ASM_MEDIA_FMT_AAC_AOT_PS 29 +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0 +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3 + +struct asm_aac_enc_cfg_v2_t { + + /* Encoding rate in bits per second.*/ + uint32_t bit_rate; + + /* + * Encoding mode. + * Supported values: + * #ASM_MEDIA_FMT_AAC_AOT_LC + * #ASM_MEDIA_FMT_AAC_AOT_SBR + * #ASM_MEDIA_FMT_AAC_AOT_PS + */ + uint32_t enc_mode; + + /* + * AAC format flag. + * Supported values: + * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS + * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW + */ + uint16_t aac_fmt_flag; + + /* + * Number of channels to encode. + * Supported values: + * 0 - Native mode + * 1 - Mono + * 2 - Stereo + * Other values are not supported. + * @note1hang The eAAC+ encoder mode supports only stereo. + * Native mode indicates that encoding must be performed with the + * number of channels at the input. + * The number of channels must not change during encoding. + */ + uint16_t channel_cfg; + + /* + * Number of samples per second. + * Supported values: - 0 -- Native mode - For other values, + * Native mode indicates that encoding must be performed with the + * sampling rate at the input. + * The sampling rate must not change during encoding. + */ + uint32_t sample_rate; +} __packed; + +/* FMT ID for apt-X Classic */ +#define ASM_MEDIA_FMT_APTX 0x000131ff + +/* FMT ID for apt-X HD */ +#define ASM_MEDIA_FMT_APTX_HD 0x00013200 + +#define PCM_CHANNEL_L 1 +#define PCM_CHANNEL_R 2 +#define PCM_CHANNEL_C 3 + +struct asm_custom_enc_cfg_aptx_t { + uint32_t sample_rate; + /* Mono or stereo */ + uint16_t num_channels; + uint16_t reserved; + /* num_ch == 1, then PCM_CHANNEL_C, + * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R} + */ + uint8_t channel_mapping[8]; + uint32_t custom_size; +} __packed; + +struct afe_enc_fmt_id_param_t { + /* + * Supported values: + * #ASM_MEDIA_FMT_SBC + * #ASM_MEDIA_FMT_AAC_V2 + * Any OpenDSP supported values + */ + uint32_t fmt_id; +} __packed; + +struct afe_port_media_type_t { + /* + * Minor version + * @values #AFE_API_VERSION_PORT_MEDIA_TYPE. + */ + uint32_t minor_version; + + /* + * Sampling rate of the port. + * @values + * #AFE_PORT_SAMPLE_RATE_8K + * #AFE_PORT_SAMPLE_RATE_11_025K + * #AFE_PORT_SAMPLE_RATE_12K + * #AFE_PORT_SAMPLE_RATE_16K + * #AFE_PORT_SAMPLE_RATE_22_05K + * #AFE_PORT_SAMPLE_RATE_24K + * #AFE_PORT_SAMPLE_RATE_32K + * #AFE_PORT_SAMPLE_RATE_44_1K + * #AFE_PORT_SAMPLE_RATE_48K + * #AFE_PORT_SAMPLE_RATE_88_2K + * #AFE_PORT_SAMPLE_RATE_96K + * #AFE_PORT_SAMPLE_RATE_176_4K + * #AFE_PORT_SAMPLE_RATE_192K + * #AFE_PORT_SAMPLE_RATE_352_8K + * #AFE_PORT_SAMPLE_RATE_384K + */ + uint32_t sample_rate; + + /* + * Bit width of the sample. + * @values 16, 24 + */ + uint16_t bit_width; + + /* + * Number of channels. + * @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT + */ + uint16_t num_channels; + + /* + * Data format supported by this port. + * If the port media type and device media type are different, + * it signifies a encoding/decoding use case + * @values + * #AFE_PORT_DATA_FORMAT_PCM + * #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED + */ + uint16_t data_format; + + /*This field must be set to zero.*/ + uint16_t reserved; +} __packed; + +union afe_enc_config_data { + struct asm_sbc_enc_cfg_t sbc_config; + struct asm_aac_enc_cfg_v2_t aac_config; + struct asm_custom_enc_cfg_aptx_t aptx_config; +}; + +struct afe_enc_config { + u32 format; + union afe_enc_config_data data; +}; + +struct afe_enc_cfg_blk_param_t { + uint32_t enc_cfg_blk_size; + /* + *Size of the encoder configuration block that follows this member + */ + union afe_enc_config_data enc_blk_config; +}; + +/* + * Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter. + */ +struct avs_enc_packetizer_id_param_t { + /* + * Supported values: + * #AVS_MODULE_ID_PACKETIZER_COP + * Any OpenDSP supported values + */ + uint32_t enc_packetizer_id; +}; + +union afe_port_config { + struct afe_param_id_pcm_cfg pcm; + struct afe_param_id_i2s_cfg i2s; + struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; + struct afe_param_id_slimbus_cfg slim_sch; + struct afe_param_id_rt_proxy_port_cfg rtproxy; + struct afe_param_id_internal_bt_fm_cfg int_bt_fm; + struct afe_param_id_pseudo_port_cfg pseudo_port; + struct afe_param_id_device_hw_delay_cfg hw_delay; + struct afe_param_id_spdif_cfg spdif; + struct afe_param_id_set_topology_cfg topology; + struct afe_param_id_tdm_cfg tdm; + struct afe_param_id_usb_audio_cfg usb_audio; + struct afe_enc_fmt_id_param_t enc_fmt; + struct afe_port_media_type_t media_type; + struct afe_enc_cfg_blk_param_t enc_blk_param; + struct avs_enc_packetizer_id_param_t enc_pkt_id_param; +} __packed; + +#define AFE_PORT_CMD_DEVICE_START 0x000100E5 + +/* Payload of the #AFE_PORT_CMD_DEVICE_START.*/ +struct afe_port_cmd_device_start { + struct apr_hdr hdr; + u16 port_id; +/* Port interface and direction (Rx or Tx) to start. An even + * number represents the Rx direction, and an odd number represents + * the Tx direction. + */ + + + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0.*/ + +} __packed; + +#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6 + +/* Payload of the #AFE_PORT_CMD_DEVICE_STOP. +*/ +struct afe_port_cmd_device_stop { + struct apr_hdr hdr; + u16 port_id; +/* Port interface and direction (Rx or Tx) to start. An even + * number represents the Rx direction, and an odd number represents + * the Tx direction. + */ + + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0.*/ +} __packed; + +#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA + +/* Memory map regions command payload used by the + * #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS . + * This structure allows clients to map multiple shared memory + * regions in a single command. Following this structure are + * num_regions of afe_service_shared_map_region_payload. + */ +struct afe_service_cmd_shared_mem_map_regions { + struct apr_hdr hdr; +u16 mem_pool_id; +/* Type of memory on which this memory region is mapped. + * Supported values: + * - #ADSP_MEMORY_MAP_EBI_POOL + * - #ADSP_MEMORY_MAP_SMI_POOL + * - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL + * - Other values are reserved + * + * The memory pool ID implicitly defines the characteristics of the + * memory. Characteristics may include alignment type, permissions, + * etc. + * + * ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory + * ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory + * ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte + * addressable, and 4 KB aligned. + */ + + + u16 num_regions; +/* Number of regions to map. + * Supported values: + * - Any value greater than zero + */ + + u32 property_flag; +/* Configures one common property for all the regions in the + * payload. + * + * Supported values: - 0x00000000 to 0x00000001 + * + * b0 - bit 0 indicates physical or virtual mapping 0 Shared memory + * address provided in afe_service_shared_map_region_payloadis a + * physical address. The shared memory needs to be mapped( hardware + * TLB entry) and a software entry needs to be added for internal + * book keeping. + * + * 1 Shared memory address provided in + * afe_service_shared_map_region_payloadis a virtual address. The + * shared memory must not be mapped (since hardware TLB entry is + * already available) but a software entry needs to be added for + * internal book keeping. This can be useful if two services with in + * ADSP is communicating via APR. They can now directly communicate + * via the Virtual address instead of Physical address. The virtual + * regions must be contiguous. num_regions must be 1 in this case. + * + * b31-b1 - reserved bits. must be set to zero + */ + + +} __packed; +/* Map region payload used by the + * afe_service_shared_map_region_payloadstructure. + */ +struct afe_service_shared_map_region_payload { + u32 shm_addr_lsw; +/* least significant word of starting address in the memory + * region to map. It must be contiguous memory, and it must be 4 KB + * aligned. + * Supported values: - Any 32 bit value + */ + + + u32 shm_addr_msw; +/* most significant word of startng address in the memory region + * to map. For 32 bit shared memory address, this field must be set + * to zero. For 36 bit shared memory address, bit31 to bit 4 must be + * set to zero + * + * Supported values: - For 32 bit shared memory address, this field + * must be set to zero. - For 36 bit shared memory address, bit31 to + * bit 4 must be set to zero - For 64 bit shared memory address, any + * 32 bit value + */ + + + u32 mem_size_bytes; +/* Number of bytes in the region. The aDSP will always map the + * regions as virtual contiguous memory, but the memory size must be + * in multiples of 4 KB to avoid gaps in the virtually contiguous + * mapped memory. + * + * Supported values: - multiples of 4KB + */ + +} __packed; + +#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB +struct afe_service_cmdrsp_shared_mem_map_regions { + u32 mem_map_handle; +/* A memory map handle encapsulating shared memory attributes is + * returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is + * successful. In the case of failure , a generic APR error response + * is returned to the client. + * + * Supported Values: - Any 32 bit value + */ + +} __packed; +#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC +/* Memory unmap regions command payload used by the + * #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS + * + * This structure allows clients to unmap multiple shared memory + * regions in a single command. + */ + + +struct afe_service_cmd_shared_mem_unmap_regions { + struct apr_hdr hdr; +u32 mem_map_handle; +/* memory map handle returned by + * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands + * + * Supported Values: + * - Any 32 bit value + */ +} __packed; + +/* Used by RTAC */ +struct afe_rtac_get_param_v2 { + u16 port_id; +/* Port interface and direction (Rx or Tx) to start. */ + + u16 payload_size; +/* Maximum data size of the parameter ID/module ID combination. + * This is a multiple of four bytes + * Supported values: > 0 + */ + + u32 payload_address_lsw; +/* LSW of 64 bit Payload address. Address should be 32-byte, + * 4kbyte aligned and must be contig memory. + */ + + + u32 payload_address_msw; +/* MSW of 64 bit Payload address. In case of 32-bit shared + * memory address, this field must be set to zero. In case of 36-bit + * shared memory address, bit-4 to bit-31 must be set to zero. + * Address should be 32-byte, 4kbyte aligned and must be contiguous + * memory. + */ + + u32 mem_map_handle; +/* Memory map handle returned by + * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands. + * Supported Values: - NULL -- Message. The parameter data is + * in-band. - Non-NULL -- The parameter data is Out-band.Pointer to + * - the physical address in shared memory of the payload data. + * For detailed payload content, see the afe_port_param_data_v2 + * structure + */ + + + u32 module_id; +/* ID of the module to be queried. + * Supported values: Valid module ID + */ + + u32 param_id; +/* ID of the parameter to be queried. + * Supported values: Valid parameter ID + */ +} __packed; + +#define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0 + +/* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command, + * which queries for one post/preprocessing parameter of a + * stream. + */ +struct afe_port_cmd_get_param_v2 { + struct apr_hdr apr_hdr; + + /* Port interface and direction (Rx or Tx) to start. */ + u16 port_id; + + /* Maximum data size of the parameter ID/module ID combination. + * This is a multiple of four bytes + * Supported values: > 0 + */ + u16 payload_size; + + /* The memory mapping header to be used when requesting outband */ + struct mem_mapping_hdr mem_hdr; + + /* The module ID of the parameter data requested */ + u32 module_id; + + /* The parameter ID of the parameter data requested */ + u32 param_id; + + /* The header information for the parameter data */ + struct param_hdr_v1 param_hdr; +} __packed; + +#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106 + +/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which + * responds to an #AFE_PORT_CMD_GET_PARAM_V2 command. + * + * Immediately following this structure is the parameters structure + * (afe_port_param_data) containing the response(acknowledgment) + * parameter payload. This payload is included for an in-band + * scenario. For an address/shared memory-based set parameter, this + * payload is not needed. + */ + + +struct afe_port_cmdrsp_get_param_v2 { + u32 status; + struct param_hdr_v1 param_hdr; + u8 param_data[0]; +} __packed; + +#define AFE_PORT_CMD_GET_PARAM_V3 0x000100FB +struct afe_port_cmd_get_param_v3 { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* Port ID of the AFE port to configure. Port interface and direction + * (Rx or Tx) to configure. An even number represents the Rx direction, + * and an odd number represents the Tx direction. + */ + u16 port_id; + + /* Reserved. This field must be set to zero. */ + u16 reserved; + + /* The memory mapping header to be used when requesting outband */ + struct mem_mapping_hdr mem_hdr; + + /* The header information for the parameter data */ + struct param_hdr_v3 param_hdr; +} __packed; + +#define AFE_PORT_CMDRSP_GET_PARAM_V3 0x00010108 +struct afe_port_cmdrsp_get_param_v3 { + /* The status of the command */ + uint32_t status; + + /* The header information for the parameter data */ + struct param_hdr_v3 param_hdr; + + /* The parameter data to be filled when sent inband */ + u8 param_data[0]; +} __packed; + +#define AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG 0x0001028C +#define AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG 0x1 +/* + * Payload of the AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG parameter used by + * AFE_MODULE_AUDIO_DEV_INTERFACE. +*/ +struct afe_param_id_lpass_core_shared_clk_cfg { + u32 lpass_core_shared_clk_cfg_minor_version; +/* + * Minor version used for lpass core shared clock configuration + * Supported value: AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG + */ + u32 enable; +/* + * Specifies whether the lpass core shared clock is + * enabled (1) or disabled (0). + */ +} __packed; + +/* adsp_afe_service_commands.h */ + +#define ADSP_MEMORY_MAP_EBI_POOL 0 + +#define ADSP_MEMORY_MAP_SMI_POOL 1 +#define ADSP_MEMORY_MAP_IMEM_POOL 2 +#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3 +/* +* Definition of virtual memory flag +*/ +#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1 + +/* +* Definition of physical memory flag +*/ +#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0 + +#define NULL_POPP_TOPOLOGY 0x00010C68 +#define NULL_COPP_TOPOLOGY 0x00010312 +#define DEFAULT_COPP_TOPOLOGY 0x00010314 +#define DEFAULT_POPP_TOPOLOGY 0x00010BE4 +#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY 0x0001076B +#define COMPRESSED_PASSTHROUGH_NONE_TOPOLOGY 0x00010774 +#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71 +#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72 +#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75 +#define VPM_TX_DM_RFECNS_COPP_TOPOLOGY 0x00010F86 +#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_DTS_HPX 0x10015002 +#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_AUDIOSPHERE 0x10028000 + +/* Memory map regions command payload used by the + * #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS + * commands. + * + * This structure allows clients to map multiple shared memory + * regions in a single command. Following this structure are + * num_regions of avs_shared_map_region_payload. + */ + + +struct avs_cmd_shared_mem_map_regions { + struct apr_hdr hdr; + u16 mem_pool_id; +/* Type of memory on which this memory region is mapped. + * + * Supported values: - #ADSP_MEMORY_MAP_EBI_POOL - + * #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL + * (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values + * are reserved + * + * The memory ID implicitly defines the characteristics of the + * memory. Characteristics may include alignment type, permissions, + * etc. + * + * SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned. + */ + + + u16 num_regions; + /* Number of regions to map.*/ + + u32 property_flag; +/* Configures one common property for all the regions in the + * payload. No two regions in the same memory map regions cmd can + * have differnt property. Supported values: - 0x00000000 to + * 0x00000001 + * + * b0 - bit 0 indicates physical or virtual mapping 0 shared memory + * address provided in avs_shared_map_regions_payload is physical + * address. The shared memory needs to be mapped( hardware TLB + * entry) + * + * and a software entry needs to be added for internal book keeping. + * + * 1 Shared memory address provided in MayPayload[usRegions] is + * virtual address. The shared memory must not be mapped (since + * hardware TLB entry is already available) but a software entry + * needs to be added for internal book keeping. This can be useful + * if two services with in ADSP is communicating via APR. They can + * now directly communicate via the Virtual address instead of + * Physical address. The virtual regions must be contiguous. + * + * b31-b1 - reserved bits. must be set to zero + */ + +} __packed; + +struct avs_shared_map_region_payload { + u32 shm_addr_lsw; +/* least significant word of shared memory address of the memory + * region to map. It must be contiguous memory, and it must be 4 KB + * aligned. + */ + + u32 shm_addr_msw; +/* most significant word of shared memory address of the memory + * region to map. For 32 bit shared memory address, this field must + * tbe set to zero. For 36 bit shared memory address, bit31 to bit 4 + * must be set to zero + */ + + u32 mem_size_bytes; +/* Number of bytes in the region. + * + * The aDSP will always map the regions as virtual contiguous + * memory, but the memory size must be in multiples of 4 KB to avoid + * gaps in the virtually contiguous mapped memory. + */ + +} __packed; + +struct avs_cmd_shared_mem_unmap_regions { + struct apr_hdr hdr; + u32 mem_map_handle; +/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS + * , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands + */ + +} __packed; + +/* Memory map command response payload used by the + * #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS + * ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS + */ + + +struct avs_cmdrsp_shared_mem_map_regions { + u32 mem_map_handle; +/* A memory map handle encapsulating shared memory attributes is + * returned + */ + +} __packed; + +/*adsp_audio_memmap_api.h*/ + +/* ASM related data structures */ +struct asm_wma_cfg { + u16 format_tag; + u16 ch_cfg; + u32 sample_rate; + u32 avg_bytes_per_sec; + u16 block_align; + u16 valid_bits_per_sample; + u32 ch_mask; + u16 encode_opt; + u16 adv_encode_opt; + u32 adv_encode_opt2; + u32 drc_peak_ref; + u32 drc_peak_target; + u32 drc_ave_ref; + u32 drc_ave_target; +} __packed; + +struct asm_wmapro_cfg { + u16 format_tag; + u16 ch_cfg; + u32 sample_rate; + u32 avg_bytes_per_sec; + u16 block_align; + u16 valid_bits_per_sample; + u32 ch_mask; + u16 encode_opt; + u16 adv_encode_opt; + u32 adv_encode_opt2; + u32 drc_peak_ref; + u32 drc_peak_target; + u32 drc_ave_ref; + u32 drc_ave_target; +} __packed; + +struct asm_aac_cfg { + u16 format; + u16 aot; + u16 ep_config; + u16 section_data_resilience; + u16 scalefactor_data_resilience; + u16 spectral_data_resilience; + u16 ch_cfg; + u16 reserved; + u32 sample_rate; +} __packed; + +struct asm_amrwbplus_cfg { + u32 size_bytes; + u32 version; + u32 num_channels; + u32 amr_band_mode; + u32 amr_dtx_mode; + u32 amr_frame_fmt; + u32 amr_lsf_idx; +} __packed; + +struct asm_flac_cfg { + u32 sample_rate; + u32 ext_sample_rate; + u32 min_frame_size; + u32 max_frame_size; + u16 stream_info_present; + u16 min_blk_size; + u16 max_blk_size; + u16 ch_cfg; + u16 sample_size; + u16 md5_sum; +}; + +struct asm_alac_cfg { + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +}; + +struct asm_g711_dec_cfg { + u32 sample_rate; +}; + +struct asm_vorbis_cfg { + u32 bit_stream_fmt; +}; + +struct asm_ape_cfg { + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +}; + +struct asm_dsd_cfg { + u16 num_version; + u16 is_bitwise_big_endian; + u16 dsd_channel_block_size; + u16 num_channels; + u8 channel_mapping[8]; + u32 dsd_data_rate; +}; + +struct asm_softpause_params { + u32 enable; + u32 period; + u32 step; + u32 rampingcurve; +} __packed; + +struct asm_softvolume_params { + u32 period; + u32 step; + u32 rampingcurve; +} __packed; + +struct asm_stream_pan_ctrl_params { + uint16_t num_output_channels; + uint16_t num_input_channels; + uint16_t output_channel_map[8]; + uint16_t input_channel_map[8]; + uint32_t gain[64]; +} __packed; + +struct adm_matrix_ramp_gains_params { + uint16_t session_id; + uint16_t be_id; + uint16_t num_gains; + uint16_t path; + uint16_t channels; + uint16_t gain_value[32]; +} __packed; + +struct adm_matrix_mute_params { + uint16_t session_id; + uint16_t be_id; + uint16_t channels; + uint16_t path; + uint8_t mute_flag[32]; +} __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 + +/* Second low frequency channel. */ +#define PCM_CHANNEL_LFE2 17 + +/* Side left channel. */ +#define PCM_CHANNEL_SL 18 + +/* Side right channel. */ +#define PCM_CHANNEL_SR 19 + +/* Top front left channel. */ +#define PCM_CHANNEL_TFL 20 + +/* Left vertical height channel. */ +#define PCM_CHANNEL_LVH 20 + +/* Top front right channel. */ +#define PCM_CHANNEL_TFR 21 + +/* Right vertical height channel. */ +#define PCM_CHANNEL_RVH 21 + +/* Top center channel. */ +#define PCM_CHANNEL_TC 22 + +/* Top back left channel. */ +#define PCM_CHANNEL_TBL 23 + +/* Top back right channel. */ +#define PCM_CHANNEL_TBR 24 + +/* Top side left channel. */ +#define PCM_CHANNEL_TSL 25 + +/* Top side right channel. */ +#define PCM_CHANNEL_TSR 26 + +/* Top back center channel. */ +#define PCM_CHANNEL_TBC 27 + +/* Bottom front center channel. */ +#define PCM_CHANNEL_BFC 28 + +/* Bottom front left channel. */ +#define PCM_CHANNEL_BFL 29 + +/* Bottom front right channel. */ +#define PCM_CHANNEL_BFR 30 + +/* Left wide channel. */ +#define PCM_CHANNEL_LW 31 + +/* Right wide channel. */ +#define PCM_CHANNEL_RW 32 + +/* Left side direct channel. */ +#define PCM_CHANNEL_LSD 33 + +/* Right side direct channel. */ +#define PCM_CHANNEL_RSD 34 + +#define PCM_FORMAT_MAX_NUM_CHANNEL 8 + +#define PCM_FORMAT_MAX_NUM_CHANNEL_V2 32 + +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 + +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC + +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C + +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 0x00013222 + +#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF + +#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0 + +#define ASM_MEDIA_FMT_GENERIC_COMPRESSED 0x00013212 + +#define ASM_MAX_EQ_BANDS 12 + +#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98 + +struct asm_data_cmd_media_fmt_update_v2 { +u32 fmt_blk_size; + /* Media format block size in bytes.*/ +} __packed; + +struct asm_generic_compressed_fmt_blk_t { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + + /* + * Channel mapping array of bitstream output. + * Channel[i] mapping describes channel i inside the buffer, where + * i < num_channels. All valid used channels must be + * present at the beginning of the array. + */ + uint8_t channel_mapping[8]; + + /* + * Number of channels of the incoming bitstream. + * Supported values: 1,2,3,4,5,6,7,8 + */ + uint16_t num_channels; + + /* + * Nominal bits per sample value of the incoming bitstream. + * Supported values: 16, 32 + */ + uint16_t bits_per_sample; + + /* + * Nominal sampling rate of the incoming bitstream. + * Supported values: 8000, 11025, 16000, 22050, 24000, 32000, + * 44100, 48000, 88200, 96000, 176400, 192000, + * 352800, 384000 + */ + uint32_t sampling_rate; + +} __packed; + + +/* Command to send sample rate & channels for IEC61937 (compressed) or IEC60958 + * (pcm) streams. Both audio standards use the same format and are used for + * HDMI or SPDIF. + */ +#define ASM_DATA_CMD_IEC_60958_MEDIA_FMT 0x0001321E + +struct asm_iec_compressed_fmt_blk_t { + struct apr_hdr hdr; + + /* + * Nominal sampling rate of the incoming bitstream. + * Supported values: 8000, 11025, 16000, 22050, 24000, 32000, + * 44100, 48000, 88200, 96000, 176400, 192000, + * 352800, 384000 + */ + uint32_t sampling_rate; + + /* + * Number of channels of the incoming bitstream. + * Supported values: 1,2,3,4,5,6,7,8 + */ + uint32_t num_channels; + +} __packed; + +struct asm_multi_channel_pcm_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + + u16 num_channels; + /* Number of channels. Supported values: 1 to 8 */ + u16 bits_per_sample; +/* Number of bits per sample per channel. * Supported values: + * 16, 24 * When used for playback, the client must send 24-bit + * samples packed in 32-bit words. The 24-bit samples must be placed + * in the most significant 24 bits of the 32-bit word. When used for + * recording, the aDSP sends 24-bit samples packed in 32-bit words. + * The 24-bit samples are placed in the most significant 24 bits of + * the 32-bit word. + */ + + + u32 sample_rate; +/* Number of samples per second (in Hertz). + * Supported values: 2000 to 48000 + */ + + u16 is_signed; + /* Flag that indicates the samples are signed (1). */ + + u16 reserved; + /* reserved field for 32 bit alignment. must be set to zero. */ + + u8 channel_mapping[8]; +/* Channel array of size 8. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_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; + +struct asm_multi_channel_pcm_fmt_blk_v5 { + uint16_t num_channels; +/* + * Number of channels + * Supported values: 1 to 32 + */ + + uint16_t bits_per_sample; +/* + * Number of bits per sample per channel + * Supported values: 16, 24, 32 + */ + + uint32_t sample_rate; +/* + * Number of samples per second + * Supported values: 2000 to 48000, 96000,192000 Hz + */ + + uint16_t is_signed; +/* Flag that indicates that PCM samples are signed (1) */ + + uint16_t sample_word_size; +/* + * Size in bits of the word that holds a sample of a channel. + * Supported values: 12,24,32 + */ + uint16_t endianness; +/* + * Flag to indicate the endianness of the pcm sample + * Supported values: 0 - Little endian (all other formats) + * 1 - Big endian (AIFF) + */ + uint16_t mode; +/* + * Mode to provide additional info about the pcm input data. + * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b, + * Q31 for unpacked 24b or 32b) + * 15 - for 16 bit + * 23 - for 24b packed or 8.24 format + * 31 - for 24b unpacked or 32bit + */ + + uint8_t channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL_V2]; +/* + * Each element, i, in the array describes channel i inside the buffer where + * 0 <= i < num_channels. Unused channels are set to 0. + */ +} __packed; + +/* + * Payload of the multichannel PCM configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format. + */ +struct asm_multi_channel_pcm_fmt_blk_param_v3 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + struct asm_multi_channel_pcm_fmt_blk_v3 param; +} __packed; + +/* + * Payload of the multichannel PCM configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format. + */ +struct asm_multi_channel_pcm_fmt_blk_param_v4 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + struct asm_multi_channel_pcm_fmt_blk_v4 param; +} __packed; + +/* + * Payload of the multichannel PCM configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format. + */ +struct asm_multi_channel_pcm_fmt_blk_param_v5 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + struct asm_multi_channel_pcm_fmt_blk_v5 param; +} __packed; + +struct asm_stream_cmd_set_encdec_param { + u32 param_id; + /* ID of the parameter. */ + + u32 param_size; +/* Data size of this parameter, in bytes. The size is a multiple + * of 4 bytes. + */ + +} __packed; + +struct asm_enc_cfg_blk_param_v2 { + u32 frames_per_buf; +/* Number of encoded frames to pack into each buffer. + * + * @note1hang This is only guidance information for the aDSP. The + * number of encoded frames put into each buffer (specified by the + * client) is less than or equal to this number. + */ + + u32 enc_cfg_blk_size; +/* Size in bytes of the encoder configuration block that follows + * this member. + */ + +} __packed; + +/* @brief Dolby Digital Plus end point configuration structure + */ +struct asm_dec_ddp_endp_param_v2 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + int endp_param_value; +} __packed; + +/* + * Payload of the multichannel PCM encoder configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format. + */ +struct asm_multi_channel_pcm_enc_cfg_v5 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + uint16_t num_channels; + /* + * Number of PCM channels. + * @values + * - 0 -- Native mode + * - 1 -- 8 channels + * Native mode indicates that encoding must be performed with the number + * of channels at the input. + */ + uint16_t bits_per_sample; + /* + * Number of bits per sample per channel. + * @values 16, 24 + */ + uint32_t sample_rate; + /* + * Number of samples per second. + * @values 0, 8000 to 48000 Hz + * A value of 0 indicates the native sampling rate. Encoding is + * performed at the input sampling rate. + */ + uint16_t is_signed; + /* + * Flag that indicates the PCM samples are signed (1). Currently, only + * signed PCM samples are supported. + */ + uint16_t sample_word_size; + /* + * The size in bits of the word that holds a sample of a channel. + * @values 16, 24, 32 + * 16-bit samples are always placed in 16-bit words: + * sample_word_size = 1. + * 24-bit samples can be placed in 32-bit words or in consecutive + * 24-bit words. + * - If sample_word_size = 32, 24-bit samples are placed in the + * most significant 24 bits of a 32-bit word. + * - If sample_word_size = 24, 24-bit samples are placed in + * 24-bit words. @tablebulletend + */ + uint16_t endianness; + /* + * Flag to indicate the endianness of the pcm sample + * Supported values: 0 - Little endian (all other formats) + * 1 - Big endian (AIFF) + */ + uint16_t mode; + /* + * Mode to provide additional info about the pcm input data. + * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b, + * Q31 for unpacked 24b or 32b) + * 15 - for 16 bit + * 23 - for 24b packed or 8.24 format + */ + uint8_t channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL_V2]; + /* + * Channel mapping array expected at the encoder output. + * Channel[i] mapping describes channel i inside the buffer, where + * 0 @le i < num_channels. All valid used channels must be present at + * the beginning of the array. + * If Native mode is set for the channels, this field is ignored. + * @values See Section @xref{dox:PcmChannelDefs} + */ +} __packed; + +/* + * Payload of the multichannel PCM encoder configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format. + */ + +struct asm_multi_channel_pcm_enc_cfg_v4 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + uint16_t num_channels; + /* + * Number of PCM channels. + * @values + * - 0 -- Native mode + * - 1 -- 8 channels + * Native mode indicates that encoding must be performed with the number + * of channels at the input. + */ + uint16_t bits_per_sample; + /* + * Number of bits per sample per channel. + * @values 16, 24 + */ + uint32_t sample_rate; + /* + * Number of samples per second. + * @values 0, 8000 to 48000 Hz + * A value of 0 indicates the native sampling rate. Encoding is + * performed at the input sampling rate. + */ + uint16_t is_signed; + /* + * Flag that indicates the PCM samples are signed (1). Currently, only + * signed PCM samples are supported. + */ + uint16_t sample_word_size; + /* + * The size in bits of the word that holds a sample of a channel. + * @values 16, 24, 32 + * 16-bit samples are always placed in 16-bit words: + * sample_word_size = 1. + * 24-bit samples can be placed in 32-bit words or in consecutive + * 24-bit words. + * - If sample_word_size = 32, 24-bit samples are placed in the + * most significant 24 bits of a 32-bit word. + * - If sample_word_size = 24, 24-bit samples are placed in + * 24-bit words. @tablebulletend + */ + uint8_t channel_mapping[8]; + /* + * Channel mapping array expected at the encoder output. + * Channel[i] mapping describes channel i inside the buffer, where + * 0 @le i < num_channels. All valid used channels must be present at + * the beginning of the array. + * If Native mode is set for the channels, this field is ignored. + * @values See Section @xref{dox:PcmChannelDefs} + */ + uint16_t endianness; + /* + * Flag to indicate the endianness of the pcm sample + * Supported values: 0 - Little endian (all other formats) + * 1 - Big endian (AIFF) + */ + uint16_t mode; + /* + * Mode to provide additional info about the pcm input data. + * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b, + * Q31 for unpacked 24b or 32b) + * 15 - for 16 bit + * 23 - for 24b packed or 8.24 format + * 31 - for 24b unpacked or 32bit + */ +} __packed; + +/* + * Payload of the multichannel PCM encoder configuration parameters in + * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format. + */ + +struct asm_multi_channel_pcm_enc_cfg_v3 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + uint16_t num_channels; + /* + * Number of PCM channels. + * @values + * - 0 -- Native mode + * - 1 -- 8 channels + * Native mode indicates that encoding must be performed with the number + * of channels at the input. + */ + uint16_t bits_per_sample; + /* + * Number of bits per sample per channel. + * @values 16, 24 + */ + uint32_t sample_rate; + /* + * Number of samples per second. + * @values 0, 8000 to 48000 Hz + * A value of 0 indicates the native sampling rate. Encoding is + * performed at the input sampling rate. + */ + uint16_t is_signed; + /* + * Flag that indicates the PCM samples are signed (1). Currently, only + * signed PCM samples are supported. + */ + uint16_t sample_word_size; + /* + * The size in bits of the word that holds a sample of a channel. + * @values 16, 24, 32 + * 16-bit samples are always placed in 16-bit words: + * sample_word_size = 1. + * 24-bit samples can be placed in 32-bit words or in consecutive + * 24-bit words. + * - If sample_word_size = 32, 24-bit samples are placed in the + * most significant 24 bits of a 32-bit word. + * - If sample_word_size = 24, 24-bit samples are placed in + * 24-bit words. @tablebulletend + */ + uint8_t channel_mapping[8]; + /* + * Channel mapping array expected at the encoder output. + * Channel[i] mapping describes channel i inside the buffer, where + * 0 @le i < num_channels. All valid used channels must be present at + * the beginning of the array. + * If Native mode is set for the channels, this field is ignored. + * @values See Section @xref{dox:PcmChannelDefs} + */ +}; + +/* @brief Multichannel PCM encoder configuration structure used + * in the #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command. + */ + +struct asm_multi_channel_pcm_enc_cfg_v2 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + uint16_t num_channels; +/*< Number of PCM channels. + * + * Supported values: - 0 -- Native mode - 1 -- 8 Native mode + * indicates that encoding must be performed with the number of + * channels at the input. + */ + + uint16_t bits_per_sample; +/*< Number of bits per sample per channel. + * Supported values: 16, 24 + */ + + uint32_t sample_rate; +/*< Number of samples per second (in Hertz). + * + * Supported values: 0, 8000 to 48000 A value of 0 indicates the + * native sampling rate. Encoding is performed at the input sampling + * rate. + */ + + uint16_t is_signed; +/*< Specifies whether the samples are signed (1). Currently, + * only signed samples are supported. + */ + + uint16_t reserved; +/*< reserved field for 32 bit alignment. must be set to zero.*/ + + + uint8_t channel_mapping[8]; +} __packed; + +#define ASM_MEDIA_FMT_MP3 0x00010BE9 +#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6 + +/* @xreflabel + * {hdr:AsmMediaFmtDolbyAac} Media format ID for the + * Dolby AAC decoder. This format ID is be used if the client wants + * to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC + * contents. + */ + +#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86 + +/* Enumeration for the audio data transport stream AAC format. */ +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0 + +/* Enumeration for low overhead audio stream AAC format. */ +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS 1 + +/* Enumeration for the audio data interchange format + * AAC format. + */ +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF 2 + +/* Enumeration for the raw AAC format. */ +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3 + +/* Enumeration for the AAC LATM format. */ +#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM 4 + +#define ASM_MEDIA_FMT_AAC_AOT_LC 2 +#define ASM_MEDIA_FMT_AAC_AOT_SBR 5 +#define ASM_MEDIA_FMT_AAC_AOT_PS 29 +#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22 + +struct asm_aac_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + + u16 aac_fmt_flag; +/* Bitstream format option. + * Supported values: + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW + */ + + u16 audio_objype; +/* Audio Object Type (AOT) present in the AAC stream. + * Supported values: + * - #ASM_MEDIA_FMT_AAC_AOT_LC + * - #ASM_MEDIA_FMT_AAC_AOT_SBR + * - #ASM_MEDIA_FMT_AAC_AOT_BSAC + * - #ASM_MEDIA_FMT_AAC_AOT_PS + * - Otherwise -- Not supported + */ + + u16 channel_config; +/* Number of channels present in the AAC stream. + * Supported values: + * - 1 -- Mono + * - 2 -- Stereo + * - 6 -- 5.1 content + */ + + u16 total_size_of_PCE_bits; +/* greater or equal to zero. * -In case of RAW formats and + * channel config = 0 (PCE), client can send * the bit stream + * containing PCE immediately following this structure * (in-band). + * -This number does not include bits included for 32 bit alignment. + * -If zero, then the PCE info is assumed to be available in the + * audio -bit stream & not in-band. + */ + + u32 sample_rate; +/* Number of samples per second (in Hertz). + * + * Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000, + * 44100, 48000 + * + * This field must be equal to the sample rate of the AAC-LC + * decoder's output. - For MP4 or 3GP containers, this is indicated + * by the samplingFrequencyIndex field in the AudioSpecificConfig + * element. - For ADTS format, this is indicated by the + * samplingFrequencyIndex in the ADTS fixed header. - For ADIF + * format, this is indicated by the samplingFrequencyIndex in the + * program_config_element present in the ADIF header. + */ + +} __packed; + +struct asm_aac_enc_cfg_v2 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u32 bit_rate; + /* Encoding rate in bits per second. */ + u32 enc_mode; +/* Encoding mode. + * Supported values: + * - #ASM_MEDIA_FMT_AAC_AOT_LC + * - #ASM_MEDIA_FMT_AAC_AOT_SBR + * - #ASM_MEDIA_FMT_AAC_AOT_PS + */ + u16 aac_fmt_flag; +/* AAC format flag. + * Supported values: + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS + * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW + */ + u16 channel_cfg; +/* Number of channels to encode. + * Supported values: + * - 0 -- Native mode + * - 1 -- Mono + * - 2 -- Stereo + * - Other values are not supported. + * @note1hang The eAAC+ encoder mode supports only stereo. + * Native mode indicates that encoding must be performed with the + * number of channels at the input. + * The number of channels must not change during encoding. + */ + + u32 sample_rate; +/* Number of samples per second. + * Supported values: - 0 -- Native mode - For other values, + * Native mode indicates that encoding must be performed with the + * sampling rate at the input. + * The sampling rate must not change during encoding. + */ + +} __packed; + +#define ASM_MEDIA_FMT_G711_ALAW_FS 0x00010BF7 +#define ASM_MEDIA_FMT_G711_MLAW_FS 0x00010C2E + +struct asm_g711_enc_cfg_v2 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u32 sample_rate; +/* + * Number of samples per second. + * Supported values: 8000, 16000 Hz + */ + +} __packed; + +struct asm_vorbis_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + u32 bit_stream_fmt; +/* Bit stream format. + * Supported values: + * - 0 -- Raw bitstream + * - 1 -- Transcoded bitstream + * + * Transcoded bitstream containing the size of the frame as the first + * word in each frame. + */ + +} __packed; + +struct asm_flac_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + + u16 is_stream_info_present; +/* Specifies whether stream information is present in the FLAC format + * block. + * + * Supported values: + * - 0 -- Stream information is not present in this message + * - 1 -- Stream information is present in this message + * + * When set to 1, the FLAC bitstream was successfully parsed by the + * client, and other fields in the FLAC format block can be read by the + * decoder to get metadata stream information. + */ + + u16 num_channels; +/* Number of channels for decoding. + * Supported values: 1 to 2 + */ + + u16 min_blk_size; +/* Minimum block size (in samples) used in the stream. It must be less + * than or equal to max_blk_size. + */ + + u16 max_blk_size; +/* Maximum block size (in samples) used in the stream. If the + * minimum block size equals the maximum block size, a fixed block + * size stream is implied. + */ + + u16 md5_sum[8]; +/* MD5 signature array of the unencoded audio data. This allows the + * decoder to determine if an error exists in the audio data, even when + * the error does not result in an invalid bitstream. + */ + + u32 sample_rate; +/* Number of samples per second. + * Supported values: 8000 to 48000 Hz + */ + + u32 min_frame_size; +/* Minimum frame size used in the stream. + * Supported values: + * - > 0 bytes + * - 0 -- The value is unknown + */ + + u32 max_frame_size; +/* Maximum frame size used in the stream. + * Supported values: + * -- > 0 bytes + * -- 0 . The value is unknown + */ + + u16 sample_size; +/* Bits per sample.Supported values: 8, 16 */ + + u16 reserved; +/* Clients must set this field to zero + */ + +} __packed; + +struct asm_alac_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; + +} __packed; + +struct asm_g711_dec_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + u32 sample_rate; +} __packed; + +struct asm_ape_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; + +} __packed; + +struct asm_dsd_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + + u16 num_version; + u16 is_bitwise_big_endian; + u16 dsd_channel_block_size; + u16 num_channels; + u8 channel_mapping[8]; + u32 dsd_data_rate; + +} __packed; + +#define ASM_MEDIA_FMT_AMRNB_FS 0x00010BEB + +/* Enumeration for 4.75 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR475 0 + +/* Enumeration for 5.15 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR515 1 + +/* Enumeration for 5.90 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR59 2 + +/* Enumeration for 6.70 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR67 3 + +/* Enumeration for 7.40 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR74 4 + +/* Enumeration for 7.95 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR795 5 + +/* Enumeration for 10.20 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR102 6 + +/* Enumeration for 12.20 kbps AMR-NB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR122 7 + +/* Enumeration for AMR-NB Discontinuous Transmission mode off. */ +#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF 0 + +/* Enumeration for AMR-NB DTX mode VAD1. */ +#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 1 + +/* Enumeration for AMR-NB DTX mode VAD2. */ +#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD2 2 + +/* Enumeration for AMR-NB DTX mode auto. + */ +#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_AUTO 3 + +struct asm_amrnb_enc_cfg { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u16 enc_mode; +/* AMR-NB encoding rate. + * Supported values: + * Use the ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_* + * macros + */ + + u16 dtx_mode; +/* Specifies whether DTX mode is disabled or enabled. + * Supported values: + * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF + * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 + */ +} __packed; + +#define ASM_MEDIA_FMT_AMRWB_FS 0x00010BEC + +/* Enumeration for 6.6 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR66 0 + +/* Enumeration for 8.85 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR885 1 + +/* Enumeration for 12.65 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1265 2 + +/* Enumeration for 14.25 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1425 3 + +/* Enumeration for 15.85 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1585 4 + +/* Enumeration for 18.25 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1825 5 + +/* Enumeration for 19.85 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1985 6 + +/* Enumeration for 23.05 kbps AMR-WB Encoding mode. */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2305 7 + +/* Enumeration for 23.85 kbps AMR-WB Encoding mode. + */ +#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2385 8 + +struct asm_amrwb_enc_cfg { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u16 enc_mode; +/* AMR-WB encoding rate. + * Suupported values: + * Use the ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_* + * macros + */ + + u16 dtx_mode; +/* Specifies whether DTX mode is disabled or enabled. + * Supported values: + * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF + * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 + */ +} __packed; + +#define ASM_MEDIA_FMT_V13K_FS 0x00010BED + +/* Enumeration for 14.4 kbps V13K Encoding mode. */ +#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 0 + +/* Enumeration for 12.2 kbps V13K Encoding mode. */ +#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220 1 + +/* Enumeration for 11.2 kbps V13K Encoding mode. */ +#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120 2 + +/* Enumeration for 9.0 kbps V13K Encoding mode. */ +#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90 3 + +/* Enumeration for 7.2 kbps V13K eEncoding mode. */ +#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720 4 + +/* Enumeration for 1/8 vocoder rate.*/ +#define ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE 1 + +/* Enumeration for 1/4 vocoder rate. */ +#define ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE 2 + +/* Enumeration for 1/2 vocoder rate. */ +#define ASM_MEDIA_FMT_VOC_HALF_RATE 3 + +/* Enumeration for full vocoder rate. + */ +#define ASM_MEDIA_FMT_VOC_FULL_RATE 4 + +struct asm_v13k_enc_cfg { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + u16 max_rate; +/* Maximum allowed encoder frame rate. + * Supported values: + * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE + * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE + * - #ASM_MEDIA_FMT_VOC_HALF_RATE + * - #ASM_MEDIA_FMT_VOC_FULL_RATE + */ + + u16 min_rate; +/* Minimum allowed encoder frame rate. + * Supported values: + * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE + * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE + * - #ASM_MEDIA_FMT_VOC_HALF_RATE + * - #ASM_MEDIA_FMT_VOC_FULL_RATE + */ + + u16 reduced_rate_cmd; +/* Reduced rate command, used to change + * the average bitrate of the V13K + * vocoder. + * Supported values: + * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 (Default) + * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220 + * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120 + * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90 + * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720 + */ + + u16 rate_mod_cmd; +/* Rate modulation command. Default = 0. + *- If bit 0=1, rate control is enabled. + *- If bit 1=1, the maximum number of consecutive full rate + * frames is limited with numbers supplied in + * bits 2 to 10. + *- If bit 1=0, the minimum number of non-full rate frames + * in between two full rate frames is forced to + * the number supplied in bits 2 to 10. In both cases, if necessary, + * half rate is used to substitute full rate. - Bits 15 to 10 are + * reserved and must all be set to zero. + */ + +} __packed; + +#define ASM_MEDIA_FMT_EVRC_FS 0x00010BEE + +/* EVRC encoder configuration structure used in the + * #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command. + */ +struct asm_evrc_enc_cfg { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + u16 max_rate; +/* Maximum allowed encoder frame rate. + * Supported values: + * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE + * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE + * - #ASM_MEDIA_FMT_VOC_HALF_RATE + * - #ASM_MEDIA_FMT_VOC_FULL_RATE + */ + + u16 min_rate; +/* Minimum allowed encoder frame rate. + * Supported values: + * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE + * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE + * - #ASM_MEDIA_FMT_VOC_HALF_RATE + * - #ASM_MEDIA_FMT_VOC_FULL_RATE + */ + + u16 rate_mod_cmd; +/* Rate modulation command. Default: 0. + * - If bit 0=1, rate control is enabled. + * - If bit 1=1, the maximum number of consecutive full rate frames + * is limited with numbers supplied in bits 2 to 10. + * + * - If bit 1=0, the minimum number of non-full rate frames in + * between two full rate frames is forced to the number supplied in + * bits 2 to 10. In both cases, if necessary, half rate is used to + * substitute full rate. + * + * - Bits 15 to 10 are reserved and must all be set to zero. + */ + + u16 reserved; + /* Reserved. Clients must set this field to zero. */ +} __packed; + +#define ASM_MEDIA_FMT_WMA_V10PRO_V2 0x00010DA7 + +struct asm_wmaprov10_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + + u16 fmtag; +/* WMA format type. + * Supported values: + * - 0x162 -- WMA 9 Pro + * - 0x163 -- WMA 9 Pro Lossless + * - 0x166 -- WMA 10 Pro + * - 0x167 -- WMA 10 Pro Lossless + */ + + u16 num_channels; +/* Number of channels encoded in the input stream. + * Supported values: 1 to 8 + */ + + u32 sample_rate; +/* Number of samples per second (in Hertz). + * Supported values: 11025, 16000, 22050, 32000, 44100, 48000, + * 88200, 96000 + */ + + u32 avg_bytes_per_sec; +/* Bitrate expressed as the average bytes per second. + * Supported values: 2000 to 96000 + */ + + u16 blk_align; +/* Size of the bitstream packet size in bytes. WMA Pro files + * have a payload of one block per bitstream packet. + * Supported values: @le 13376 + */ + + u16 bits_per_sample; +/* Number of bits per sample in the encoded WMA stream. + * Supported values: 16, 24 + */ + + u32 channel_mask; +/* Bit-packed double word (32-bits) that indicates the + * recommended speaker positions for each source channel. + */ + + u16 enc_options; +/* Bit-packed word with values that indicate whether certain + * features of the bitstream are used. + * Supported values: - 0x0001 -- ENCOPT3_PURE_LOSSLESS - 0x0006 -- + * ENCOPT3_FRM_SIZE_MOD - 0x0038 -- ENCOPT3_SUBFRM_DIV - 0x0040 -- + * ENCOPT3_WRITE_FRAMESIZE_IN_HDR - 0x0080 -- + * ENCOPT3_GENERATE_DRC_PARAMS - 0x0100 -- ENCOPT3_RTMBITS + */ + + + u16 usAdvancedEncodeOpt; + /* Advanced encoding option. */ + + u32 advanced_enc_options2; + /* Advanced encoding option 2. */ + +} __packed; + +#define ASM_MEDIA_FMT_WMA_V9_V2 0x00010DA8 +struct asm_wmastdv9_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + u16 fmtag; +/* WMA format tag. + * Supported values: 0x161 (WMA 9 standard) + */ + + u16 num_channels; +/* Number of channels in the stream. + * Supported values: 1, 2 + */ + + u32 sample_rate; +/* Number of samples per second (in Hertz). + * Supported values: 48000 + */ + + u32 avg_bytes_per_sec; + /* Bitrate expressed as the average bytes per second. */ + + u16 blk_align; +/* Block align. All WMA files with a maximum packet size of + * 13376 are supported. + */ + + + u16 bits_per_sample; +/* Number of bits per sample in the output. + * Supported values: 16 + */ + + u32 channel_mask; +/* Channel mask. + * Supported values: + * - 3 -- Stereo (front left/front right) + * - 4 -- Mono (center) + */ + + u16 enc_options; + /* Options used during encoding. */ + + u16 reserved; + +} __packed; + +#define ASM_MEDIA_FMT_WMA_V8 0x00010D91 + +struct asm_wmastdv8_enc_cfg { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + u32 bit_rate; + /* Encoding rate in bits per second. */ + + u32 sample_rate; +/* Number of samples per second. + * + * Supported values: + * - 0 -- Native mode + * - Other Supported values are 22050, 32000, 44100, and 48000. + * + * Native mode indicates that encoding must be performed with the + * sampling rate at the input. + * The sampling rate must not change during encoding. + */ + + u16 channel_cfg; +/* Number of channels to encode. + * Supported values: + * - 0 -- Native mode + * - 1 -- Mono + * - 2 -- Stereo + * - Other values are not supported. + * + * Native mode indicates that encoding must be performed with the + * number of channels at the input. + * The number of channels must not change during encoding. + */ + + u16 reserved; + /* Reserved. Clients must set this field to zero.*/ + } __packed; + +#define ASM_MEDIA_FMT_AMR_WB_PLUS_V2 0x00010DA9 + +struct asm_amrwbplus_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + u32 amr_frame_fmt; +/* AMR frame format. + * Supported values: + * - 6 -- Transport Interface Format (TIF) + * - Any other value -- File storage format (FSF) + * + * TIF stream contains 2-byte header for each frame within the + * superframe. FSF stream contains one 2-byte header per superframe. + */ + +} __packed; + +#define ASM_MEDIA_FMT_AC3 0x00010DEE +#define ASM_MEDIA_FMT_EAC3 0x00010DEF +#define ASM_MEDIA_FMT_DTS 0x00010D88 +#define ASM_MEDIA_FMT_MP2 0x00010DE9 +#define ASM_MEDIA_FMT_FLAC 0x00010C16 +#define ASM_MEDIA_FMT_ALAC 0x00012F31 +#define ASM_MEDIA_FMT_VORBIS 0x00010C15 +#define ASM_MEDIA_FMT_APE 0x00012F32 +#define ASM_MEDIA_FMT_DSD 0x00012F3E +#define ASM_MEDIA_FMT_TRUEHD 0x00013215 +/* 0x0 is used for fomat ID since ADSP dynamically determines the + * format encapsulated in the IEC61937 (compressed) or IEC60958 + * (pcm) packets. + */ +#define ASM_MEDIA_FMT_IEC 0x00000000 + +/* Media format ID for adaptive transform acoustic coding. This + * ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED command + * only. + */ + +#define ASM_MEDIA_FMT_ATRAC 0x00010D89 + +/* Media format ID for metadata-enhanced audio transmission. + * This ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED + * command only. + */ + +#define ASM_MEDIA_FMT_MAT 0x00010D8A + +/* adsp_media_fmt.h */ + +#define ASM_DATA_CMD_WRITE_V2 0x00010DAB + +struct asm_data_cmd_write_v2 { + struct apr_hdr hdr; + u32 buf_addr_lsw; +/* The 64 bit address msw-lsw should be a valid, mapped address. + * 64 bit address should be a multiple of 32 bytes + */ + + u32 buf_addr_msw; +/* The 64 bit address msw-lsw should be a valid, mapped address. + * 64 bit address should be a multiple of 32 bytes. + * -Address of the buffer containing the data to be decoded. + * The buffer should be aligned to a 32 byte boundary. + * -In the case of 32 bit Shared memory address, msw field must + * -be set to zero. + * -In the case of 36 bit shared memory address, bit 31 to bit 4 + * -of msw must be set to zero. + */ + u32 mem_map_handle; +/* memory map handle returned by DSP through + * ASM_CMD_SHARED_MEM_MAP_REGIONS command + */ + u32 buf_size; +/* Number of valid bytes available in the buffer for decoding. The + * first byte starts at buf_addr. + */ + + u32 seq_id; + /* Optional buffer sequence ID. */ + + u32 timestamp_lsw; +/* Lower 32 bits of the 64-bit session time in microseconds of the + * first buffer sample. + */ + + u32 timestamp_msw; +/* Upper 32 bits of the 64-bit session time in microseconds of the + * first buffer sample. + */ + + u32 flags; +/* Bitfield of flags. + * Supported values for bit 31: + * - 1 -- Valid timestamp. + * - 0 -- Invalid timestamp. + * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG as the bitmask and + * #ASM_SHIFTIMESTAMP_VALID_FLAG as the shift value to set this bit. + * Supported values for bit 30: + * - 1 -- Last buffer. + * - 0 -- Not the last buffer. + * + * Supported values for bit 29: + * - 1 -- Continue the timestamp from the previous buffer. + * - 0 -- Timestamp of the current buffer is not related + * to the timestamp of the previous buffer. + * - Use #ASM_BIT_MASKS_CONTINUE_FLAG and #ASM_SHIFTS_CONTINUE_FLAG + * to set this bit. + * + * Supported values for bit 4: + * - 1 -- End of the frame. + * - 0 -- Not the end of frame, or this information is not known. + * - Use #ASM_BIT_MASK_EOF_FLAG as the bitmask and #ASM_SHIFT_EOF_FLAG + * as the shift value to set this bit. + * + * All other bits are reserved and must be set to 0. + * + * If bit 31=0 and bit 29=1: The timestamp of the first sample in + * this buffer continues from the timestamp of the last sample in + * the previous buffer. If there is no previous buffer (i.e., this + * is the first buffer sent after opening the stream or after a + * flush operation), or if the previous buffer does not have a valid + * timestamp, the samples in the current buffer also do not have a + * valid timestamp. They are played out as soon as possible. + * + * + * If bit 31=0 and bit 29=0: No timestamp is associated with the + * first sample in this buffer. The samples are played out as soon + * as possible. + * + * + * If bit 31=1 and bit 29 is ignored: The timestamp specified in + * this payload is honored. + * + * + * If bit 30=0: Not the last buffer in the stream. This is useful + * in removing trailing samples. + * + * + * For bit 4: The client can set this flag for every buffer sent in + * which the last byte is the end of a frame. If this flag is set, + * the buffer can contain data from multiple frames, but it should + * always end at a frame boundary. Restrictions allow the aDSP to + * detect an end of frame without requiring additional processing. + */ + +} __packed; + +#define ASM_DATA_CMD_READ_V2 0x00010DAC + +struct asm_data_cmd_read_v2 { + struct apr_hdr hdr; + u32 buf_addr_lsw; +/* the 64 bit address msw-lsw should be a valid mapped address + * and should be a multiple of 32 bytes + */ + + + u32 buf_addr_msw; +/* the 64 bit address msw-lsw should be a valid mapped address + * and should be a multiple of 32 bytes. +* - Address of the buffer where the DSP puts the encoded data, +* potentially, at an offset specified by the uOffset field in +* ASM_DATA_EVENT_READ_DONE structure. The buffer should be aligned +* to a 32 byte boundary. +*- In the case of 32 bit Shared memory address, msw field must +*- be set to zero. +*- In the case of 36 bit shared memory address, bit 31 to bit +*- 4 of msw must be set to zero. +*/ + u32 mem_map_handle; +/* memory map handle returned by DSP through + * ASM_CMD_SHARED_MEM_MAP_REGIONS command. + */ + + u32 buf_size; +/* Number of bytes available for the aDSP to write. The aDSP + * starts writing from buf_addr. + */ + + u32 seq_id; + /* Optional buffer sequence ID. + */ +} __packed; + +#define ASM_DATA_CMD_EOS 0x00010BDB +#define ASM_DATA_EVENT_RENDERED_EOS 0x00010C1C +#define ASM_DATA_EVENT_EOS 0x00010BDD + +#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99 +struct asm_data_event_write_done_v2 { + u32 buf_addr_lsw; + /* lsw of the 64 bit address */ + u32 buf_addr_msw; + /* msw of the 64 bit address. address given by the client in + * ASM_DATA_CMD_WRITE_V2 command. + */ + u32 mem_map_handle; + /* memory map handle in the ASM_DATA_CMD_WRITE_V2 */ + + u32 status; +/* Status message (error code) that indicates whether the + * referenced buffer has been successfully consumed. + * Supported values: Refer to @xhyperref{Q3,[Q3]} + */ +} __packed; + +#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A + +/* Definition of the frame metadata flag bitmask.*/ +#define ASM_BIT_MASK_FRAME_METADATA_FLAG (0x40000000UL) + +/* Definition of the frame metadata flag shift value. */ +#define ASM_SHIFT_FRAME_METADATA_FLAG 30 + +struct asm_data_event_read_done_v2 { + u32 status; +/* Status message (error code). + * Supported values: Refer to @xhyperref{Q3,[Q3]} + */ + +u32 buf_addr_lsw; +/* 64 bit address msw-lsw is a valid, mapped address. 64 bit + * address is a multiple of 32 bytes. + */ + +u32 buf_addr_msw; +/* 64 bit address msw-lsw is a valid, mapped address. 64 bit +* address is a multiple of 32 bytes. +* +* -Same address provided by the client in ASM_DATA_CMD_READ_V2 +* -In the case of 32 bit Shared memory address, msw field is set to +* zero. +* -In the case of 36 bit shared memory address, bit 31 to bit 4 +* -of msw is set to zero. +*/ + +u32 mem_map_handle; +/* memory map handle in the ASM_DATA_CMD_READ_V2 */ + +u32 enc_framesotal_size; +/* Total size of the encoded frames in bytes. + * Supported values: >0 + */ + +u32 offset; +/* Offset (from buf_addr) to the first byte of the first encoded + * frame. All encoded frames are consecutive, starting from this + * offset. + * Supported values: > 0 + */ + +u32 timestamp_lsw; +/* Lower 32 bits of the 64-bit session time in microseconds of + * the first sample in the buffer. If Bit 5 of mode_flags flag of + * ASM_STREAM_CMD_OPEN_READ_V2 is 1 then the 64 bit timestamp is + * absolute capture time otherwise it is relative session time. The + * absolute timestamp doesnt reset unless the system is reset. + */ + + +u32 timestamp_msw; +/* Upper 32 bits of the 64-bit session time in microseconds of + * the first sample in the buffer. + */ + + +u32 flags; +/* Bitfield of flags. Bit 30 indicates whether frame metadata is + * present. If frame metadata is present, num_frames consecutive + * instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start + * at the buffer address. + * Supported values for bit 31: + * - 1 -- Timestamp is valid. + * - 0 -- Timestamp is invalid. + * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and + * #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit. + * + * Supported values for bit 30: + * - 1 -- Frame metadata is present. + * - 0 -- Frame metadata is absent. + * - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and + * #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit. + * + * All other bits are reserved; the aDSP sets them to 0. + */ + +u32 num_frames; +/* Number of encoded frames in the buffer. */ + +u32 seq_id; +/* Optional buffer sequence ID. */ +} __packed; + +struct asm_data_read_buf_metadata_v2 { + u32 offset; +/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to + * the frame associated with this metadata. + * Supported values: > 0 + */ + +u32 frm_size; +/* Size of the encoded frame in bytes. + * Supported values: > 0 + */ + +u32 num_encoded_pcm_samples; +/* Number of encoded PCM samples (per channel) in the frame + * associated with this metadata. + * Supported values: > 0 + */ + +u32 timestamp_lsw; +/* Lower 32 bits of the 64-bit session time in microseconds of the + * first sample for this frame. + * If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1 + * then the 64 bit timestamp is absolute capture time otherwise it + * is relative session time. The absolute timestamp doesnt reset + * unless the system is reset. + */ + + +u32 timestamp_msw; +/* Lower 32 bits of the 64-bit session time in microseconds of the + * first sample for this frame. + */ + +u32 flags; +/* Frame flags. + * Supported values for bit 31: + * - 1 -- Time stamp is valid + * - 0 -- Time stamp is not valid + * - All other bits are reserved; the aDSP sets them to 0. +*/ +} __packed; + +/* Notifies the client of a change in the data sampling rate or + * Channel mode. This event is raised by the decoder service. The + * event is enabled through the mode flags of + * #ASM_STREAM_CMD_OPEN_WRITE_V2 or + * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change + * in the output sampling frequency or the number/positioning of + * output channels, or if it is the first frame decoded.The new + * sampling frequency or the new channel configuration is + * communicated back to the client asynchronously. + */ + +#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65 + +/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event. + * This event is raised when the following conditions are both true: + * - The event is enabled through the mode_flags of + * #ASM_STREAM_CMD_OPEN_WRITE_V2 or + * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change + * in either the output sampling frequency or the number/positioning + * of output channels, or if it is the first frame decoded. + * This event is not raised (even if enabled) if the decoder is + * MIDI, because + */ + + +struct asm_data_event_sr_cm_change_notify { + u32 sample_rate; +/* New sampling rate (in Hertz) after detecting a change in the + * bitstream. + * Supported values: 2000 to 48000 + */ + + u16 num_channels; +/* New number of channels after detecting a change in the + * bitstream. + * Supported values: 1 to 8 + */ + + + u16 reserved; + /* Reserved for future use. This field must be set to 0.*/ + + u8 channel_mapping[8]; + +} __packed; + +/* Notifies the client of a data sampling rate or channel mode + * change. This event is raised by the encoder service. + * This event is raised when : + * - Native mode encoding was requested in the encoder + * configuration (i.e., the channel number was 0), the sample rate + * was 0, or both were 0. + * + * - The input data frame at the encoder is the first one, or the + * sampling rate/channel mode is different from the previous input + * data frame. + * + */ +#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE + +struct asm_data_event_enc_sr_cm_change_notify { + u32 sample_rate; +/* New sampling rate (in Hertz) after detecting a change in the + * input data. + * Supported values: 2000 to 48000 + */ + + + u16 num_channels; +/* New number of channels after detecting a change in the input + * data. Supported values: 1 to 8 + */ + + + u16 bits_per_sample; +/* New bits per sample after detecting a change in the input + * data. + * Supported values: 16, 24 + */ + + + u8 channel_mapping[8]; + +} __packed; +#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87 + + +/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command, + * which is used to indicate the IEC 60958 frame rate of a given + * packetized audio stream. + */ + +struct asm_data_cmd_iec_60958_frame_rate { + u32 frame_rate; +/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream. + * Supported values: Any valid frame rate + */ +} __packed; + +/* adsp_asm_data_commands.h*/ +/* Definition of the stream ID bitmask.*/ +#define ASM_BIT_MASK_STREAM_ID (0x000000FFUL) + +/* Definition of the stream ID shift value.*/ +#define ASM_SHIFT_STREAM_ID 0 + +/* Definition of the session ID bitmask.*/ +#define ASM_BIT_MASK_SESSION_ID (0x0000FF00UL) + +/* Definition of the session ID shift value.*/ +#define ASM_SHIFT_SESSION_ID 8 + +/* Definition of the service ID bitmask.*/ +#define ASM_BIT_MASK_SERVICE_ID (0x00FF0000UL) + +/* Definition of the service ID shift value.*/ +#define ASM_SHIFT_SERVICE_ID 16 + +/* Definition of the domain ID bitmask.*/ +#define ASM_BIT_MASK_DOMAIN_ID (0xFF000000UL) + +/* Definition of the domain ID shift value.*/ +#define ASM_SHIFT_DOMAIN_ID 24 + +#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92 +#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93 +#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94 + +/* adsp_asm_service_commands.h */ + +#define ASM_MAX_SESSION_ID (15) + +/* Maximum number of sessions.*/ +#define ASM_MAX_NUM_SESSIONS ASM_MAX_SESSION_ID + +/* Maximum number of streams per session.*/ +#define ASM_MAX_STREAMS_PER_SESSION (8) +#define ASM_SESSION_CMD_RUN_V2 0x00010DAA +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE 0 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2 +#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY 3 + +#define ASM_BIT_MASK_RUN_STARTIME (0x00000003UL) + +/* Bit shift value used to specify the start time for the + * ASM_SESSION_CMD_RUN_V2 command. + */ +#define ASM_SHIFT_RUN_STARTIME 0 +struct asm_session_cmd_run_v2 { + struct apr_hdr hdr; + u32 flags; +/* Specifies whether to run immediately or at a specific + * rendering time or with a specified delay. Run with delay is + * useful for delaying in case of ASM loopback opened through + * ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME + * and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag. + * + * + *Bits 0 and 1 can take one of four possible values: + * + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME + *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY + * + *All other bits are reserved; clients must set them to zero. + */ + + u32 time_lsw; +/* Lower 32 bits of the time in microseconds used to align the + * session origin time. When bits 0-1 of flags is + * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of + * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, + * maximum value of the 64 bit delay is 150 ms. + */ + + u32 time_msw; +/* Upper 32 bits of the time in microseconds used to align the + * session origin time. When bits 0-1 of flags is + * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of + * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, + * maximum value of the 64 bit delay is 150 ms. + */ + +} __packed; + +#define ASM_SESSION_CMD_PAUSE 0x00010BD3 +#define ASM_SESSION_CMD_SUSPEND 0x00010DEC +#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D +#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5 + +struct asm_session_cmd_rgstr_rx_underflow { + struct apr_hdr hdr; + u16 enable_flag; +/* Specifies whether a client is to receive events when an Rx + * session underflows. + * Supported values: + * - 0 -- Do not send underflow events + * - 1 -- Send underflow events + */ + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6 + +struct asm_session_cmd_regx_overflow { + struct apr_hdr hdr; + u16 enable_flag; +/* Specifies whether a client is to receive events when a Tx +* session overflows. + * Supported values: + * - 0 -- Do not send overflow events + * - 1 -- Send overflow events + */ + + u16 reserved; + /* Reserved. This field must be set to zero.*/ +} __packed; + +#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17 +#define ASM_SESSION_EVENTX_OVERFLOW 0x00010C18 +#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E + +struct asm_session_cmdrsp_get_sessiontime_v3 { + u32 status; + /* Status message (error code). + * Supported values: Refer to @xhyperref{Q3,[Q3]} + */ + + u32 sessiontime_lsw; + /* Lower 32 bits of the current session time in microseconds.*/ + + u32 sessiontime_msw; + /* Upper 32 bits of the current session time in microseconds.*/ + + u32 absolutetime_lsw; +/* Lower 32 bits in micro seconds of the absolute time at which + * the * sample corresponding to the above session time gets + * rendered * to hardware. This absolute time may be slightly in the + * future or past. + */ + + + u32 absolutetime_msw; +/* Upper 32 bits in micro seconds of the absolute time at which + * the * sample corresponding to the above session time gets + * rendered to * hardware. This absolute time may be slightly in the + * future or past. + */ + +} __packed; + +#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2 0x00010D9F + +struct asm_session_cmd_adjust_session_clock_v2 { + struct apr_hdr hdr; +u32 adjustime_lsw; +/* Lower 32 bits of the signed 64-bit quantity that specifies the + * adjustment time in microseconds to the session clock. + * + * Positive values indicate advancement of the session clock. + * Negative values indicate delay of the session clock. + */ + + + u32 adjustime_msw; +/* Upper 32 bits of the signed 64-bit quantity that specifies + * the adjustment time in microseconds to the session clock. + * Positive values indicate advancement of the session clock. + * Negative values indicate delay of the session clock. + */ + +} __packed; + +#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 0x00010DA0 + +struct asm_session_cmdrsp_adjust_session_clock_v2 { + u32 status; +/* Status message (error code). + * Supported values: Refer to @xhyperref{Q3,[Q3]} + * An error means the session clock is not adjusted. In this case, + * the next two fields are irrelevant. + */ + + + u32 actual_adjustime_lsw; +/* Lower 32 bits of the signed 64-bit quantity that specifies + * the actual adjustment in microseconds performed by the aDSP. + * A positive value indicates advancement of the session clock. A + * negative value indicates delay of the session clock. + */ + + + u32 actual_adjustime_msw; +/* Upper 32 bits of the signed 64-bit quantity that specifies + * the actual adjustment in microseconds performed by the aDSP. + * A positive value indicates advancement of the session clock. A + * negative value indicates delay of the session clock. + */ + + + u32 cmd_latency_lsw; +/* Lower 32 bits of the unsigned 64-bit quantity that specifies + * the amount of time in microseconds taken to perform the session + * clock adjustment. + */ + + + u32 cmd_latency_msw; +/* Upper 32 bits of the unsigned 64-bit quantity that specifies + * the amount of time in microseconds taken to perform the session + * clock adjustment. + */ + +} __packed; + +#define ASM_SESSION_CMD_GET_PATH_DELAY_V2 0x00010DAF +#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0 + +struct asm_session_cmdrsp_get_path_delay_v2 { + u32 status; +/* Status message (error code). Whether this get delay operation + * is successful or not. Delay value is valid only if status is + * success. + * Supported values: Refer to @xhyperref{Q5,[Q5]} + */ + + u32 audio_delay_lsw; + /* Upper 32 bits of the aDSP delay in microseconds. */ + + u32 audio_delay_msw; + /* Lower 32 bits of the aDSP delay in microseconds. */ + +} __packed; + +/* adsp_asm_session_command.h*/ +#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3 + +#define ASM_LOW_LATENCY_STREAM_SESSION 0x10000000 + +#define ASM_ULTRA_LOW_LATENCY_STREAM_SESSION 0x20000000 + +#define ASM_ULL_POST_PROCESSING_STREAM_SESSION 0x40000000 + +#define ASM_LEGACY_STREAM_SESSION 0 + + +struct asm_stream_cmd_open_write_v3 { + struct apr_hdr hdr; + uint32_t mode_flags; +/* Mode flags that configure the stream to notify the client + * whenever it detects an SR/CM change at the input to its POPP. + * Supported values for bits 0 to 1: + * - Reserved; clients must set them to zero. + * Supported values for bit 2: + * - 0 -- SR/CM change notification event is disabled. + * - 1 -- SR/CM change notification event is enabled. + * - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and + * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit. + * + * Supported values for bit 31: + * - 0 -- Stream to be opened in on-Gapless mode. + * - 1 -- Stream to be opened in Gapless mode. In Gapless mode, + * successive streams must be opened with same session ID but + * different stream IDs. + * + * - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and + * #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit. + * + * + * @note1hang MIDI and DTMF streams cannot be opened in Gapless mode. + */ + + uint16_t sink_endpointype; +/*< Sink point type. + * Supported values: + * - 0 -- Device matrix + * - Other values are reserved. + * + * The device matrix is the gateway to the hardware ports. + */ + + uint16_t bits_per_sample; +/*< Number of bits per sample processed by ASM modules. + * Supported values: 16 and 24 bits per sample + */ + + uint32_t postprocopo_id; +/*< Specifies the topology (order of processing) of + * postprocessing algorithms. <i>None</i> means no postprocessing. + * Supported values: + * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT + * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL + * - #ASM_STREAM_POSTPROCOPO_ID_NONE + * + * This field can also be enabled through SetParams flags. + */ + + uint32_t dec_fmt_id; +/*< Configuration ID of the decoder media format. + * + * Supported values: + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_ADPCM + * - #ASM_MEDIA_FMT_MP3 + * - #ASM_MEDIA_FMT_AAC_V2 + * - #ASM_MEDIA_FMT_DOLBY_AAC + * - #ASM_MEDIA_FMT_AMRNB_FS + * - #ASM_MEDIA_FMT_AMRWB_FS + * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2 + * - #ASM_MEDIA_FMT_V13K_FS + * - #ASM_MEDIA_FMT_EVRC_FS + * - #ASM_MEDIA_FMT_EVRCB_FS + * - #ASM_MEDIA_FMT_EVRCWB_FS + * - #ASM_MEDIA_FMT_SBC + * - #ASM_MEDIA_FMT_WMA_V10PRO_V2 + * - #ASM_MEDIA_FMT_WMA_V9_V2 + * - #ASM_MEDIA_FMT_AC3 + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_G711_ALAW_FS + * - #ASM_MEDIA_FMT_G711_MLAW_FS + * - #ASM_MEDIA_FMT_G729A_FS + * - #ASM_MEDIA_FMT_FR_FS + * - #ASM_MEDIA_FMT_VORBIS + * - #ASM_MEDIA_FMT_FLAC + * - #ASM_MEDIA_FMT_ALAC + * - #ASM_MEDIA_FMT_APE + * - #ASM_MEDIA_FMT_EXAMPLE + */ +} __packed; + +#define ASM_STREAM_CMD_OPEN_PULL_MODE_WRITE 0x00010DD9 + +/* Bitmask for the stream_perf_mode subfield. */ +#define ASM_BIT_MASK_STREAM_PERF_FLAG_PULL_MODE_WRITE 0xE0000000UL + +/* Bitmask for the stream_perf_mode subfield. */ +#define ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE 29 + +#define ASM_STREAM_CMD_OPEN_PUSH_MODE_READ 0x00010DDA + +#define ASM_BIT_MASK_STREAM_PERF_FLAG_PUSH_MODE_READ 0xE0000000UL + +#define ASM_SHIFT_STREAM_PERF_FLAG_PUSH_MODE_READ 29 + +#define ASM_DATA_EVENT_WATERMARK 0x00010DDB + +struct asm_shared_position_buffer { + volatile uint32_t frame_counter; +/* Counter used to handle interprocessor synchronization issues. + * When frame_counter is 0: read_index, wall_clock_us_lsw, and + * wall_clock_us_msw are invalid. + * Supported values: >= 0. + */ + + volatile uint32_t index; +/* Index in bytes from where the aDSP is reading/writing. + * Supported values: 0 to circular buffer size - 1 + */ + + volatile uint32_t wall_clock_us_lsw; +/* Lower 32 bits of the 64-bit wall clock time in microseconds when the + * read index was updated. + * Supported values: >= 0 + */ + + volatile uint32_t wall_clock_us_msw; +/* Upper 32 bits of the 64 bit wall clock time in microseconds when the + * read index was updated + * Supported values: >= 0 + */ +} __packed; + +struct asm_shared_watermark_level { + uint32_t watermark_level_bytes; +} __packed; + +struct asm_stream_cmd_open_shared_io { + struct apr_hdr hdr; + uint32_t mode_flags; + uint16_t endpoint_type; + uint16_t topo_bits_per_sample; + uint32_t topo_id; + uint32_t fmt_id; + uint32_t shared_pos_buf_phy_addr_lsw; + uint32_t shared_pos_buf_phy_addr_msw; + uint16_t shared_pos_buf_mem_pool_id; + uint16_t shared_pos_buf_num_regions; + uint32_t shared_pos_buf_property_flag; + uint32_t shared_circ_buf_start_phy_addr_lsw; + uint32_t shared_circ_buf_start_phy_addr_msw; + uint32_t shared_circ_buf_size; + uint16_t shared_circ_buf_mem_pool_id; + uint16_t shared_circ_buf_num_regions; + uint32_t shared_circ_buf_property_flag; + uint32_t num_watermark_levels; + struct asm_multi_channel_pcm_fmt_blk_v3 fmt; + struct avs_shared_map_region_payload map_region_pos_buf; + struct avs_shared_map_region_payload map_region_circ_buf; + struct asm_shared_watermark_level watermark[0]; +} __packed; + +#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4 + +/* Definition of the timestamp type flag bitmask */ +#define ASM_BIT_MASKIMESTAMPYPE_FLAG (0x00000020UL) + +/* Definition of the timestamp type flag shift value. */ +#define ASM_SHIFTIMESTAMPYPE_FLAG 5 + +/* Relative timestamp is identified by this value.*/ +#define ASM_RELATIVEIMESTAMP 0 + +/* Absolute timestamp is identified by this value.*/ +#define ASM_ABSOLUTEIMESTAMP 1 + +/* Bit value for Low Latency Tx stream subfield */ +#define ASM_LOW_LATENCY_TX_STREAM_SESSION 1 + +/* Bit shift for the stream_perf_mode subfield. */ +#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29 + +struct asm_stream_cmd_open_read_v3 { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode flags that indicate whether meta information per encoded + * frame is to be provided. + * Supported values for bit 4: + * + * - 0 -- Return data buffer contains all encoded frames only; it + * does not contain frame metadata. + * + * - 1 -- Return data buffer contains an array of metadata and + * encoded frames. + * + * - Use #ASM_BIT_MASK_META_INFO_FLAG as the bitmask and + * #ASM_SHIFT_META_INFO_FLAG as the shift value for this bit. + * + * + * Supported values for bit 5: + * + * - ASM_RELATIVEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will have + * - relative time-stamp. + * - ASM_ABSOLUTEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will + * - have absolute time-stamp. + * + * - Use #ASM_BIT_MASKIMESTAMPYPE_FLAG as the bitmask and + * #ASM_SHIFTIMESTAMPYPE_FLAG as the shift value for this bit. + * + * All other bits are reserved; clients must set them to zero. + */ + + u32 src_endpointype; +/* Specifies the endpoint providing the input samples. + * Supported values: + * - 0 -- Device matrix + * - All other values are reserved; clients must set them to zero. + * Otherwise, an error is returned. + * The device matrix is the gateway from the tunneled Tx ports. + */ + + u32 preprocopo_id; +/* Specifies the topology (order of processing) of preprocessing + * algorithms. <i>None</i> means no preprocessing. + * Supported values: + * - #ASM_STREAM_PREPROCOPO_ID_DEFAULT + * - #ASM_STREAM_PREPROCOPO_ID_NONE + * + * This field can also be enabled through SetParams flags. + */ + + u32 enc_cfg_id; +/* Media configuration ID for encoded output. + * Supported values: + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_AAC_V2 + * - #ASM_MEDIA_FMT_AMRNB_FS + * - #ASM_MEDIA_FMT_AMRWB_FS + * - #ASM_MEDIA_FMT_V13K_FS + * - #ASM_MEDIA_FMT_EVRC_FS + * - #ASM_MEDIA_FMT_EVRCB_FS + * - #ASM_MEDIA_FMT_EVRCWB_FS + * - #ASM_MEDIA_FMT_SBC + * - #ASM_MEDIA_FMT_G711_ALAW_FS + * - #ASM_MEDIA_FMT_G711_MLAW_FS + * - #ASM_MEDIA_FMT_G729A_FS + * - #ASM_MEDIA_FMT_EXAMPLE + * - #ASM_MEDIA_FMT_WMA_V8 + */ + + u16 bits_per_sample; +/* Number of bits per sample processed by ASM modules. + * Supported values: 16 and 24 bits per sample + */ + + u16 reserved; +/* Reserved for future use. This field must be set to zero.*/ +} __packed; + +#define ASM_POPP_OUTPUT_SR_NATIVE_RATE 0 + +/* Enumeration for the maximum sampling rate at the POPP output.*/ +#define ASM_POPP_OUTPUT_SR_MAX_RATE 48000 + +#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D +#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D + +struct asm_stream_cmd_open_readwrite_v2 { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode flags. + * Supported values for bit 2: + * - 0 -- SR/CM change notification event is disabled. + * - 1 -- SR/CM change notification event is enabled. Use + * #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and + * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or + * getting this flag. + * + * Supported values for bit 4: + * - 0 -- Return read data buffer contains all encoded frames only; it + * does not contain frame metadata. + * - 1 -- Return read data buffer contains an array of metadata and + * encoded frames. + * + * All other bits are reserved; clients must set them to zero. + */ + + u32 postprocopo_id; +/* Specifies the topology (order of processing) of postprocessing + * algorithms. <i>None</i> means no postprocessing. + * + * Supported values: + * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT + * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL + * - #ASM_STREAM_POSTPROCOPO_ID_NONE + */ + + u32 dec_fmt_id; +/* Specifies the media type of the input data. PCM indicates that + * no decoding must be performed, e.g., this is an NT encoder + * session. + * Supported values: + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_ADPCM + * - #ASM_MEDIA_FMT_MP3 + * - #ASM_MEDIA_FMT_AAC_V2 + * - #ASM_MEDIA_FMT_DOLBY_AAC + * - #ASM_MEDIA_FMT_AMRNB_FS + * - #ASM_MEDIA_FMT_AMRWB_FS + * - #ASM_MEDIA_FMT_V13K_FS + * - #ASM_MEDIA_FMT_EVRC_FS + * - #ASM_MEDIA_FMT_EVRCB_FS + * - #ASM_MEDIA_FMT_EVRCWB_FS + * - #ASM_MEDIA_FMT_SBC + * - #ASM_MEDIA_FMT_WMA_V10PRO_V2 + * - #ASM_MEDIA_FMT_WMA_V9_V2 + * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2 + * - #ASM_MEDIA_FMT_AC3 + * - #ASM_MEDIA_FMT_G711_ALAW_FS + * - #ASM_MEDIA_FMT_G711_MLAW_FS + * - #ASM_MEDIA_FMT_G729A_FS + * - #ASM_MEDIA_FMT_EXAMPLE + */ + + u32 enc_cfg_id; +/* Specifies the media type for the output of the stream. PCM + * indicates that no encoding must be performed, e.g., this is an NT + * decoder session. + * Supported values: + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_AAC_V2 + * - #ASM_MEDIA_FMT_AMRNB_FS + * - #ASM_MEDIA_FMT_AMRWB_FS + * - #ASM_MEDIA_FMT_V13K_FS + * - #ASM_MEDIA_FMT_EVRC_FS + * - #ASM_MEDIA_FMT_EVRCB_FS + * - #ASM_MEDIA_FMT_EVRCWB_FS + * - #ASM_MEDIA_FMT_SBC + * - #ASM_MEDIA_FMT_G711_ALAW_FS + * - #ASM_MEDIA_FMT_G711_MLAW_FS + * - #ASM_MEDIA_FMT_G729A_FS + * - #ASM_MEDIA_FMT_EXAMPLE + * - #ASM_MEDIA_FMT_WMA_V8 + */ + + u16 bits_per_sample; +/* Number of bits per sample processed by ASM modules. + * Supported values: 16 and 24 bits per sample + */ + + u16 reserved; +/* Reserved for future use. This field must be set to zero.*/ + +} __packed; + +#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E +struct asm_stream_cmd_open_loopback_v2 { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode flags. + * Bit 0-31: reserved; client should set these bits to 0 + */ + u16 src_endpointype; + /* Endpoint type. 0 = Tx Matrix */ + u16 sink_endpointype; + /* Endpoint type. 0 = Rx Matrix */ + u32 postprocopo_id; +/* Postprocessor topology ID. Specifies the topology of + * postprocessing algorithms. + */ + + u16 bits_per_sample; +/* The number of bits per sample processed by ASM modules + * Supported values: 16 and 24 bits per sample + */ + u16 reserved; +/* Reserved for future use. This field must be set to zero. */ +} __packed; + + +#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA + +/* Bitmask for the stream's Performance mode. */ +#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \ + (0x70000000UL) + +/* Bit shift for the stream's Performance mode. */ +#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28 + +/* Bitmask for the decoder converter enable flag. */ +#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL) + +/* Shift value for the decoder converter enable flag. */ +#define ASM_SHIFT_DECODER_CONVERTER_FLAG 3 + +/* Converter mode is None (Default). */ +#define ASM_CONVERTER_MODE_NONE 0 + +/* Converter mode is DDP-to-DD. */ +#define ASM_DDP_DD_CONVERTER_MODE 1 + +/* Identifies a special converter mode where source and sink formats + * are the same but postprocessing must applied. Therefore, Decode + * @rarrow Re-encode is necessary. + */ +#define ASM_POST_PROCESS_CONVERTER_MODE 2 + + +struct asm_stream_cmd_open_transcode_loopback_t { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode Flags specifies the performance mode in which this stream + * is to be opened. + * Supported values{for bits 30 to 28}(stream_perf_mode flag) + * + * #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward + * compatibility to the original behavior + * of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK + * + * #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using + * shortened buffers in low latency POPP + * - Recommendation: Do not enable high latency algorithms. They might + * negate the benefits of opening a low latency stream, and they + * might also suffer quality degradation from unexpected jitter. + * - This Low Latency mode is supported only for PCM In and PCM Out + * loopbacks. An error is returned if Low Latency mode is opened for + * other transcode loopback modes. + * - To configure this subfield, use + * ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and + * ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK. + * + * Supported values{for bits 6 to 3} (decoder-converter compatibility) + * #ASM_CONVERTER_MODE_NONE (0x0) -- Default + * #ASM_DDP_DD_CONVERTER_MODE (0x1) + * #ASM_POST_PROCESS_CONVERTER_MODE (0x2) + * 0x3-0xF -- Reserved for future use + * - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and + * ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit + * All other bits are reserved; clients must set them to 0. + */ + + u32 src_format_id; +/* Specifies the media format of the input audio stream. + * + * Supported values + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 + * - #ASM_MEDIA_FMT_DTS + * - #ASM_MEDIA_FMT_EAC3_DEC + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_AC3_DEC + * - #ASM_MEDIA_FMT_AC3 + */ + u32 sink_format_id; +/* Specifies the media format of the output stream. + * + * Supported values + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 + * - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode) + */ + + u32 audproc_topo_id; +/* Postprocessing topology ID, which specifies the topology (order of + * processing) of postprocessing algorithms. + * + * Supported values + * - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT + * - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER + * - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL + * - #ASM_STREAM_POSTPROC_TOPO_ID_NONE + * Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES. + * This field is ignored for the Converter mode, in which no + * postprocessing is performed. + */ + + u16 src_endpoint_type; +/* Specifies the source endpoint that provides the input samples. + * + * Supported values + * - 0 -- Tx device matrix or stream router (gateway to the hardware + * ports) + * - All other values are reserved + * Clients must set this field to 0. Otherwise, an error is returned. + */ + + u16 sink_endpoint_type; +/* Specifies the sink endpoint type. + * + * Supported values + * - 0 -- Rx device matrix or stream router (gateway to the hardware + * ports) + * - All other values are reserved + * Clients must set this field to 0. Otherwise, an error is returned. + */ + + u16 bits_per_sample; +/* Number of bits per sample processed by the ASM modules. + * Supported values 16, 24 + */ + + u16 reserved; +/* This field must be set to 0. + */ +} __packed; + + +#define ASM_STREAM_CMD_CLOSE 0x00010BCD +#define ASM_STREAM_CMD_FLUSH 0x00010BCE + + +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09 +#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1 +#define ASM_STREAM_CMD_SET_PP_PARAMS_V3 0x0001320D + +/* + * Structure for the ASM Stream Set PP Params command. Parameter data must be + * pre-packed with the correct header for either V2 or V3 when sent in-band. + * Use q6core_pack_pp_params to pack the header and data correctly depending on + * Instance ID support. + */ +struct asm_stream_cmd_set_pp_params { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* The memory mapping header to be used when sending out of band */ + struct mem_mapping_hdr mem_hdr; + + /* The total size of the payload, including the parameter header */ + u32 payload_size; + + /* The parameter data to be filled when sent inband. Parameter data + * must be pre-packed with parameter header and then copied here. Use + * q6core_pack_pp_params to pack the header and param data correctly. + */ + u32 param_data[0]; +} __packed; + +#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2 +#define ASM_STREAM_CMD_GET_PP_PARAMS_V3 0x0001320E + +struct asm_stream_cmd_get_pp_params_v2 { + u32 data_payload_addr_lsw; + /* LSW of the parameter data payload address. */ + u32 data_payload_addr_msw; +/* MSW of the parameter data payload address. + * - Size of the shared memory, if specified, shall be large enough + * to contain the whole ParamData payload, including Module ID, + * Param ID, Param Size, and Param Values + * - Must be set to zero for in-band data + * - In the case of 32 bit Shared memory address, msw field must be + * set to zero. + * - In the case of 36 bit shared memory address, bit 31 to bit 4 of + * msw must be set to zero. + */ + + u32 mem_map_handle; +/* Supported Values: Any. +* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS +* command. +* if mmhandle is NULL, the ParamData payloads in the ACK are within the +* message payload (in-band). +* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the +* address specified in the address msw and lsw. +* (out-of-band). +*/ + + u32 module_id; + /* Unique module ID. */ + + u32 param_id; + /* Unique parameter ID. */ + + u16 param_max_size; +/* Maximum data size of the module_id/param_id combination. This + * is a multiple of 4 bytes. + */ + + + u16 reserved; +/* Reserved for backward compatibility. Clients must set this +* field to zero. +*/ + +} __packed; + +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10 + +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM_V2 0x00013218 + +struct asm_stream_cmd_set_encdec_param_v2 { + u16 service_id; + /* 0 - ASM_ENCODER_SVC; 1 - ASM_DECODER_SVC */ + + u16 reserved; + + u32 param_id; + /* ID of the parameter. */ + + u32 param_size; + /* + * Data size of this parameter, in bytes. The size is a multiple + * of 4 bytes. + */ +} __packed; + +#define ASM_STREAM_CMD_REGISTER_ENCDEC_EVENTS 0x00013219 + +#define ASM_STREAM_CMD_ENCDEC_EVENTS 0x0001321A + +#define AVS_PARAM_ID_RTIC_SHARED_MEMORY_ADDR 0x00013237 + +struct avs_rtic_shared_mem_addr { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param_v2 encdec; + u32 shm_buf_addr_lsw; + /* Lower 32 bit of the RTIC shared memory */ + + u32 shm_buf_addr_msw; + /* Upper 32 bit of the RTIC shared memory */ + + u32 buf_size; + /* Size of buffer */ + + u16 shm_buf_mem_pool_id; + /* ADSP_MEMORY_MAP_SHMEM8_4K_POOL */ + + u16 shm_buf_num_regions; + /* number of regions to map */ + + u32 shm_buf_flag; + /* buffer property flag */ + + struct avs_shared_map_region_payload map_region; + /* memory map region*/ +} __packed; + +#define AVS_PARAM_ID_RTIC_EVENT_ACK 0x00013238 + +struct avs_param_rtic_event_ack { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param_v2 encdec; +} __packed; + +#define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13 + +struct asm_bitrate_param { + u32 bitrate; +/* Maximum supported bitrate. Only the AAC encoder is supported.*/ + +} __packed; + +#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3 +#define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63 + +/* Flag to turn off both SBR and PS processing, if they are + * present in the bitstream. + */ + +#define ASM_AAC_SBR_OFF_PS_OFF (2) + +/* Flag to turn on SBR but turn off PS processing,if they are + * present in the bitstream. + */ + +#define ASM_AAC_SBR_ON_PS_OFF (1) + +/* Flag to turn on both SBR and PS processing, if they are + * present in the bitstream (default behavior). + */ + + +#define ASM_AAC_SBR_ON_PS_ON (0) + +/* Structure for an AAC SBR PS processing flag. */ + +/* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +struct asm_aac_sbr_ps_flag_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + + u32 sbr_ps_flag; +/* Control parameter to enable or disable SBR/PS processing in + * the AAC bitstream. Use the following macros to set this field: + * - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS + * processing, if they are present in the bitstream. + * - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS + * processing, if they are present in the bitstream. + * - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing, + * if they are present in the bitstream (default behavior). + * - All other values are invalid. + * Changes are applied to the next decoded frame. + */ +} __packed; + +#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64 + +/* First single channel element in a dual mono bitstream.*/ +#define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1) + +/* Second single channel element in a dual mono bitstream.*/ +#define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2) + +/* Structure for AAC decoder dual mono channel mapping. */ + + +struct asm_aac_dual_mono_mapping_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + u16 left_channel_sce; + u16 right_channel_sce; + +} __packed; + +#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4 +#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V3 0x0001320F + +struct asm_stream_cmdrsp_get_pp_params_v2 { + u32 status; +} __packed; + +#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73 + +/* Enumeration for both vocals in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_NO_VOCAL (0) + +/* Enumeration for only the left vocal in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_LEFT_VOCAL (1) + +/* Enumeration for only the right vocal in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2) + +/* Enumeration for both vocal channels in a karaoke stream.*/ +#define AC3_KARAOKE_MODE_BOTH_VOCAL (3) +#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74 +/* Enumeration for the Custom Analog mode.*/ +#define AC3_DRC_MODE_CUSTOM_ANALOG (0) + +/* Enumeration for the Custom Digital mode.*/ +#define AC3_DRC_MODE_CUSTOM_DIGITAL (1) +/* Enumeration for the Line Out mode (light compression).*/ +#define AC3_DRC_MODE_LINE_OUT (2) + +/* Enumeration for the RF remodulation mode (heavy compression).*/ +#define AC3_DRC_MODE_RF_REMOD (3) +#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75 + +/* Enumeration for playing dual mono in stereo mode.*/ +#define AC3_DUAL_MONO_MODE_STEREO (0) + +/* Enumeration for playing left mono.*/ +#define AC3_DUAL_MONO_MODE_LEFT_MONO (1) + +/* Enumeration for playing right mono.*/ +#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2) + +/* Enumeration for mixing both dual mono channels and playing them.*/ +#define AC3_DUAL_MONO_MODE_MIXED_MONO (3) +#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76 + +/* Enumeration for using the Downmix mode indicated in the bitstream. */ + +#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0) + +/* Enumeration for Surround Compatible mode (preserves the + * surround information). + */ + +#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1) +/* Enumeration for Mono Compatible mode (if the output is to be + * further downmixed to mono). + */ + +#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2) + +/* ID of the AC3 PCM scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78 + +/* ID of the AC3 DRC boost scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79 + +/* ID of the AC3 DRC cut scale factor parameter in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A + +/* Structure for AC3 Generic Parameter. */ + +/* Payload of the AC3 parameters in the + * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + */ +struct asm_ac3_generic_param { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + u32 generic_parameter; +/* AC3 generic parameter. Select from one of the following + * possible values. + * + * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are: + * - AC3_KARAOKE_MODE_NO_VOCAL + * - AC3_KARAOKE_MODE_LEFT_VOCAL + * - AC3_KARAOKE_MODE_RIGHT_VOCAL + * - AC3_KARAOKE_MODE_BOTH_VOCAL + * + * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are: + * - AC3_DRC_MODE_CUSTOM_ANALOG + * - AC3_DRC_MODE_CUSTOM_DIGITAL + * - AC3_DRC_MODE_LINE_OUT + * - AC3_DRC_MODE_RF_REMOD + * + * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are: + * - AC3_DUAL_MONO_MODE_STEREO + * - AC3_DUAL_MONO_MODE_LEFT_MONO + * - AC3_DUAL_MONO_MODE_RIGHT_MONO + * - AC3_DUAL_MONO_MODE_MIXED_MONO + * + * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are: + * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT + * - AC3_STEREO_DOWNMIX_MODE_LT_RT + * - AC3_STEREO_DOWNMIX_MODE_LO_RO + * + * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + * + * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + * + * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are + * 0 to 1 in Q31 format. + */ +} __packed; + +/* Enumeration for Raw mode (no downmixing), which specifies + * that all channels in the bitstream are to be played out as is + * without any downmixing. (Default) + */ + +#define WMAPRO_CHANNEL_MASK_RAW (-1) + +/* Enumeration for setting the channel mask to 0. The 7.1 mode + * (Home Theater) is assigned. + */ + + +#define WMAPRO_CHANNEL_MASK_ZERO 0x0000 + +/* Speaker layout mask for one channel (Home Theater, mono). + * - Speaker front center + */ +#define WMAPRO_CHANNEL_MASK_1_C 0x0004 + +/* Speaker layout mask for two channels (Home Theater, stereo). + * - Speaker front left + * - Speaker front right + */ +#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003 + +/* Speaker layout mask for three channels (Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + */ +#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007 + +/* Speaker layout mask for two channels (stereo). + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030 + +/* Speaker layout mask for four channels. + * - Speaker front left + * - Speaker front right + * - Speaker back left + * - Speaker back right +*/ +#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033 + +/* Speaker layout mask for four channels (Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back center +*/ +#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107 +/* Speaker layout mask for five channels. + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037 + +/* Speaker layout mask for five channels (5 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607 +/* Speaker layout mask for six channels (5.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back left + * - Speaker back right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F +/* Speaker layout mask for six channels (5.1 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F +/* Speaker layout mask for six channels (5.1 mode, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker back center + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137 +/* Speaker layout mask for six channels (5.1 mode, Home Theater, + * no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back center + * - Speaker side left + * - Speaker side right + */ +#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707 + +/* Speaker layout mask for seven channels (6.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back left + * - Speaker back right + * - Speaker back center + */ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F + +/* Speaker layout mask for seven channels (6.1 mode, Home + * Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker low frequency + * - Speaker back center + * - Speaker side left + * - Speaker side right +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F + +/* Speaker layout mask for seven channels (6.1 mode, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker front left of center + * - Speaker front right of center +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7 + +/* Speaker layout mask for seven channels (6.1 mode, Home + * Theater, no LFE). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + * - Speaker front left of center + * - Speaker front right of center +*/ +#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637 + +/* Speaker layout mask for eight channels (7.1 mode). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker back left + * - Speaker back right + * - Speaker low frequency + * - Speaker front left of center + * - Speaker front right of center + */ +#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \ + 0x00FF + +/* Speaker layout mask for eight channels (7.1 mode, Home Theater). + * - Speaker front left + * - Speaker front right + * - Speaker front center + * - Speaker side left + * - Speaker side right + * - Speaker low frequency + * - Speaker front left of center + * - Speaker front right of center + * +*/ +#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \ + 0x063F + +#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82 + +/* Maximum number of decoder output channels.*/ +#define MAX_CHAN_MAP_CHANNELS 16 + +#define MAX_CHAN_MAP_CHANNELS_V2 32 + +/* 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; + +/* 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_v2 { + 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_V2 + * + * 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_V2]; +} __packed; + +#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84 + +/* Bitmask for the IEC 61937 enable flag.*/ +#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL) + +/* Shift value for the IEC 61937 enable flag.*/ +#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0 + +/* Bitmask for the IEC 60958 enable flag.*/ +#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL) + +/* Shift value for the IEC 60958 enable flag.*/ +#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1 + +/* Payload format for open write compressed comand */ + +/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED + * comand, which opens a stream for a given session ID and stream ID + * to be rendered in the compressed format. + */ + +struct asm_stream_cmd_open_write_compressed { + struct apr_hdr hdr; + u32 flags; +/* Mode flags that configure the stream for a specific format. + * Supported values: + * - Bit 0 -- IEC 61937 compatibility + * - 0 -- Stream is not in IEC 61937 format + * - 1 -- Stream is in IEC 61937 format + * - Bit 1 -- IEC 60958 compatibility + * - 0 -- Stream is not in IEC 60958 format + * - 1 -- Stream is in IEC 60958 format + * - Bits 2 to 31 -- 0 (Reserved) + * + * For the same stream, bit 0 cannot be set to 0 and bit 1 cannot + * be set to 1. A compressed stream connot have IEC 60958 + * packetization applied without IEC 61937 packetization. + * @note1hang Currently, IEC 60958 packetized input streams are not + * supported. + */ + + + u32 fmt_id; +/* Specifies the media type of the HDMI stream to be opened. + * Supported values: + * - #ASM_MEDIA_FMT_AC3 + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_DTS + * - #ASM_MEDIA_FMT_ATRAC + * - #ASM_MEDIA_FMT_MAT + * + * @note1hang This field must be set to a valid media type even if + * IEC 61937 packetization is not performed by the aDSP. + */ + +} __packed; + + +/* + Indicates the number of samples per channel to be removed from the + beginning of the stream. +*/ +#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67 +/* + Indicates the number of samples per channel to be removed from + the end of the stream. +*/ +#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68 +struct asm_data_cmd_remove_silence { + struct apr_hdr hdr; + u32 num_samples_to_remove; + /**< Number of samples per channel to be removed. + + @values 0 to (2@sscr{32}-1) */ +} __packed; + +#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95 + +struct asm_stream_cmd_open_read_compressed { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode flags that indicate whether meta information per encoded + * frame is to be provided. + * Supported values for bit 4: + * - 0 -- Return data buffer contains all encoded frames only; it does + * not contain frame metadata. + * - 1 -- Return data buffer contains an array of metadata and encoded + * frames. + * - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and + * #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit. + * All other bits are reserved; clients must set them to zero. + */ + + u32 frames_per_buf; +/* Indicates the number of frames that need to be returned per + * read buffer + * Supported values: should be greater than 0 + */ + +} __packed; + +/* adsp_asm_stream_commands.h*/ + + +/* adsp_asm_api.h (no changes)*/ +#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \ + 0x00010BE4 +#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \ + 0x00010D83 +#define ASM_STREAM_POSTPROCOPO_ID_NONE \ + 0x00010C68 +#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \ + 0x00010D8B +#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \ + ASM_STREAM_POSTPROCOPO_ID_DEFAULT +#define ASM_STREAM_PREPROCOPO_ID_NONE \ + ASM_STREAM_POSTPROCOPO_ID_NONE +#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \ + 0x00010312 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \ + 0x00010313 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \ + 0x00010314 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\ + 0x00010704 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\ + 0x0001070D +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\ + 0x0001070E +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\ + 0x0001070F +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRC_V3 \ + 0x11000000 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \ + 0x0001031B +#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP 0x00010315 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316 +#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO 0x00010715 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP 0x00010BE3 +#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317 +#define AUDPROC_MODULE_ID_AIG 0x00010716 +#define AUDPROC_PARAM_ID_AIG_ENABLE 0x00010717 +#define AUDPROC_PARAM_ID_AIG_CONFIG 0x00010718 + +struct Audio_AigParam { + uint16_t mode; +/*< Mode word for enabling AIG/SIG mode . + * Byte offset: 0 + */ + int16_t staticGainL16Q12; +/*< Static input gain when aigMode is set to 1. + * Byte offset: 2 + */ + int16_t initialGainDBL16Q7; +/*<Initial value that the adaptive gain update starts from dB + * Q7 Byte offset: 4 + */ + int16_t idealRMSDBL16Q7; +/*<Average RMS level that AIG attempts to achieve Q8.7 + * Byte offset: 6 + */ + int32_t noiseGateL32; +/*Threshold below which signal is considered as noise and AIG + * Byte offset: 8 + */ + int32_t minGainL32Q15; +/*Minimum gain that can be provided by AIG Q16.15 + * Byte offset: 12 + */ + int32_t maxGainL32Q15; +/*Maximum gain that can be provided by AIG Q16.15 + * Byte offset: 16 + */ + uint32_t gainAtRtUL32Q31; +/*Attack/release time for AIG update Q1.31 + * Byte offset: 20 + */ + uint32_t longGainAtRtUL32Q31; +/*Long attack/release time while updating gain for + * noise/silence Q1.31 Byte offset: 24 + */ + + uint32_t rmsTavUL32Q32; +/* RMS smoothing time constant used for long-term RMS estimate + * Q0.32 Byte offset: 28 + */ + + uint32_t gainUpdateStartTimMsUL32Q0; +/* The waiting time before which AIG starts to apply adaptive + * gain update Q32.0 Byte offset: 32 + */ + +} __packed; + + +#define ADM_MODULE_ID_EANS 0x00010C4A +#define ADM_PARAM_ID_EANS_ENABLE 0x00010C4B +#define ADM_PARAM_ID_EANS_PARAMS 0x00010C4C + +struct adm_eans_enable { + + uint32_t enable_flag; +/*< Specifies whether EANS is disabled (0) or enabled + * (nonzero). + * This is supported only for sampling rates of 8, 12, 16, 24, 32, + * and 48 kHz. It is not supported for sampling rates of 11.025, + * 22.05, or 44.1 kHz. + */ + +} __packed; + + +struct adm_eans_params { + int16_t eans_mode; +/*< Mode word for enabling/disabling submodules. + * Byte offset: 0 + */ + + int16_t eans_input_gain; +/*< Q2.13 input gain to the EANS module. + * Byte offset: 2 + */ + + int16_t eans_output_gain; +/*< Q2.13 output gain to the EANS module. + * Byte offset: 4 + */ + + int16_t eansarget_ns; +/*< Target noise suppression level in dB. + * Byte offset: 6 + */ + + int16_t eans_s_alpha; +/*< Q3.12 over-subtraction factor for stationary noise + * suppression. + * Byte offset: 8 + */ + + int16_t eans_n_alpha; +/* < Q3.12 over-subtraction factor for nonstationary noise + * suppression. + * Byte offset: 10 + */ + + int16_t eans_n_alphamax; +/*< Q3.12 maximum over-subtraction factor for nonstationary + * noise suppression. + * Byte offset: 12 + */ + int16_t eans_e_alpha; +/*< Q15 scaling factor for excess noise suppression. + * Byte offset: 14 + */ + + int16_t eans_ns_snrmax; +/*< Upper boundary in dB for SNR estimation. + * Byte offset: 16 + */ + + int16_t eans_sns_block; +/*< Quarter block size for stationary noise suppression. + * Byte offset: 18 + */ + + int16_t eans_ns_i; +/*< Initialization block size for noise suppression. + * Byte offset: 20 + */ + int16_t eans_np_scale; +/*< Power scale factor for nonstationary noise update. + * Byte offset: 22 + */ + + int16_t eans_n_lambda; +/*< Smoothing factor for higher level nonstationary noise + * update. + * Byte offset: 24 + */ + + int16_t eans_n_lambdaf; +/*< Medium averaging factor for noise update. + * Byte offset: 26 + */ + + int16_t eans_gs_bias; +/*< Bias factor in dB for gain calculation. + * Byte offset: 28 + */ + + int16_t eans_gs_max; +/*< SNR lower boundary in dB for aggressive gain calculation. + * Byte offset: 30 + */ + + int16_t eans_s_alpha_hb; +/*< Q3.12 over-subtraction factor for high-band stationary + * noise suppression. + * Byte offset: 32 + */ + + int16_t eans_n_alphamax_hb; +/*< Q3.12 maximum over-subtraction factor for high-band + * nonstationary noise suppression. + * Byte offset: 34 + */ + + int16_t eans_e_alpha_hb; +/*< Q15 scaling factor for high-band excess noise suppression. + * Byte offset: 36 + */ + + int16_t eans_n_lambda0; +/*< Smoothing factor for nonstationary noise update during + * speech activity. + * Byte offset: 38 + */ + + int16_t thresh; +/*< Threshold for generating a binary VAD decision. + * Byte offset: 40 + */ + + int16_t pwr_scale; +/*< Indirect lower boundary of the noise level estimate. + * Byte offset: 42 + */ + + int16_t hangover_max; +/*< Avoids mid-speech clipping and reliably detects weak speech + * bursts at the end of speech activity. + * Byte offset: 44 + */ + + int16_t alpha_snr; +/*< Controls responsiveness of the VAD. + * Byte offset: 46 + */ + + int16_t snr_diff_max; +/*< Maximum SNR difference. Decreasing this parameter value may + * help in making correct decisions during abrupt changes; however, + * decreasing too much may increase false alarms during long + * pauses/silences. + * Byte offset: 48 + */ + + int16_t snr_diff_min; +/*< Minimum SNR difference. Decreasing this parameter value may + * help in making correct decisions during abrupt changes; however, + * decreasing too much may increase false alarms during long + * pauses/silences. + * Byte offset: 50 + */ + + int16_t init_length; +/*< Defines the number of frames for which a noise level + * estimate is set to a fixed value. + * Byte offset: 52 + */ + + int16_t max_val; +/*< Defines the upper limit of the noise level. + * Byte offset: 54 + */ + + int16_t init_bound; +/*< Defines the initial bounding value for the noise level + * estimate. This is used during the initial segment defined by the + * init_length parameter. + * Byte offset: 56 + */ + + int16_t reset_bound; +/*< Reset boundary for noise tracking. + * Byte offset: 58 + */ + + int16_t avar_scale; +/*< Defines the bias factor in noise estimation. + * Byte offset: 60 + */ + + int16_t sub_nc; +/*< Defines the window length for noise estimation. + * Byte offset: 62 + */ + + int16_t spow_min; +/*< Defines the minimum signal power required to update the + * boundaries for the noise floor estimate. + * Byte offset: 64 + */ + + int16_t eans_gs_fast; +/*< Fast smoothing factor for postprocessor gain. + * Byte offset: 66 + */ + + int16_t eans_gs_med; +/*< Medium smoothing factor for postprocessor gain. + * Byte offset: 68 + */ + + int16_t eans_gs_slow; +/*< Slow smoothing factor for postprocessor gain. + * Byte offset: 70 + */ + + int16_t eans_swb_salpha; +/*< Q3.12 super wideband aggressiveness factor for stationary + * noise suppression. + * Byte offset: 72 + */ + + int16_t eans_swb_nalpha; +/*< Q3.12 super wideband aggressiveness factor for + * nonstationary noise suppression. + * Byte offset: 74 + */ +} __packed; +#define ADM_MODULE_IDX_MIC_GAIN_CTRL 0x00010C35 + +/* @addtogroup audio_pp_param_ids + * ID of the Tx mic gain control parameter used by the + * #ADM_MODULE_IDX_MIC_GAIN_CTRL module. + * @messagepayload + * @structure{admx_mic_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex} + */ +#define ADM_PARAM_IDX_MIC_GAIN 0x00010C36 + +/* Structure for a Tx mic gain parameter for the mic gain + * control module. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the + * Tx Mic Gain Control module. + */ +struct admx_mic_gain { + uint16_t tx_mic_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero. */ +} __packed; + +/* end_addtogroup audio_pp_param_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the Rx Codec Gain Control module. + * + * This module supports the following parameter ID: + * - #ADM_PARAM_ID_RX_CODEC_GAIN + */ +#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL 0x00010C37 + +/* @addtogroup audio_pp_param_ids + * ID of the Rx codec gain control parameter used by the + * #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module. + * + * @messagepayload + * @structure{adm_rx_codec_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex} +*/ +#define ADM_PARAM_ID_RX_CODEC_GAIN 0x00010C38 + +/* Structure for the Rx common codec gain control module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter + * in the Rx Codec Gain Control module. + */ + + +struct adm_rx_codec_gain { + uint16_t rx_codec_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_param_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the HPF Tuning Filter module on the Tx path. + * This module supports the following parameter IDs: + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN + * - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS + */ +#define ADM_MODULE_ID_HPF_IIRX_FILTER 0x00010C3D + +/* @addtogroup audio_pp_param_ids */ +/* ID of the Tx HPF IIR filter enable parameter used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_enable_cfg} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG 0x00010C3E + +/* ID of the Tx HPF IIR filter pregain parameter used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_pre_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN 0x00010C3F + +/* ID of the Tx HPF IIR filter configuration parameters used by the + * #ADM_MODULE_ID_HPF_IIRX_FILTER module. + * @parspace Message payload + * @structure{adm_hpfx_iir_filter_cfg_params} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA + * RAMS.tex} + */ +#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS 0x00010C40 + +/* Structure for enabling a configuration parameter for + * the HPF IIR tuning filter module on the Tx path. + */ + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG + * parameter in the Tx path HPF Tuning Filter module. + */ +struct adm_hpfx_iir_filter_enable_cfg { + uint32_t enable_flag; +/*< Specifies whether the HPF tuning filter is disabled (0) or + * enabled (nonzero). + */ +} __packed; + + +/* Structure for the pregain parameter for the HPF + IIR tuning filter module on the Tx path. */ + + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter + * in the Tx path HPF Tuning Filter module. + */ +struct adm_hpfx_iir_filter_pre_gain { + uint16_t pre_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + + +/* Structure for the configuration parameter for the + HPF IIR tuning filter module on the Tx path. */ + + +/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS + * parameters in the Tx path HPF Tuning Filter module. \n + * \n + * This structure is followed by tuning filter coefficients as follows: \n + * - Sequence of int32_t FilterCoeffs. + * Each band has five coefficients, each in int32_t format in the order of + * b0, b1, b2, a1, a2. + * - Sequence of int16_t NumShiftFactor. + * One int16_t per band. The numerator shift factor is related to the Q + * factor of the filter coefficients. + * - Sequence of uint16_t PanSetting. + * One uint16_t for each band to indicate application of the filter to + * left (0), right (1), or both (2) channels. + */ +struct adm_hpfx_iir_filter_cfg_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @addtogroup audio_pp_module_ids */ +/* ID of the Tx path IIR Tuning Filter module. + * This module supports the following parameter IDs: + * - #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG + */ +#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41 + +/* ID of the Rx path IIR Tuning Filter module for the left channel. + * The parameter IDs of the IIR tuning filter module + * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning + * filter. + * + * Pan parameters are not required for this per-channel IIR filter; the pan + * parameters are ignored by this module. + */ +#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER 0x00010705 + +/* ID of the the Rx path IIR Tuning Filter module for the right + * channel. + * The parameter IDs of the IIR tuning filter module + * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx + * tuning filter. + * + * Pan parameters are not required for this per-channel IIR filter; + * the pan parameters are ignored by this module. + */ +#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER 0x00010706 + +/* end_addtogroup audio_pp_module_ids */ + +/* @addtogroup audio_pp_param_ids */ + +/* ID of the Tx IIR filter enable parameter used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_enable_cfg} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG 0x00010C42 + +/* ID of the Tx IIR filter pregain parameter used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_pre_gain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN 0x00010C43 + +/* ID of the Tx IIR filter configuration parameters used by the + * #ADM_MODULE_IDX_IIR_FILTER module. + * @parspace Message payload + * @structure{admx_iir_filter_cfg_params} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex} + */ +#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS 0x00010C44 + +/* Structure for enabling the configuration parameter for the + * IIR filter module on the Tx path. + */ + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG + * parameter in the Tx Path IIR Tuning Filter module. + */ + +struct admx_iir_filter_enable_cfg { + uint32_t enable_flag; +/*< Specifies whether the IIR tuning filter is disabled (0) or + * enabled (nonzero). + */ + +} __packed; + + +/* Structure for the pregain parameter for the + * IIR filter module on the Tx path. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN + * parameter in the Tx Path IIR Tuning Filter module. + */ + +struct admx_iir_filter_pre_gain { + uint16_t pre_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + + +/* Structure for the configuration parameter for the + * IIR filter module on the Tx path. + */ + + +/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS + * parameter in the Tx Path IIR Tuning Filter module. \n + * \n + * This structure is followed by the HPF IIR filter coefficients on + * the Tx path as follows: \n + * - Sequence of int32_t ulFilterCoeffs. Each band has five + * coefficients, each in int32_t format in the order of b0, b1, b2, + * a1, a2. + * - Sequence of int16_t sNumShiftFactor. One int16_t per band. The + * numerator shift factor is related to the Q factor of the filter + * coefficients. + * - Sequence of uint16_t usPanSetting. One uint16_t for each band + * to indicate if the filter is applied to left (0), right (1), or + * both (2) channels. + */ +struct admx_iir_filter_cfg_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the QEnsemble module. + * This module supports the following parameter IDs: + * - #ADM_PARAM_ID_QENSEMBLE_ENABLE + * - #ADM_PARAM_ID_QENSEMBLE_BACKGAIN + * - #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE + */ +#define ADM_MODULE_ID_QENSEMBLE 0x00010C59 + +/* @addtogroup audio_pp_param_ids */ +/* ID of the QEnsemble enable parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_enable} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_ENABLE 0x00010C60 + +/* ID of the QEnsemble back gain parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_param_backgain} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN 0x00010C61 + +/* ID of the QEnsemble new angle parameter used by the + * #ADM_MODULE_ID_QENSEMBLE module. + * @messagepayload + * @structure{adm_qensemble_param_set_new_angle} + * @tablespace + * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex} + */ +#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE 0x00010C62 + +/* Structure for enabling the configuration parameter for the + * QEnsemble module. + */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE + * parameter used by the QEnsemble module. + */ +struct adm_qensemble_enable { + uint32_t enable_flag; +/*< Specifies whether the QEnsemble module is disabled (0) or enabled + * (nonzero). + */ +} __packed; + + +/* Structure for the background gain for the QEnsemble module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN + * parameter used by + * the QEnsemble module. + */ +struct adm_qensemble_param_backgain { + int16_t back_gain; +/*< Linear gain in Q15 format. + * Supported values: 0 to 32767 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; +/* Structure for setting a new angle for the QEnsemble module. */ + + +/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE + * parameter used + * by the QEnsemble module. + */ +struct adm_qensemble_param_set_new_angle { + int16_t new_angle; +/*< New angle in degrees. + * Supported values: 0 to 359 + */ + + int16_t time_ms; +/*< Transition time in milliseconds to set the new angle. + * Supported values: 0 to 32767 + */ +} __packed; + + +#define ADM_CMD_GET_PP_TOPO_MODULE_LIST 0x00010349 +#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST 0x00010350 +#define ADM_CMD_GET_PP_TOPO_MODULE_LIST_V2 0x00010360 +#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST_V2 0x00010361 +#define AUDPROC_PARAM_ID_ENABLE 0x00010904 +/* + * Payload of the ADM_CMD_GET_PP_TOPO_MODULE_LIST command. + */ +struct adm_cmd_get_pp_topo_module_list { + struct apr_hdr apr_hdr; + + /* The memory mapping header to be used when requesting out of band */ + struct mem_mapping_hdr mem_hdr; + + /* + * Maximum data size of the list of modules. This + * field is a multiple of 4 bytes. + */ + uint32_t param_max_size; +} __packed; + +struct audproc_topology_module_id_info_t { + uint32_t num_modules; +} __packed; + +/* end_addtogroup audio_pp_module_ids */ + +/* @ingroup audio_pp_module_ids + * ID of the Volume Control module pre/postprocessing block. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + * - #ASM_PARAM_ID_MULTICHANNEL_GAIN + * - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG + * - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS + * - #ASM_PARAM_ID_MULTICHANNEL_GAIN + * - #ASM_PARAM_ID_MULTICHANNEL_MUTE + */ +#define ASM_MODULE_ID_VOL_CTRL 0x00010BFE +#define ASM_MODULE_ID_VOL_CTRL2 0x00010910 +#define AUDPROC_MODULE_ID_VOL_CTRL ASM_MODULE_ID_VOL_CTRL + +/* @addtogroup audio_pp_param_ids */ +/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL + * module. + * @messagepayload + * @structure{asm_volume_ctrl_master_gain} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN 0x00010BFF +#define AUDPROC_PARAM_ID_VOL_CTRL_MASTER_GAIN ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + +/* ID of the left/right channel gain parameter used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_volume_ctrl_lr_chan_gain} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MULTICHANNEL_GAIN.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN 0x00010C00 + +/* ID of the mute configuration parameter used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_volume_ctrl_mute_config} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex} + */ +#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG 0x00010C01 + +/* ID of the soft stepping volume parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + * @messagepayload + * @structure{asm_soft_step_volume_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET + * ERS.tex} + */ +#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS 0x00010C29 +#define AUDPROC_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS\ + ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + +/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL + * module. + */ +#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A + +/* ID of the multiple-channel volume control parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + */ +#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713 + +/* ID of the multiple-channel mute configuration parameters used by the + * #ASM_MODULE_ID_VOL_CTRL module. + */ + +#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714 + +/* Structure for the master gain parameter for a volume control + * module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN + * parameter used by the Volume Control module. + */ + + + +struct asm_volume_ctrl_master_gain { + uint16_t master_gain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero. + */ +} __packed; + + +struct asm_volume_ctrl_lr_chan_gain { + uint16_t l_chan_gain; + /*< Linear gain in Q13 format for the left channel. */ + + uint16_t r_chan_gain; + /*< Linear gain in Q13 format for the right channel.*/ +} __packed; + +struct audproc_chmixer_param_coeff { + uint32_t index; + uint16_t num_output_channels; + uint16_t num_input_channels; + uint32_t payload[0]; +} __packed; + + +/* ID of the Multichannel Volume Control parameters used by + * AUDPROC_MODULE_ID_VOL_CTRL. + */ +#define AUDPROC_PARAM_ID_MULTICHANNEL_GAIN 0x00010713 + +/* Payload of the AUDPROC_PARAM_ID_MULTICHANNEL_GAIN channel type/gain + * pairs used by the Volume Control module. + * This structure immediately follows the + * audproc_volume_ctrl_multichannel_gain_t structure. + */ +struct audproc_volume_ctrl_channel_type_gain_pair { + uint8_t channel_type; + /* Channel type for which the gain setting is to be applied. */ + + uint8_t reserved1; + uint8_t reserved2; + uint8_t reserved3; + + uint32_t gain; + /* Gain value for this channel in Q28 format. */ +} __packed; + +/* Payload of the AUDPROC_PARAM_ID_MULTICHANNEL_MUTE parameters used by + * the Volume Control module. + */ +#define ASM_MAX_CHANNELS 8 +struct audproc_volume_ctrl_multichannel_gain { + 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. + */ + + struct audproc_volume_ctrl_channel_type_gain_pair + gain_data[ASM_MAX_CHANNELS]; + /* Array of channel type/mute setting pairs. */ +} __packed; + +/* Structure for the mute configuration parameter for a + volume control module. */ + + +/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG + * parameter used by the Volume Control module. + */ + + +struct asm_volume_ctrl_mute_config { + uint32_t mute_flag; +/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/ + +} __packed; + +/* + * Supported parameters for a soft stepping linear ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0 + +/* + * Exponential ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1 + +/* + * Logarithmic ramping curve. + */ +#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2 + +/* Structure for holding soft stepping volume parameters. */ + + +/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS + * parameters used by the Volume Control module. + */ +struct asm_soft_step_volume_params { + uint32_t period; +/*< Period in milliseconds. + * Supported values: 0 to 15000 + */ + + uint32_t step; +/*< Step in microseconds. + * Supported values: 0 to 15000000 + */ + + uint32_t ramping_curve; +/*< Ramping curve type. + * Supported values: + * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR + * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP + * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG + */ +} __packed; + + +/* Structure for holding soft pause parameters. */ + + +/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS + * parameters used by the Volume Control module. + */ + + +struct asm_soft_pause_params { + uint32_t enable_flag; +/*< Specifies whether soft pause is disabled (0) or enabled + * (nonzero). + */ + + + + uint32_t period; +/*< Period in milliseconds. + * Supported values: 0 to 15000 + */ + + uint32_t step; +/*< Step in microseconds. + * Supported values: 0 to 15000000 + */ + + uint32_t ramping_curve; +/*< Ramping curve. + * Supported values: + * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR + * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP + * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG + */ +} __packed; + + +/* Maximum number of channels.*/ +#define VOLUME_CONTROL_MAX_CHANNELS 8 + +/* Structure for holding one channel type - gain pair. */ + + +/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel + * type/gain pairs used by the Volume Control module. \n \n This + * structure immediately follows the + * asm_volume_ctrl_multichannel_gain structure. + */ + + +struct asm_volume_ctrl_channeltype_gain_pair { + uint8_t channeltype; + /* + * Channel type for which the gain setting is to be applied. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_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 { + uint32_t num_channels; + /* + * Number of channels for which gain values are provided. Any + * channels present in the data for which gain is not provided are + * set to unity gain. + * Supported values: 1 to 8 + */ + + struct asm_volume_ctrl_channeltype_gain_pair + gain_data[VOLUME_CONTROL_MAX_CHANNELS]; + /* Array of channel type/gain pairs.*/ +} __packed; + + +/* Structure for holding one channel type - mute pair. */ + + +/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel + * type/mute setting pairs used by the Volume Control module. \n \n + * This structure immediately follows the + * asm_volume_ctrl_multichannel_mute structure. + */ + + +struct asm_volume_ctrl_channelype_mute_pair { + uint8_t channelype; +/*< Channel type for which the mute setting is to be applied. + * Supported values: + * - #PCM_CHANNEL_L + * - #PCM_CHANNEL_R + * - #PCM_CHANNEL_C + * - #PCM_CHANNEL_LS + * - #PCM_CHANNEL_RS + * - #PCM_CHANNEL_LFE + * - #PCM_CHANNEL_CS + * - #PCM_CHANNEL_LB + * - #PCM_CHANNEL_RB + * - #PCM_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 { + uint32_t num_channels; +/*< Number of channels for which mute configuration is + * provided. Any channels present in the data for which mute + * configuration is not provided are set to unmute. + * Supported values: 1 to 8 + */ + +struct asm_volume_ctrl_channelype_mute_pair + mute_data[VOLUME_CONTROL_MAX_CHANNELS]; + /*< Array of channel type/mute setting pairs.*/ +} __packed; +/* end_addtogroup audio_pp_param_ids */ + +/* audio_pp_module_ids + * ID of the IIR Tuning Filter module. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG + * - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN + * - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS + */ +#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02 + +/* @addtogroup audio_pp_param_ids */ +/* ID of the IIR tuning filter enable parameter used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + * @messagepayload + * @structure{asm_iiruning_filter_enable} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO + * NFIG.tex} + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03 + +/* ID of the IIR tuning filter pregain parameter used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04 + +/* ID of the IIR tuning filter configuration parameters used by the + * #ASM_MODULE_ID_IIRUNING_FILTER module. + */ +#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05 + +/* Structure for an enable configuration parameter for an + * IIR tuning filter module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG + * parameter used by the IIR Tuning Filter module. + */ +struct asm_iiruning_filter_enable { + uint32_t enable_flag; +/*< Specifies whether the IIR tuning filter is disabled (0) or + * enabled (1). + */ +} __packed; + +/* Structure for the pregain parameter for an IIR tuning filter module. */ + + +/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN + * parameters used by the IIR Tuning Filter module. + */ +struct asm_iiruning_filter_pregain { + uint16_t pregain; + /*< Linear gain in Q13 format. */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* Structure for the configuration parameter for an IIR tuning filter + * module. + */ + + +/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS + * parameters used by the IIR Tuning Filter module. \n + * \n + * This structure is followed by the IIR filter coefficients: \n + * - Sequence of int32_t FilterCoeffs \n + * Five coefficients for each band. Each coefficient is in int32_t format, in + * the order of b0, b1, b2, a1, a2. + * - Sequence of int16_t NumShiftFactor \n + * One int16_t per band. The numerator shift factor is related to the Q + * factor of the filter coefficients. + * - Sequence of uint16_t PanSetting \n + * One uint16_t per band, indicating if the filter is applied to left (0), + * right (1), or both (2) channels. + */ +struct asm_iir_filter_config_params { + uint16_t num_biquad_stages; +/*< Number of bands. + * Supported values: 0 to 20 + */ + + uint16_t reserved; + /*< Clients must set this field to zero.*/ +} __packed; + +/* audio_pp_module_ids + * ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx + * paths. + * This module supports the following parameter IDs: + * - #ASM_PARAM_ID_MBDRC_ENABLE + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + */ +#define ASM_MODULE_ID_MBDRC 0x00010C06 + +/* audio_pp_param_ids */ +/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module. + * @messagepayload + * @structure{asm_mbdrc_enable} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex} + */ +#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07 + +/* ID of the MBDRC configuration parameters used by the + * #ASM_MODULE_ID_MBDRC module. + * @messagepayload + * @structure{asm_mbdrc_config_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex} + * + * @parspace Sub-band DRC configuration parameters + * @structure{asm_subband_drc_config_params} + * @tablespace + * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex} + * + * @keep{6} + * To obtain legacy ADRC from MBDRC, use the calibration tool to: + * + * - Enable MBDRC (EnableFlag = TRUE) + * - Set number of bands to 1 (uiNumBands = 1) + * - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1) + * - Clear the first band mute flag (MuteFlag[0] = 0) + * - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000) + * - Use the legacy ADRC parameters to calibrate the rest of the MBDRC + * parameters. + */ +#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08 + +/* end_addtogroup audio_pp_param_ids */ + +/* audio_pp_module_ids + * ID of the MMBDRC module version 2 pre/postprocessing block. + * This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in + * the length of the filters used in each sub-band. + * This module supports the following parameter ID: + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 + */ +#define ASM_MODULE_ID_MBDRCV2 0x0001070B + +/* @addtogroup audio_pp_param_ids */ +/* ID of the configuration parameters used by the + * #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure + * of the MBDRC v2 pre/postprocessing block. + * The update to this configuration structure from the original + * MBDRC is the number of filter coefficients in the filter + * structure. The sequence for is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t + * padding + * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + + * uint16_t padding + * This block uses the same parameter structure as + * #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS. + */ +#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \ + 0x0001070C + +#define ASM_MODULE_ID_MBDRCV3 0x0001090B +/* + * ID of the MMBDRC module version 3 pre/postprocessing block. + * This module differs from MBDRCv2 (#ASM_MODULE_ID_MBDRCV2) in + * that it supports both 16- and 24-bit data. + * This module supports the following parameter ID: + * - #ASM_PARAM_ID_MBDRC_ENABLE + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_V3 + * - #ASM_PARAM_ID_MBDRC_FILTER_XOVER_FREQS + */ + +/* Structure for the enable parameter for an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the + * MBDRC module. + */ +struct asm_mbdrc_enable { + uint32_t enable_flag; +/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/ +} __packed; + +/* Structure for the configuration parameters for an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS + * parameters used by the MBDRC module. \n \n Following this + * structure is the payload for sub-band DRC configuration + * parameters (asm_subband_drc_config_params). This sub-band + * structure must be repeated for each band. + */ + + +struct asm_mbdrc_config_params { + uint16_t num_bands; +/*< Number of bands. + * Supported values: 1 to 5 + */ + + int16_t limiterhreshold; +/*< Threshold in decibels for the limiter output. + * Supported values: -72 to 18 \n + * Recommended value: 3994 (-0.22 db in Q3.12 format) + */ + + int16_t limiter_makeup_gain; +/*< Makeup gain in decibels for the limiter output. + * Supported values: -42 to 42 \n + * Recommended value: 256 (0 dB in Q7.8 format) + */ + + int16_t limiter_gc; +/*< Limiter gain recovery coefficient. + * Supported values: 0.5 to 0.99 \n + * Recommended value: 32440 (0.99 in Q15 format) + */ + + int16_t limiter_delay; +/*< Limiter delay in samples. + * Supported values: 0 to 10 \n + * Recommended value: 262 (0.008 samples in Q15 format) + */ + + int16_t limiter_max_wait; +/*< Maximum limiter waiting time in samples. + * Supported values: 0 to 10 \n + * Recommended value: 262 (0.008 samples in Q15 format) + */ +} __packed; + +/* DRC configuration structure for each sub-band of an MBDRC module. */ + + +/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC + * configuration parameters for each sub-band in the MBDRC module. + * After this DRC structure is configured for valid bands, the next + * MBDRC setparams expects the sequence of sub-band MBDRC filter + * coefficients (the length depends on the number of bands) plus the + * mute flag for that band plus uint16_t padding. + * + * @keep{10} + * The filter coefficient and mute flag are of type int16_t: + * - FIR coefficient = int16_t firFilter + * - Mute flag = int16_t fMuteFlag + * + * The sequence is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding + * - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding + * + * For improved filterbank, the sequence is as follows: + * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding + * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding + * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding + * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding + * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding + */ +struct asm_subband_drc_config_params { + int16_t drc_stereo_linked_flag; +/*< Specifies whether all stereo channels have the same applied + * dynamics (1) or if they process their dynamics independently (0). + * Supported values: + * - 0 -- Not linked + * - 1 -- Linked + */ + + int16_t drc_mode; +/*< Specifies whether DRC mode is bypassed for sub-bands. + * Supported values: + * - 0 -- Disabled + * - 1 -- Enabled + */ + + int16_t drc_down_sample_level; +/*< DRC down sample level. + * Supported values: @ge 1 + */ + + int16_t drc_delay; +/*< DRC delay in samples. + * Supported values: 0 to 1200 + */ + + uint16_t drc_rmsime_avg_const; +/*< RMS signal energy time-averaging constant. + * Supported values: 0 to 2^16-1 + */ + + uint16_t drc_makeup_gain; +/*< DRC makeup gain in decibels. + * Supported values: 258 to 64917 + */ + /* Down expander settings */ + int16_t down_expdrhreshold; +/*< Down expander threshold. + * Supported Q7 format values: 1320 to up_cmpsrhreshold + */ + + int16_t down_expdr_slope; +/*< Down expander slope. + * Supported Q8 format values: -32768 to 0. + */ + + uint32_t down_expdr_attack; +/*< Down expander attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t down_expdr_release; +/*< Down expander release constant. + * Supported Q31 format values: 19685 to 2^31 + */ + + uint16_t down_expdr_hysteresis; +/*< Down expander hysteresis constant. + * Supported Q14 format values: 1 to 32690 + */ + + uint16_t reserved; + /*< Clients must set this field to zero. */ + + int32_t down_expdr_min_gain_db; +/*< Down expander minimum gain. + * Supported Q23 format values: -805306368 to 0. + */ + + /* Up compressor settings */ + + int16_t up_cmpsrhreshold; +/*< Up compressor threshold. + * Supported Q7 format values: down_expdrhreshold to + * down_cmpsrhreshold. + */ + + uint16_t up_cmpsr_slope; +/*< Up compressor slope. + * Supported Q16 format values: 0 to 64881. + */ + + uint32_t up_cmpsr_attack; +/*< Up compressor attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t up_cmpsr_release; +/*< Up compressor release constant. + * Supported Q31 format values: 19685 to 2^31. + */ + + uint16_t up_cmpsr_hysteresis; +/*< Up compressor hysteresis constant. + * Supported Q14 format values: 1 to 32690. + */ + + /* Down compressor settings */ + + int16_t down_cmpsrhreshold; +/*< Down compressor threshold. + * Supported Q7 format values: up_cmpsrhreshold to 11560. + */ + + uint16_t down_cmpsr_slope; +/*< Down compressor slope. + * Supported Q16 format values: 0 to 64881. + */ + + uint16_t reserved1; +/*< Clients must set this field to zero. */ + + uint32_t down_cmpsr_attack; +/*< Down compressor attack constant. + * Supported Q31 format values: 196844 to 2^31. + */ + + uint32_t down_cmpsr_release; +/*< Down compressor release constant. + * Supported Q31 format values: 19685 to 2^31. + */ + + uint16_t down_cmpsr_hysteresis; +/*< Down compressor hysteresis constant. + * Supported Q14 values: 1 to 32690. + */ + + uint16_t reserved2; +/*< Clients must set this field to zero.*/ +} __packed; + +#define ASM_MODULE_ID_EQUALIZER 0x00010C27 +#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28 + +#define ASM_MAX_EQ_BANDS 12 + +struct asm_eq_per_band_params { + uint32_t band_idx; +/*< Band index. + * Supported values: 0 to 11 + */ + + uint32_t filterype; +/*< Type of filter. + * Supported values: + * - #ASM_PARAM_EQYPE_NONE + * - #ASM_PARAM_EQ_BASS_BOOST + * - #ASM_PARAM_EQ_BASS_CUT + * - #ASM_PARAM_EQREBLE_BOOST + * - #ASM_PARAM_EQREBLE_CUT + * - #ASM_PARAM_EQ_BAND_BOOST + * - #ASM_PARAM_EQ_BAND_CUT + */ + + uint32_t center_freq_hz; + /*< Filter band center frequency in Hertz. */ + + int32_t filter_gain; +/*< Filter band initial gain. + * Supported values: +12 to -12 dB in 1 dB increments + */ + + int32_t q_factor; +/*< Filter band quality factor expressed as a Q8 number, i.e., a + * fixed-point number with q factor of 8. For example, 3000/(2^8). + */ +} __packed; + +struct asm_eq_params { + uint32_t enable_flag; +/*< Specifies whether the equalizer module is disabled (0) or enabled + * (nonzero). + */ + + uint32_t num_bands; +/*< Number of bands. + * Supported values: 1 to 12 + */ + struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS]; + +} __packed; + +/* No equalizer effect.*/ +#define ASM_PARAM_EQYPE_NONE 0 + +/* Bass boost equalizer effect.*/ +#define ASM_PARAM_EQ_BASS_BOOST 1 + +/*Bass cut equalizer effect.*/ +#define ASM_PARAM_EQ_BASS_CUT 2 + +/* Treble boost equalizer effect */ +#define ASM_PARAM_EQREBLE_BOOST 3 + +/* Treble cut equalizer effect.*/ +#define ASM_PARAM_EQREBLE_CUT 4 + +/* Band boost equalizer effect.*/ +#define ASM_PARAM_EQ_BAND_BOOST 5 + +/* Band cut equalizer effect.*/ +#define ASM_PARAM_EQ_BAND_CUT 6 + +/* Get & set params */ +#define VSS_ICOMMON_CMD_SET_PARAM_V2 0x0001133D +#define VSS_ICOMMON_CMD_GET_PARAM_V2 0x0001133E +#define VSS_ICOMMON_RSP_GET_PARAM 0x00011008 +#define VSS_ICOMMON_CMD_SET_PARAM_V3 0x00013245 +#define VSS_ICOMMON_CMD_GET_PARAM_V3 0x00013246 +#define VSS_ICOMMON_RSP_GET_PARAM_V3 0x00013247 + +#define VSS_MAX_AVCS_NUM_SERVICES 25 + +/** ID of the Bass Boost module. + This module supports the following parameter IDs: + - #AUDPROC_PARAM_ID_BASS_BOOST_ENABLE + - #AUDPROC_PARAM_ID_BASS_BOOST_MODE + - #AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH +*/ +#define AUDPROC_MODULE_ID_BASS_BOOST 0x000108A1 +/** ID of the Bass Boost enable parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_ENABLE 0x000108A2 +/** ID of the Bass Boost mode parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_MODE 0x000108A3 +/** ID of the Bass Boost strength parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH 0x000108A4 + +/** ID of the PBE module. + This module supports the following parameter IDs: + - #AUDPROC_PARAM_ID_PBE_ENABLE + - #AUDPROC_PARAM_ID_PBE_PARAM_CONFIG +*/ +#define AUDPROC_MODULE_ID_PBE 0x00010C2A +/** ID of the Bass Boost enable parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_PBE_ENABLE 0x00010C2B +/** ID of the Bass Boost mode parameter used by + AUDPROC_MODULE_ID_BASS_BOOST. +*/ +#define AUDPROC_PARAM_ID_PBE_PARAM_CONFIG 0x00010C49 + +/** ID of the Virtualizer module. This module supports the + following parameter IDs: + - #AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE + - #AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH + - #AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE + - #AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST +*/ +#define AUDPROC_MODULE_ID_VIRTUALIZER 0x000108A5 +/** ID of the Virtualizer enable parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE 0x000108A6 +/** ID of the Virtualizer strength parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH 0x000108A7 +/** ID of the Virtualizer out type parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE 0x000108A8 +/** ID of the Virtualizer out type parameter used by + AUDPROC_MODULE_ID_VIRTUALIZER. +*/ +#define AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST 0x000108A9 + +/** ID of the Reverb module. This module supports the following + parameter IDs: + - #AUDPROC_PARAM_ID_REVERB_ENABLE + - #AUDPROC_PARAM_ID_REVERB_MODE + - #AUDPROC_PARAM_ID_REVERB_PRESET + - #AUDPROC_PARAM_ID_REVERB_WET_MIX + - #AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST + - #AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL + - #AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL + - #AUDPROC_PARAM_ID_REVERB_DECAY_TIME + - #AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO + - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL + - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY + - #AUDPROC_PARAM_ID_REVERB_LEVEL + - #AUDPROC_PARAM_ID_REVERB_DELAY + - #AUDPROC_PARAM_ID_REVERB_DIFFUSION + - #AUDPROC_PARAM_ID_REVERB_DENSITY +*/ +#define AUDPROC_MODULE_ID_REVERB 0x000108AA +/** ID of the Reverb enable parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ENABLE 0x000108AB +/** ID of the Reverb mode parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_MODE 0x000108AC +/** ID of the Reverb preset parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_PRESET 0x000108AD +/** ID of the Reverb wet mix parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_WET_MIX 0x000108AE +/** ID of the Reverb gain adjust parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST 0x000108AF +/** ID of the Reverb room level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL 0x000108B0 +/** ID of the Reverb room hf level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL 0x000108B1 +/** ID of the Reverb decay time parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DECAY_TIME 0x000108B2 +/** ID of the Reverb decay hf ratio parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO 0x000108B3 +/** ID of the Reverb reflections level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL 0x000108B4 +/** ID of the Reverb reflections delay parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY 0x000108B5 +/** ID of the Reverb level parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_LEVEL 0x000108B6 +/** ID of the Reverb delay parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DELAY 0x000108B7 +/** ID of the Reverb diffusion parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DIFFUSION 0x000108B8 +/** ID of the Reverb density parameter used by + AUDPROC_MODULE_ID_REVERB. +*/ +#define AUDPROC_PARAM_ID_REVERB_DENSITY 0x000108B9 + +/** ID of the Popless Equalizer module. This module supports the + following parameter IDs: + - #AUDPROC_PARAM_ID_EQ_ENABLE + - #AUDPROC_PARAM_ID_EQ_CONFIG + - #AUDPROC_PARAM_ID_EQ_NUM_BANDS + - #AUDPROC_PARAM_ID_EQ_BAND_LEVELS + - #AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE + - #AUDPROC_PARAM_ID_EQ_BAND_FREQS + - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE + - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ + - #AUDPROC_PARAM_ID_EQ_BAND_INDEX + - #AUDPROC_PARAM_ID_EQ_PRESET_ID + - #AUDPROC_PARAM_ID_EQ_NUM_PRESETS + - #AUDPROC_PARAM_ID_EQ_GET_PRESET_NAME +*/ +#define AUDPROC_MODULE_ID_POPLESS_EQUALIZER 0x000108BA +/** ID of the Popless Equalizer enable parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_ENABLE 0x000108BB +/** ID of the Popless Equalizer config parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_CONFIG 0x000108BC +/** ID of the Popless Equalizer number of bands parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_NUM_BANDS 0x000108BD +/** ID of the Popless Equalizer band levels parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_LEVELS 0x000108BE +/** ID of the Popless Equalizer band level range parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE 0x000108BF +/** ID of the Popless Equalizer band frequencies parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is + used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_FREQS 0x000108C0 +/** ID of the Popless Equalizer single band frequency range + parameter used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. + This param ID is used for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE 0x000108C1 +/** ID of the Popless Equalizer single band frequency parameter + used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID + is used for set param only. +*/ +#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ 0x000108C2 +/** ID of the Popless Equalizer band index parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. +*/ +#define AUDPROC_PARAM_ID_EQ_BAND_INDEX 0x000108C3 +/** ID of the Popless Equalizer preset id parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_PRESET_ID 0x000108C4 +/** ID of the Popless Equalizer number of presets parameter used + by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_NUM_PRESETS 0x000108C5 +/** ID of the Popless Equalizer preset name parameter used by + AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used + for get param only. +*/ +#define AUDPROC_PARAM_ID_EQ_PRESET_NAME 0x000108C6 + +/* Set Q6 topologies */ +#define ASM_CMD_ADD_TOPOLOGIES 0x00010DBE +#define ADM_CMD_ADD_TOPOLOGIES 0x00010335 +#define AFE_CMD_ADD_TOPOLOGIES 0x000100f8 +/* structure used for both ioctls */ +struct cmd_set_topologies { + struct apr_hdr hdr; + u32 payload_addr_lsw; + /* LSW of parameter data payload address.*/ + u32 payload_addr_msw; + /* MSW of parameter data payload address.*/ + u32 mem_map_handle; + /* Memory map handle returned by mem map command */ + u32 payload_size; + /* Size in bytes of the variable payload in shared memory */ +} __packed; + +/* This module represents the Rx processing of Feedback speaker protection. + * It contains the excursion control, thermal protection, + * analog clip manager features in it. + * This module id will support following param ids. + * - AFE_PARAM_ID_FBSP_MODE_RX_CFG + */ + +#define AFE_MODULE_FB_SPKR_PROT_RX 0x0001021C +#define AFE_MODULE_FB_SPKR_PROT_V2_RX 0x0001025F + +#define AFE_PARAM_ID_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 param_hdr_v3 pdata; + struct afe_sp_th_vi_ftm_params param; +} __packed; + +struct afe_sp_th_vi_get_param_resp { + uint32_t status; + struct param_hdr_v3 pdata; + struct afe_sp_th_vi_ftm_params param; +} __packed; + + +#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 param_hdr_v3 pdata; + struct afe_sp_ex_vi_ftm_params param; +} __packed; + +struct afe_sp_ex_vi_get_param_resp { + uint32_t status; + struct param_hdr_v3 pdata; + struct afe_sp_ex_vi_ftm_params param; +} __packed; + +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_get_vi_calib { + struct apr_hdr hdr; + struct mem_mapping_hdr mem_hdr; + struct param_hdr_v3 pdata; + struct asm_calib_res_cfg res_cfg; +} __packed; + +struct afe_spkr_prot_calib_get_resp { + uint32_t status; + struct param_hdr_v3 pdata; + struct asm_calib_res_cfg res_cfg; +} __packed; + + +/* 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 ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF +/* DTS Eagle */ +#define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C +#define AUDPROC_MODULE_ID_DTS_HPX_POSTMIX 0x0001077B +#define ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX 0x00010DED +#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS 0x10015000 +#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER 0x10015001 + +/* Opcode to set BT address and license for aptx decoder */ +#define APTX_DECODER_BT_ADDRESS 0x00013201 +#define APTX_CLASSIC_DEC_LICENSE_ID 0x00013202 + +struct aptx_dec_bt_addr_cfg { + uint32_t lap; + uint32_t uap; + uint32_t nap; +} __packed; + +struct aptx_dec_bt_dev_addr { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct aptx_dec_bt_addr_cfg bt_addr_cfg; +} __packed; + +struct asm_aptx_dec_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmtblk; + u32 sample_rate; +/* Number of samples per second. + * Supported values: 44100 and 48000 Hz + */ +} __packed; + +/* Q6Core Specific */ +#define AVCS_CMD_GET_FWK_VERSION (0x0001292C) +#define AVCS_CMDRSP_GET_FWK_VERSION (0x0001292D) + +#define AVCS_SERVICE_ID_ALL (0xFFFFFFFF) +#define APRV2_IDS_SERVICE_ID_ADSP_CVP_V (0xB) + +struct avcs_get_fwk_version { + /* + * Indicates the major version of the AVS build. + * This value is incremented on chipset family boundaries. + */ + uint32_t build_major_version; + + /* + * Minor version of the AVS build. + * This value represents the mainline to which the AVS build belongs. + */ + uint32_t build_minor_version; + + /* Indicates the AVS branch version to which the image belongs. */ + uint32_t build_branch_version; + + /* Indicates the AVS sub-branch or customer product line information. */ + uint32_t build_subbranch_version; + + /* Number of supported AVS services in the current build. */ + uint32_t num_services; +}; + +struct avs_svc_api_info { + /* + * APRV2 service IDs for the individual static services. + * + * @values + * - APRV2_IDS_SERVICE_ID_ADSP_CORE_V + * - APRV2_IDS_SERVICE_ID_ADSP_AFE_V + * - APRV2_IDS_SERVICE_ID_ADSP_ASM_V + * - APRV2_IDS_SERVICE_ID_ADSP_ADM_V + * - APRV2_IDS_SERVICE_ID_ADSP_MVM_V + * - APRV2_IDS_SERVICE_ID_ADSP_CVS_V + * - APRV2_IDS_SERVICE_ID_ADSP_CVP_V + * - APRV2_IDS_SERVICE_ID_ADSP_LSM_V + */ + uint32_t service_id; + + /* + * Indicates the API version of the service. + * + * Each new API update that warrants a change on the HLOS side triggers + * an increment in the version. + */ + uint32_t api_version; + + /* + * Indicates the API increments on a sub-branch (not on the mainline). + * + * API branch version numbers can increment independently on different + * sub-branches. + */ + uint32_t api_branch_version; +}; + +struct avcs_fwk_ver_info { + struct avcs_get_fwk_version avcs_fwk_version; + struct avs_svc_api_info services[0]; +} __packed; + +/* LSM Specific */ +#define VW_FEAT_DIM (39) + +#define APRV2_IDS_SERVICE_ID_ADSP_LSM_V (0xD) +#define APRV2_IDS_DOMAIN_ID_ADSP_V (0x4) +#define APRV2_IDS_DOMAIN_ID_APPS_V (0x5) + +#define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS (0x00012A7F) +#define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS (0x00012A80) +#define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS (0x00012A81) +#define LSM_SESSION_CMD_OPEN_TX (0x00012A82) +#define LSM_SESSION_CMD_CLOSE_TX (0x00012A88) +#define LSM_SESSION_CMD_SET_PARAMS (0x00012A83) +#define LSM_SESSION_CMD_SET_PARAMS_V2 (0x00012A8F) +#define LSM_SESSION_CMD_SET_PARAMS_V3 (0x00012A92) +#define LSM_SESSION_CMD_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_SVC_CMD_SET_PARAM_V2 (0x000100fc) +#define AFE_MODULE_CDC_DEV_CFG (0x00010234) +#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG (0x00010235) +#define AFE_PARAM_ID_CDC_REG_CFG (0x00010236) +#define AFE_PARAM_ID_CDC_REG_CFG_INIT (0x00010237) +#define AFE_PARAM_ID_CDC_REG_PAGE_CFG (0x00010296) + +#define AFE_MAX_CDC_REGISTERS_TO_CONFIG (20) + +/* AANC Port Config Specific */ +#define AFE_PARAM_ID_AANC_PORT_CONFIG (0x00010215) +#define AFE_API_VERSION_AANC_PORT_CONFIG (0x1) +#define AANC_TX_MIC_UNUSED (0) +#define AANC_TX_VOICE_MIC (1) +#define AANC_TX_ERROR_MIC (2) +#define AANC_TX_NOISE_MIC (3) +#define AFE_PORT_MAX_CHANNEL_CNT (8) +#define AFE_MODULE_AANC (0x00010214) +#define AFE_PARAM_ID_CDC_AANC_VERSION (0x0001023A) +#define AFE_API_VERSION_CDC_AANC_VERSION (0x1) +#define AANC_HW_BLOCK_VERSION_1 (1) +#define AANC_HW_BLOCK_VERSION_2 (2) + +/*Clip bank selection*/ +#define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1 +#define AFE_CLIP_MAX_BANKS 4 +#define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242 + +struct afe_param_aanc_port_cfg { + /* Minor version used for tracking the version of the module's + * source port configuration. + */ + uint32_t aanc_port_cfg_minor_version; + + /* Sampling rate of the source Tx port. 8k - 192k*/ + uint32_t tx_port_sample_rate; + + /* Channel mapping for the Tx port signal carrying Noise (X), + * Error (E), and Voice (V) signals. + */ + uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT]; + + /* Number of channels on the source Tx port. */ + uint16_t tx_port_num_channels; + + /* Port ID of the Rx path reference signal. */ + uint16_t rx_path_ref_port_id; + + /* Sampling rate of the reference port. 8k - 192k*/ + uint32_t ref_port_sample_rate; +} __packed; + +struct afe_param_id_cdc_aanc_version { + /* Minor version used for tracking the version of the module's + * hw version + */ + uint32_t cdc_aanc_minor_version; + + /* HW version. */ + uint32_t aanc_hw_version; +} __packed; + +struct afe_param_id_clip_bank_sel { + /* Minor version used for tracking the version of the module's + * hw version + */ + uint32_t minor_version; + + /* Number of banks to be read */ + uint32_t num_banks; + + uint32_t bank_map[AFE_CLIP_MAX_BANKS]; +} __packed; + +/* ERROR CODES */ +/* Success. The operation completed with no errors. */ +#define ADSP_EOK 0x00000000 +/* General failure. */ +#define ADSP_EFAILED 0x00000001 +/* Bad operation parameter. */ +#define ADSP_EBADPARAM 0x00000002 +/* Unsupported routine or operation. */ +#define ADSP_EUNSUPPORTED 0x00000003 +/* Unsupported version. */ +#define ADSP_EVERSION 0x00000004 +/* Unexpected problem encountered. */ +#define ADSP_EUNEXPECTED 0x00000005 +/* Unhandled problem occurred. */ +#define ADSP_EPANIC 0x00000006 +/* Unable to allocate resource. */ +#define ADSP_ENORESOURCE 0x00000007 +/* Invalid handle. */ +#define ADSP_EHANDLE 0x00000008 +/* Operation is already processed. */ +#define ADSP_EALREADY 0x00000009 +/* Operation is not ready to be processed. */ +#define ADSP_ENOTREADY 0x0000000A +/* Operation is pending completion. */ +#define ADSP_EPENDING 0x0000000B +/* Operation could not be accepted or processed. */ +#define ADSP_EBUSY 0x0000000C +/* Operation aborted due to an error. */ +#define ADSP_EABORTED 0x0000000D +/* Operation preempted by a higher priority. */ +#define ADSP_EPREEMPTED 0x0000000E +/* Operation requests intervention to complete. */ +#define ADSP_ECONTINUE 0x0000000F +/* Operation requests immediate intervention to complete. */ +#define ADSP_EIMMEDIATE 0x00000010 +/* Operation is not implemented. */ +#define ADSP_ENOTIMPL 0x00000011 +/* Operation needs more data or resources. */ +#define ADSP_ENEEDMORE 0x00000012 +/* Operation does not have memory. */ +#define ADSP_ENOMEMORY 0x00000014 +/* Item does not exist. */ +#define ADSP_ENOTEXIST 0x00000015 +/* Max count for adsp error code sent to HLOS*/ +#define ADSP_ERR_MAX (ADSP_ENOTEXIST + 1) +/* Operation is finished. */ +#define ADSP_ETERMINATED 0x00011174 + +/*bharath, adsp_error_codes.h */ + +/* LPASS clock for I2S Interface */ + +/* Supported OSR clock values */ +#define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ 0xBB8000 +#define Q6AFE_LPASS_OSR_CLK_11_P2896_MHZ 0xAC4400 +#define Q6AFE_LPASS_OSR_CLK_9_P600_MHZ 0x927C00 +#define Q6AFE_LPASS_OSR_CLK_8_P192_MHZ 0x7D0000 +#define Q6AFE_LPASS_OSR_CLK_6_P144_MHZ 0x5DC000 +#define Q6AFE_LPASS_OSR_CLK_4_P096_MHZ 0x3E8000 +#define Q6AFE_LPASS_OSR_CLK_3_P072_MHZ 0x2EE000 +#define Q6AFE_LPASS_OSR_CLK_2_P048_MHZ 0x1F4000 +#define Q6AFE_LPASS_OSR_CLK_1_P536_MHZ 0x177000 +#define Q6AFE_LPASS_OSR_CLK_1_P024_MHZ 0xFA000 +#define Q6AFE_LPASS_OSR_CLK_768_kHZ 0xBB800 +#define Q6AFE_LPASS_OSR_CLK_512_kHZ 0x7D000 +#define Q6AFE_LPASS_OSR_CLK_DISABLE 0x0 + +/* Supported Bit clock values */ +#define Q6AFE_LPASS_IBIT_CLK_12_P288_MHZ 0xBB8000 +#define Q6AFE_LPASS_IBIT_CLK_11_P2896_MHZ 0xAC4400 +#define Q6AFE_LPASS_IBIT_CLK_8_P192_MHZ 0x7D0000 +#define Q6AFE_LPASS_IBIT_CLK_6_P144_MHZ 0x5DC000 +#define Q6AFE_LPASS_IBIT_CLK_4_P096_MHZ 0x3E8000 +#define Q6AFE_LPASS_IBIT_CLK_3_P072_MHZ 0x2EE000 +#define Q6AFE_LPASS_IBIT_CLK_2_P8224_MHZ 0x2b1100 +#define Q6AFE_LPASS_IBIT_CLK_2_P048_MHZ 0x1F4000 +#define Q6AFE_LPASS_IBIT_CLK_1_P536_MHZ 0x177000 +#define Q6AFE_LPASS_IBIT_CLK_1_P4112_MHZ 0x158880 +#define Q6AFE_LPASS_IBIT_CLK_1_P024_MHZ 0xFA000 +#define Q6AFE_LPASS_IBIT_CLK_768_KHZ 0xBB800 +#define Q6AFE_LPASS_IBIT_CLK_512_KHZ 0x7D000 +#define Q6AFE_LPASS_IBIT_CLK_256_KHZ 0x3E800 +#define Q6AFE_LPASS_IBIT_CLK_DISABLE 0x0 + +/* Supported LPASS CLK sources */ +#define Q6AFE_LPASS_CLK_SRC_EXTERNAL 0 +#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1 + +/* Supported LPASS CLK root*/ +#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0 + +enum afe_lpass_clk_mode { + Q6AFE_LPASS_MODE_BOTH_INVALID, + Q6AFE_LPASS_MODE_CLK1_VALID, + Q6AFE_LPASS_MODE_CLK2_VALID, + Q6AFE_LPASS_MODE_BOTH_VALID, +} __packed; + +/* Clock ID Enumeration Define. */ +/* Clock ID for Primary I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100 +/* Clock ID for Primary I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101 +/* Clock ID for Secondary I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102 +/* Clock ID for Secondary I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103 +/* Clock ID for Tertiary I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104 +/* Clock ID for Tertiary I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105 +/* Clock ID for Quartnery I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106 +/* Clock ID for Quartnery I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107 +/* Clock ID for Speaker I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108 +/* Clock ID for Speaker I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109 +/* Clock ID for Speaker I2S OSR */ +#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A + +/* Clock ID for QUINARY I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B +/* Clock ID for QUINARY I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C +/* Clock ID for SENARY I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D +/* Clock ID for SENARY I2S EBIT */ +#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E +/* Clock ID for INT0 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F +/* Clock ID for INT1 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110 +/* Clock ID for INT2 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111 +/* Clock ID for INT3 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112 +/* Clock ID for INT4 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113 +/* Clock ID for INT5 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114 +/* Clock ID for INT6 I2S IBIT */ +#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115 + +/* Clock ID for Primary PCM IBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200 +/* Clock ID for Primary PCM EBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201 +/* Clock ID for Secondary PCM IBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202 +/* Clock ID for Secondary PCM EBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203 +/* Clock ID for Tertiary PCM IBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204 +/* Clock ID for Tertiary PCM EBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205 +/* Clock ID for Quartery PCM IBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206 +/* Clock ID for Quartery PCM EBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207 + +/** Clock ID for Primary TDM IBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200 +/** Clock ID for Primary TDM EBIT */ +#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201 +/** Clock ID for Secondary TDM IBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202 +/** Clock ID for Secondary TDM EBIT */ +#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203 +/** Clock ID for Tertiary TDM IBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204 +/** Clock ID for Tertiary TDM EBIT */ +#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205 +/** Clock ID for Quartery TDM IBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206 +/** Clock ID for Quartery TDM EBIT */ +#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207 + +/* Clock ID for MCLK1 */ +#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300 +/* Clock ID for MCLK2 */ +#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301 +/* Clock ID for MCLK3 */ +#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302 +/* Clock ID for MCLK4 */ +#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304 +/* Clock ID for Internal Digital Codec Core */ +#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303 +/* Clock ID for INT MCLK0 */ +#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305 +/* Clock ID for INT MCLK1 */ +#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306 +/* + * Clock ID for soundwire NPL. + * This is the clock to be used to enable NPL clock for internal Soundwire. + */ +#define AFE_CLOCK_SET_CLOCK_ID_SWR_NPL_CLK 0x307 + +/* Clock ID for AHB HDMI input */ +#define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT 0x400 + +/* Clock ID for SPDIF core */ +#define Q6AFE_LPASS_CLK_ID_SPDIF_CORE 0x500 + + +/* Clock attribute for invalid use (reserved for internal usage) */ +#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0 +/* Clock attribute for no couple case */ +#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1 +/* Clock attribute for dividend couple case */ +#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2 +/* Clock attribute for divisor couple case */ +#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3 +/* Clock attribute for invert and no couple case */ +#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4 +/* Clock set API version */ +#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1 + +struct afe_clk_set { + /* + * Minor version used for tracking clock set. + * @values #AFE_API_VERSION_CLOCK_SET + */ + uint32_t clk_set_minor_version; + + /* + * Clock ID + * @values + * - 0x100 to 0x10A - MSM8996 + * - 0x200 to 0x207 - MSM8996 + * - 0x300 to 0x302 - MSM8996 @tablebulletend + */ + uint32_t clk_id; + + /* + * Clock frequency (in Hertz) to be set. + * @values + * - >= 0 for clock frequency to set @tablebulletend + */ + uint32_t clk_freq_in_hz; + + /* Use to specific divider for two clocks if needed. + * Set to Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO for no divider + * relation clocks + * @values + * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO + * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND + * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR @tablebulletend + */ + uint16_t clk_attri; + + /* + * Specifies the root clock source. + * Currently, only Q6AFE_LPASS_CLK_ROOT_DEFAULT is valid + * @values + * - 0 @tablebulletend + */ + uint16_t clk_root; + + /* + * for enable and disable clock. + * "clk_freq_in_hz", "clk_attri", and "clk_root" + * are ignored in disable clock case. + * @values + * - 0 -- Disabled + * - 1 -- Enabled @tablebulletend + */ + uint32_t enable; +}; + +struct afe_clk_cfg { +/* Minor version used for tracking the version of the I2S + * configuration interface. + * Supported values: #AFE_API_VERSION_I2S_CONFIG + */ + u32 i2s_cfg_minor_version; + +/* clk value 1 in MHz. */ + u32 clk_val1; + +/* clk value 2 in MHz. */ + u32 clk_val2; + +/* clk_src + * #Q6AFE_LPASS_CLK_SRC_EXTERNAL + * #Q6AFE_LPASS_CLK_SRC_INTERNAL + */ + + u16 clk_src; + +/* clk_root -0 for default */ + u16 clk_root; + +/* clk_set_mode + * #Q6AFE_LPASS_MODE_BOTH_INVALID + * #Q6AFE_LPASS_MODE_CLK1_VALID + * #Q6AFE_LPASS_MODE_CLK2_VALID + * #Q6AFE_LPASS_MODE_BOTH_VALID + */ + u16 clk_set_mode; + +/* This param id is used to configure I2S clk */ + u16 reserved; +} __packed; + +/* This param id is used to configure I2S clk */ +#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238 +#define AFE_MODULE_CLOCK_SET 0x0001028F +#define AFE_PARAM_ID_CLOCK_SET 0x00010290 + +enum afe_lpass_digital_clk_src { + Q6AFE_LPASS_DIGITAL_ROOT_INVALID, + Q6AFE_LPASS_DIGITAL_ROOT_PRI_MI2S_OSR, + Q6AFE_LPASS_DIGITAL_ROOT_SEC_MI2S_OSR, + Q6AFE_LPASS_DIGITAL_ROOT_TER_MI2S_OSR, + Q6AFE_LPASS_DIGITAL_ROOT_QUAD_MI2S_OSR, + Q6AFE_LPASS_DIGITAL_ROOT_CDC_ROOT_CLK, +} __packed; + +/* This param id is used to configure internal clk */ +#define AFE_PARAM_ID_INTERNAL_DIGIATL_CDC_CLK_CONFIG 0x00010239 + +struct afe_digital_clk_cfg { +/* Minor version used for tracking the version of the I2S + * configuration interface. + * Supported values: #AFE_API_VERSION_I2S_CONFIG + */ + u32 i2s_cfg_minor_version; + +/* clk value in MHz. */ + u32 clk_val; + +/* INVALID + * PRI_MI2S_OSR + * SEC_MI2S_OSR + * TER_MI2S_OSR + * QUAD_MI2S_OSR + * DIGT_CDC_ROOT + */ + u16 clk_root; + +/* This field must be set to zero. */ + u16 reserved; +} __packed; + +/* + * Opcode for AFE to start DTMF. + */ +#define AFE_PORTS_CMD_DTMF_CTL 0x00010102 + +/** DTMF payload.*/ +struct afe_dtmf_generation_command { + struct apr_hdr hdr; + + /* + * Duration of the DTMF tone in ms. + * -1 -> continuous, + * 0 -> disable + */ + int64_t duration_in_ms; + + /* + * The DTMF high tone frequency. + */ + uint16_t high_freq; + + /* + * The DTMF low tone frequency. + */ + uint16_t low_freq; + + /* + * The DTMF volume setting + */ + uint16_t gain; + + /* + * The number of ports to enable/disable on. + */ + uint16_t num_ports; + + /* + * The Destination ports - array . + * For DTMF on multiple ports, portIds needs to + * be populated numPorts times. + */ + uint16_t port_ids; + + /* + * variable for 32 bit alignment of APR packet. + */ + uint16_t reserved; +} __packed; + +enum afe_config_type { + AFE_SLIMBUS_SLAVE_PORT_CONFIG, + AFE_SLIMBUS_SLAVE_CONFIG, + AFE_CDC_REGISTERS_CONFIG, + AFE_AANC_VERSION, + AFE_CDC_CLIP_REGISTERS_CONFIG, + AFE_CLIP_BANK_SEL, + AFE_CDC_REGISTER_PAGE_CONFIG, + AFE_MAX_CONFIG_TYPES, +}; + +struct afe_param_slimbus_slave_port_cfg { + uint32_t minor_version; + uint16_t slimbus_dev_id; + uint16_t slave_dev_pgd_la; + uint16_t slave_dev_intfdev_la; + uint16_t bit_width; + uint16_t data_format; + uint16_t num_channels; + uint16_t slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; +} __packed; + +struct afe_param_cdc_slimbus_slave_cfg { + uint32_t minor_version; + uint32_t device_enum_addr_lsw; + uint32_t device_enum_addr_msw; + uint16_t tx_slave_port_offset; + uint16_t rx_slave_port_offset; +} __packed; + +struct afe_param_cdc_reg_cfg { + uint32_t minor_version; + uint32_t reg_logical_addr; + uint32_t reg_field_type; + uint32_t reg_field_bit_mask; + uint16_t reg_bit_width; + uint16_t reg_offset_scale; +} __packed; + +#define AFE_API_VERSION_CDC_REG_PAGE_CFG 1 + +enum { + AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_0 = 0, + AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_1, + AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_2, + AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_3, +}; + +struct afe_param_cdc_reg_page_cfg { + uint32_t minor_version; + uint32_t enable; + uint32_t proc_id; +} __packed; + +struct afe_param_cdc_reg_cfg_data { + uint32_t num_registers; + struct afe_param_cdc_reg_cfg *reg_data; +} __packed; + +struct afe_svc_cmd_set_param_v1 { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* The total size of the payload, including param_hdr_v3 */ + uint32_t payload_size; + + /* The memory mapping header to be used when sending outband */ + struct mem_mapping_hdr mem_hdr; + + /* The parameter data to be filled when sent inband */ + u32 param_data[0]; +} __packed; + +struct afe_svc_cmd_set_param_v2 { + /* APR Header */ + struct apr_hdr apr_hdr; + + /* The memory mapping header to be used when sending outband */ + struct mem_mapping_hdr mem_hdr; + + /* The total size of the payload, including param_hdr_v3 */ + u32 payload_size; + + /* The parameter data to be filled when sent inband */ + u32 param_data[0]; +} __packed; + +struct afe_param_hw_mad_ctrl { + uint32_t minor_version; + uint16_t mad_type; + uint16_t mad_enable; +} __packed; + +struct afe_port_cmd_set_aanc_acdb_table { + struct apr_hdr hdr; + struct mem_mapping_hdr mem_hdr; +} __packed; + +/* Dolby DAP topology */ +#define DOLBY_ADM_COPP_TOPOLOGY_ID 0x0001033B +#define DS2_ADM_COPP_TOPOLOGY_ID 0x1301033B + +/* RMS value from DSP */ +#define RMS_MODULEID_APPI_PASSTHRU 0x10009011 +#define RMS_PARAM_FIRST_SAMPLE 0x10009012 +#define RMS_PAYLOAD_LEN 4 + +/* Customized mixing in matix mixer */ +#define MTMX_MODULE_ID_DEFAULT_CHMIXER 0x00010341 +#define DEFAULT_CHMIXER_PARAM_ID_COEFF 0x00010342 +#define CUSTOM_STEREO_PAYLOAD_SIZE 9 +#define CUSTOM_STEREO_CMD_PARAM_SIZE 24 +#define CUSTOM_STEREO_NUM_OUT_CH 0x0002 +#define CUSTOM_STEREO_NUM_IN_CH 0x0002 +#define CUSTOM_STEREO_INDEX_PARAM 0x0002 +#define Q14_GAIN_ZERO_POINT_FIVE 0x2000 +#define Q14_GAIN_UNITY 0x4000 + +/* Ultrasound supported formats */ +#define US_POINT_EPOS_FORMAT_V2 0x0001272D +#define US_RAW_FORMAT_V2 0x0001272C +#define US_PROX_FORMAT_V4 0x0001273B +#define US_RAW_SYNC_FORMAT 0x0001272F +#define US_GES_SYNC_FORMAT 0x00012730 + +#define AFE_MODULE_GROUP_DEVICE 0x00010254 +#define AFE_PARAM_ID_GROUP_DEVICE_CFG 0x00010255 +#define AFE_PARAM_ID_GROUP_DEVICE_ENABLE 0x00010256 +#define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX 0x1102 +#define AFE_PARAM_ID_GROUP_DEVICE_I2S_CONFIG 0x00010286 + +/* 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; + +#define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_TX \ + (AFE_PORT_ID_SECONDARY_MI2S_TX + 0x100) +#define AFE_GROUP_DEVICE_ID_TERTIARY_MI2S_RX \ + (AFE_PORT_ID_TERTIARY_MI2S_RX + 0x100) +#define AFE_GROUP_DEVICE_ID_TERTIARY_MI2S_TX \ + (AFE_PORT_ID_TERTIARY_MI2S_TX + 0x100) +#define AFE_GROUP_DEVICE_ID_QUATERNARY_MI2S_RX \ + (AFE_PORT_ID_QUATERNARY_MI2S_RX + 0x100) +#define AFE_GROUP_DEVICE_ID_QUATERNARY_MI2S_TX \ + (AFE_PORT_ID_QUATERNARY_MI2S_TX + 0x100) + +#define AFE_API_VERSION_GROUP_DEVICE_I2S_CONFIG 0x1 + +/* Payload of the AFE_PARAM_ID_GROUP_DEVICE_I2S_CONFIG parameter ID +* used by AFE_MODULE_GROUP_DEVICE. +*/ +struct afe_param_id_group_device_i2s_cfg_v1 { + u32 minor_version; + /**< Minor version used to track group device configuration. + * @values #AFE_API_VERSION_GROUP_DEVICE_I2S_CONFIG + */ + + u16 group_id; + /**< ID for the group device. + * @values + * - #AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX + * - #AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_TX + * - #AFE_GROUP_DEVICE_ID_TERTIARY_MI2S_RX + * - #AFE_GROUP_DEVICE_ID_TERTIARY_MI2S_TX + * - #AFE_GROUP_DEVICE_ID_QUATERNARY_MI2S_RX + * - #AFE_GROUP_DEVICE_ID_QUATERNARY_MI2S_RX + */ + + u16 channel_mode; + /**< Group line channel mode + * @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 + */ + + 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 + */ + + u16 port_id[AFE_GROUP_DEVICE_NUM_PORTS]; + /**< Array of member port IDs of this group. + * @values + * - #AFE_PORT_ID_SECONDARY_MI2S_RX_1 + * - #AFE_PORT_ID_SECONDARY_MI2S_RX_2 + * - #AFE_PORT_ID_SECONDARY_MI2S_RX_3 + * - #AFE_PORT_ID_SECONDARY_MI2S_RX_4 + + * - #AFE_PORT_ID_SECONDARY_MI2S_TX_1 + * - #AFE_PORT_ID_SECONDARY_MI2S_TX_2 + * - #AFE_PORT_ID_SECONDARY_MI2S_TX_3 + * - #AFE_PORT_ID_SECONDARY_MI2S_TX_4 + + * - #AFE_PORT_ID_TERTIARY_MI2S_RX_1 + * - #AFE_PORT_ID_TERTIARY_MI2S_RX_2 + * - #AFE_PORT_ID_TERTIARY_MI2S_RX_3 + * - #AFE_PORT_ID_TERTIARY_MI2S_RX_4 + + * - #AFE_PORT_ID_TERTIARY_MI2S_TX_1 + * - #AFE_PORT_ID_TERTIARY_MI2S_TX_2 + * - #AFE_PORT_ID_TERTIARY_MI2S_TX_3 + * - #AFE_PORT_ID_TERTIARY_MI2S_TX_4 + + * - #AFE_PORT_ID_QUATERNARY_MI2S_RX_1 + * - #AFE_PORT_ID_QUATERNARY_MI2S_RX_2 + * - #AFE_PORT_ID_QUATERNARY_MI2S_RX_3 + * - #AFE_PORT_ID_QUATERNARY_MI2S_RX_4 + + * - #AFE_PORT_ID_QUATERNARY_MI2S_TX_1 + * - #AFE_PORT_ID_QUATERNARY_MI2S_TX_2 + * - #AFE_PORT_ID_QUATERNARY_MI2S_TX_3 + * - #AFE_PORT_ID_QUATERNARY_MI2S_TX_4 + * @tablebulletend + */ + + u16 bit_width; + /**< Bit width of the sample. + * @values 16, 24, (32) + */ + + u16 reserved; +} __packed; + +struct afe_param_id_group_device_enable { + u16 group_id; + u16 enable; +} __packed; + +union afe_port_group_mi2s_config { + struct afe_param_id_group_device_i2s_cfg_v1 i2s_cfg; + struct afe_param_id_group_device_enable group_enable; +} __packed; + +struct afe_i2s_port_config { + struct afe_param_id_i2s_cfg i2s_cfg; + struct afe_param_id_slot_mapping_cfg slot_mapping; +} __packed; + +/* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_ENABLE + * parameter, which enables or + * disables any module. + * The fixed size of this structure is four bytes. + */ + +struct afe_group_device_enable { + u16 group_id; + /* valid value is AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX */ + u16 enable; + /* Enables (1) or disables (0) the module. */ +} __packed; + +union afe_port_group_config { + struct afe_group_device_group_cfg group_cfg; + struct afe_group_device_enable group_enable; + struct afe_param_id_group_device_tdm_cfg tdm_cfg; +} __packed; + +/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to specify + * the timing statistics of the corresponding device interface. + * Client can periodically query for the device time statistics to help adjust + * the PLL based on the drift value. The get param command must be sent to + * AFE port ID corresponding to device interface + + * This parameter ID supports following get param commands: + * #AFE_PORT_CMD_GET_PARAM_V2 and + * #AFE_PORT_CMD_GET_PARAM_V3. + */ +#define AFE_PARAM_ID_DEV_TIMING_STATS 0x000102AD + +/* Version information used to handle future additions to AFE device + * interface timing statistics (for backward compatibility). + */ +#define AFE_API_VERSION_DEV_TIMING_STATS 0x1 + +/* Enumeration for specifying a sink(Rx) device */ +#define AFE_SINK_DEVICE 0x0 + +/* Enumeration for specifying a source(Tx) device */ +#define AFE_SOURCE_DEVICE 0x1 + +/* Enumeration for specifying the drift reference is of type AV Timer */ +#define AFE_REF_TIMER_TYPE_AVTIMER 0x0 + +/* Message payload structure for the + * AFE_PARAM_ID_DEV_TIMING_STATS parameter. + */ +struct afe_param_id_dev_timing_stats { + /* Minor version used to track the version of device interface timing + * statistics. Currently, the supported version is 1. + * @values #AFE_API_VERSION_DEV_TIMING_STATS + */ + u32 minor_version; + + /* Indicates the device interface direction as either + * source (Tx) or sink (Rx). + * @values + * #AFE_SINK_DEVICE + * #AFE_SOURCE_DEVICE + */ + u16 device_direction; + + /* Reference timer for drift accumulation and time stamp information. + * @values + * #AFE_REF_TIMER_TYPE_AVTIMER @tablebulletend + */ + u16 reference_timer; + + /* + * Flag to indicate if resync is required on the client side for + * drift correction. Flag is set to TRUE for the first get_param + * response after device interface starts. This flag value can be + * used by client to identify if device interface restart has + * happened and if any re-sync is required at their end for drift + * correction. + * @values + * 0: FALSE (Resync not required) + * 1: TRUE (Resync required) @tablebulletend + */ + u32 resync_flag; + + /* Accumulated drift value in microseconds. This value is updated + * every 100th ms. + * Positive drift value indicates AV timer is running faster than device + * Negative drift value indicates AV timer is running slower than device + * @values Any valid int32 number + */ + s32 acc_drift_value; + + /* Lower 32 bits of the 64-bit absolute timestamp of reference + * timer in microseconds. + + * This timestamp corresponds to the time when the drift values + * are accumlated for every 100th ms. + * @values Any valid uint32 number + */ + u32 ref_timer_abs_ts_lsw; + + /* Upper 32 bits of the 64-bit absolute timestamp of reference + * timer in microseconds. + * This timestamp corresponds to the time when the drift values + * are accumlated for every 100th ms. + * @values Any valid uint32 number + */ + u32 ref_timer_abs_ts_msw; +} __packed; + +struct afe_av_dev_drift_get_param_resp { + uint32_t status; + struct param_hdr_v3 pdata; + struct afe_param_id_dev_timing_stats timing_stats; +} __packed; + +/* Command for Matrix or Stream Router */ +#define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2 0x00010DCE +/* Module for AVSYNC */ +#define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC 0x00010DC6 + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the + * render window start value. This parameter is supported only for a Set + * command (not a Get command) in the Rx direction + * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). + * Render window start is a value (session time minus timestamp, or ST-TS) + * below which frames are held, and after which frames are immediately + * rendered. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1 + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the + * render window end value. This parameter is supported only for a Set + * command (not a Get command) in the Rx direction + * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value + * (session time minus timestamp) above which frames are dropped, and below + * which frames are immediately rendered. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 0x00010DD2 + +/* Generic payload of the window parameters in the + * #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module. + * This payload is supported only for a Set command + * (not a Get command) on the Rx path. + */ +struct asm_session_mtmx_strtr_param_window_v2_t { + u32 window_lsw; + /* Lower 32 bits of the render window start value. */ + + u32 window_msw; + /* Upper 32 bits of the render window start value. + + * The 64-bit number formed by window_lsw and window_msw specifies a + * signed 64-bit window value in microseconds. The sign extension is + * necessary. This value is used by the following parameter IDs: + * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2 + * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2 + * The value depends on which parameter ID is used. + * The aDSP honors the windows at a granularity of 1 ms. + */ +}; + +struct asm_session_cmd_set_mtmx_strstr_params_v2 { + uint32_t data_payload_addr_lsw; + /* Lower 32 bits of the 64-bit data payload address. */ + + uint32_t data_payload_addr_msw; + /* Upper 32 bits of the 64-bit data payload address. + * If the address is not sent (NULL), the message is in the payload. + * If the address is sent (non-NULL), the parameter data payloads + * begin at the specified address. + */ + + uint32_t mem_map_handle; + /* Unique identifier for an address. This memory map handle is returned + * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. + * values + * - NULL -- Parameter data payloads are within the message payload + * (in-band). + * - Non-NULL -- Parameter data payloads begin at the address specified + * in the data_payload_addr_lsw and data_payload_addr_msw fields + * (out-of-band). + */ + + uint32_t data_payload_size; + /* Actual size of the variable payload accompanying the message, or in + * shared memory. This field is used for parsing the parameter payload. + * values > 0 bytes + */ + + uint32_t direction; + /* Direction of the entity (matrix mixer or stream router) on which + * the parameter is to be set. + * values + * - 0 -- Rx (for Rx stream router or Rx matrix mixer) + * - 1 -- Tx (for Tx stream router or Tx matrix mixer) + */ +}; + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the + * audio client choose the rendering decision that the audio DSP should use. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD 0x00012F0D + +/* Indicates that rendering decision will be based on default rate + * (session clock based rendering, device driven). + * 1. The default session clock based rendering is inherently driven + * by the timing of the device. + * 2. After the initial decision is made (first buffer after a run + * command), subsequent data rendering decisions are made with + * respect to the rate at which the device is rendering, thus deriving + * its timing from the device. + * 3. While this decision making is simple, it has some inherent limitations + * (mentioned in the next section). + * 4. If this API is not set, the session clock based rendering will be assumed + * and this will ensure that the DSP is backward compatible. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT 0 + +/* Indicates that rendering decision will be based on local clock rate. + * 1. In the DSP loopback/client loopback use cases (frame based + * inputs), the incoming data into audio DSP is time-stamped at the + * local clock rate (STC). + * 2. This TS rate may match the incoming data rate or maybe different + * from the incoming data rate. + * 3. Regardless, the data will be time-stamped with local STC and + * therefore, the client is recommended to set this mode for these + * use cases. This method is inherently more robust to sequencing + * (AFE Start/Stop) and device switches, among other benefits. + * 4. This API will inform the DSP to compare every incoming buffer TS + * against local STC. + * 5. DSP will continue to honor render windows APIs, as before. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC 1 + +/* Structure for rendering decision parameter */ +struct asm_session_mtmx_strtr_param_render_mode_t { + /* Specifies the type of rendering decision the audio DSP should use. + * + * @values + * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT + * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC + */ + u32 flags; +} __packed; + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the + * audio client to specify the clock recovery mechanism that the audio DSP + * should use. + */ + +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD 0x00012F0E + +/* Indicates that default clock recovery will be used (no clock recovery). + * If the client wishes that no clock recovery be done, the client can + * choose this. This means that no attempt will made by the DSP to try and + * match the rates of the input and output audio. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE 0 + +/* Indicates that independent clock recovery needs to be used. + * 1. In the DSP loopback/client loopback use cases (frame based inputs), + * the client should choose the independent clock recovery option. + * 2. This basically de-couples the audio and video from knowing each others + * clock sources and lets the audio DSP independently rate match the input + * and output rates. + * 3. After drift detection, the drift correction is achieved by either pulling + * the PLLs (if applicable) or by stream to device rate matching + * (for PCM use cases) by comparing drift with respect to STC. + * 4. For passthrough use cases, since the PLL pulling is the only option, + * a best effort will be made. + * If PLL pulling is not possible / available, the rendering will be + * done without rate matching. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO 1 + +/* Payload of the #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC parameter. + */ +struct asm_session_mtmx_strtr_param_clk_rec_t { + /* Specifies the type of clock recovery that the audio DSP should + * use for rate matching. + */ + + /* @values + * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_DEFAULT + * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_INDEPENDENT + */ + u32 flags; +} __packed; + + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to + * realize smoother adjustment of audio session clock for a specified session. + * The desired audio session clock adjustment(in micro seconds) is specified + * using the command #ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2. + * Delaying/Advancing the session clock would be implemented by inserting + * interpolated/dropping audio samples in the playback path respectively. + * Also, this parameter has to be configured before the Audio Session is put + * to RUN state to avoid cold start latency/glitches in the playback. + */ + +#define ASM_SESSION_MTMX_PARAM_ADJUST_SESSION_TIME_CTL 0x00013217 + +struct asm_session_mtmx_param_adjust_session_time_ctl_t { + /* Specifies whether the module is enabled or not + * @values + * 0 -- disabled + * 1 -- enabled + */ + u32 enable; +}; + +union asm_session_mtmx_strtr_param_config { + struct asm_session_mtmx_strtr_param_window_v2_t window_param; + struct asm_session_mtmx_strtr_param_render_mode_t render_param; + struct asm_session_mtmx_strtr_param_clk_rec_t clk_rec_param; + struct asm_session_mtmx_param_adjust_session_time_ctl_t adj_time_param; +} __packed; + +struct asm_mtmx_strtr_params { + struct apr_hdr hdr; + struct asm_session_cmd_set_mtmx_strstr_params_v2 param; + struct param_hdr_v1 data; + union asm_session_mtmx_strtr_param_config config; +} __packed; + +#define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF +#define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0 + +#define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B +#define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL) + +struct asm_session_cmd_get_mtmx_strstr_params_v2 { + uint32_t data_payload_addr_lsw; + /* Lower 32 bits of the 64-bit data payload address. */ + + uint32_t data_payload_addr_msw; + /* + * Upper 32 bits of the 64-bit data payload address. + * If the address is not sent (NULL), the message is in the payload. + * If the address is sent (non-NULL), the parameter data payloads + * begin at the specified address. + */ + + uint32_t mem_map_handle; + /* + * Unique identifier for an address. This memory map handle is returned + * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command. + * values + * - NULL -- Parameter data payloads are within the message payload + * (in-band). + * - Non-NULL -- Parameter data payloads begin at the address specified + * in the data_payload_addr_lsw and data_payload_addr_msw fields + * (out-of-band). + */ + uint32_t direction; + /* + * Direction of the entity (matrix mixer or stream router) on which + * the parameter is to be set. + * values + * - 0 -- Rx (for Rx stream router or Rx matrix mixer) + * - 1 -- Tx (for Tx stream router or Tx matrix mixer) + */ + uint32_t module_id; + /* Unique module ID. */ + + uint32_t param_id; + /* Unique parameter ID. */ + + uint32_t param_max_size; +}; + +struct asm_session_mtmx_strtr_param_session_time_v3_t { + uint32_t session_time_lsw; + /* Lower 32 bits of the current session time in microseconds */ + + uint32_t session_time_msw; + /* + * Upper 32 bits of the current session time in microseconds. + * The 64-bit number formed by session_time_lsw and session_time_msw + * is treated as signed. + */ + + uint32_t absolute_time_lsw; + /* + * Lower 32 bits of the 64-bit absolute time in microseconds. + * This is the time when the sample corresponding to the + * session_time_lsw is rendered to the hardware. This absolute + * time can be slightly in the future or past. + */ + + uint32_t absolute_time_msw; + /* + * Upper 32 bits of the 64-bit absolute time in microseconds. + * This is the time when the sample corresponding to the + * session_time_msw is rendered to hardware. This absolute + * time can be slightly in the future or past. The 64-bit number + * formed by absolute_time_lsw and absolute_time_msw is treated as + * unsigned. + */ + + uint32_t time_stamp_lsw; + /* Lower 32 bits of the last processed timestamp in microseconds */ + + uint32_t time_stamp_msw; + /* + * Upper 32 bits of the last processed timestamp in microseconds. + * The 64-bit number formed by time_stamp_lsw and time_stamp_lsw + * is treated as unsigned. + */ + + uint32_t flags; + /* + * Keeps track of any additional flags needed. + * @values{for bit 31} + * - 0 -- Uninitialized/invalid + * - 1 -- Valid + * All other bits are reserved; clients must set them to zero. + */ +}; + +union asm_session_mtmx_strtr_data_type { + struct asm_session_mtmx_strtr_param_session_time_v3_t session_time; +}; + +struct asm_mtmx_strtr_get_params { + struct apr_hdr hdr; + struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info; +} __packed; + +struct asm_mtmx_strtr_get_params_cmdrsp { + uint32_t err_code; + struct param_hdr_v1 param_info; + union asm_session_mtmx_strtr_data_type param_data; +} __packed; + +#define AUDPROC_MODULE_ID_RESAMPLER 0x00010719 + +enum { + LEGACY_PCM = 0, + COMPRESSED_PASSTHROUGH, + COMPRESSED_PASSTHROUGH_CONVERT, + COMPRESSED_PASSTHROUGH_DSD, + LISTEN, + COMPRESSED_PASSTHROUGH_GEN, + COMPRESSED_PASSTHROUGH_IEC61937 +}; + +#define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770 +#define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771 + +struct adm_set_compressed_device_mute { + u32 mute_on; +} __packed; + +#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E +#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F + +struct adm_set_compressed_device_latency { + u32 latency; +} __packed; + +#define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6 +#define VOICEPROC_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_param_fluence_sourcetracking_t { + uint8_t vad[MAX_SECTORS]; + uint16_t doa_speech; + uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS]; + uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS]; +} __packed; + +#define AUDPROC_MODULE_ID_AUDIOSPHERE 0x00010916 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE 0x00010917 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH 0x00010918 +#define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE 0x00010919 + +#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT 0x0001091A +#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT 0x0001091B +#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT 0x0001091C +#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT 0x0001091D + +#define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO 0x0001091E + +#define AUDPROC_MODULE_ID_VOICE_TX_SECNS 0x10027059 +#define AUDPROC_PARAM_IDX_SEC_PRIMARY_MIC_CH 0x10014444 + +struct admx_sec_primary_mic_ch { + uint16_t version; + uint16_t reserved; + uint16_t sec_primary_mic_ch; + uint16_t reserved1; +} __packed; + +/* +* ID of the DTMF Detection module. +*/ +#define AUDPROC_MODULE_ID_DTMF_DETECTION 0x00010940 + +#endif /*_APR_AUDIO_V2_H_ */ diff --git a/include/sound/apr_audio.h b/include/sound/apr_audio.h new file mode 100644 index 000000000000..4e6e2b8405ce --- /dev/null +++ b/include/sound/apr_audio.h @@ -0,0 +1,1929 @@ +/* + * + * Copyright (c) 2010-2013, 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_H_ +#define _APR_AUDIO_H_ + +/* ASM opcodes without APR payloads*/ +#include <linux/qdsp6v2/apr.h> + +/* + * Audio Front End (AFE) + */ + +/* Port ID. Update afe_get_port_index when a new port is added here. */ +#define PRIMARY_I2S_RX 0 /* index = 0 */ +#define PRIMARY_I2S_TX 1 /* index = 1 */ +#define PCM_RX 2 /* index = 2 */ +#define PCM_TX 3 /* index = 3 */ +#define SECONDARY_I2S_RX 4 /* index = 4 */ +#define SECONDARY_I2S_TX 5 /* index = 5 */ +#define MI2S_RX 6 /* index = 6 */ +#define MI2S_TX 7 /* index = 7 */ +#define HDMI_RX 8 /* index = 8 */ +#define RSVD_2 9 /* index = 9 */ +#define RSVD_3 10 /* index = 10 */ +#define DIGI_MIC_TX 11 /* index = 11 */ +#define VOICE_RECORD_RX 0x8003 /* index = 12 */ +#define VOICE_RECORD_TX 0x8004 /* index = 13 */ +#define VOICE_PLAYBACK_TX 0x8005 /* index = 14 */ + +/* Slimbus Multi channel port id pool */ +#define SLIMBUS_0_RX 0x4000 /* index = 15 */ +#define SLIMBUS_0_TX 0x4001 /* index = 16 */ +#define SLIMBUS_1_RX 0x4002 /* index = 17 */ +#define SLIMBUS_1_TX 0x4003 /* index = 18 */ +#define SLIMBUS_2_RX 0x4004 +#define SLIMBUS_2_TX 0x4005 +#define SLIMBUS_3_RX 0x4006 +#define SLIMBUS_3_TX 0x4007 +#define SLIMBUS_4_RX 0x4008 +#define SLIMBUS_4_TX 0x4009 /* index = 24 */ + +#define INT_BT_SCO_RX 0x3000 /* index = 25 */ +#define INT_BT_SCO_TX 0x3001 /* index = 26 */ +#define INT_BT_A2DP_RX 0x3002 /* index = 27 */ +#define INT_FM_RX 0x3004 /* index = 28 */ +#define INT_FM_TX 0x3005 /* index = 29 */ +#define RT_PROXY_PORT_001_RX 0x2000 /* index = 30 */ +#define RT_PROXY_PORT_001_TX 0x2001 /* index = 31 */ +#define SECONDARY_PCM_RX 12 /* index = 32 */ +#define SECONDARY_PCM_TX 13 /* index = 33 */ +#define PSEUDOPORT_01 0x8001 /* index =34 */ + +#define AFE_PORT_INVALID 0xFFFF +#define SLIMBUS_EXTPROC_RX 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 + +struct afe_port_start_command { + struct apr_hdr hdr; + u16 port_id; + u16 gain; /* Q13 */ + u32 sample_rate; /* 8 , 16, 48khz */ +} __attribute__ ((packed)); + +#define AFE_PORT_CMD_STOP 0x000100cb +struct afe_port_stop_command { + struct apr_hdr hdr; + u16 port_id; + u16 reserved; +} __attribute__ ((packed)); + +#define AFE_PORT_CMD_APPLY_GAIN 0x000100cc +struct afe_port_gain_command { + struct apr_hdr hdr; + u16 port_id; + u16 gain;/* Q13 */ +} __attribute__ ((packed)); + +#define AFE_PORT_CMD_SIDETONE_CTL 0x000100cd +struct afe_port_sidetone_command { + struct apr_hdr hdr; + u16 rx_port_id; /* Primary i2s tx = 1 */ + /* PCM tx = 3 */ + /* Secondary i2s tx = 5 */ + /* Mi2s tx = 7 */ + /* Digital mic tx = 11 */ + u16 tx_port_id; /* Primary i2s rx = 0 */ + /* PCM rx = 2 */ + /* Secondary i2s rx = 4 */ + /* Mi2S rx = 6 */ + /* HDMI rx = 8 */ + u16 gain; /* Q13 */ + u16 enable; /* 1 = enable, 0 = disable */ +} __attribute__ ((packed)); + +#define AFE_PORT_CMD_LOOPBACK 0x000100ce +struct afe_loopback_command { + struct apr_hdr hdr; + u16 tx_port_id; /* Primary i2s rx = 0 */ + /* PCM rx = 2 */ + /* Secondary i2s rx = 4 */ + /* Mi2S rx = 6 */ + /* HDMI rx = 8 */ + u16 rx_port_id; /* Primary i2s tx = 1 */ + /* PCM tx = 3 */ + /* Secondary i2s tx = 5 */ + /* Mi2s tx = 7 */ + /* Digital mic tx = 11 */ + u16 mode; /* Default -1, DSP will conver + the tx to rx format */ + u16 enable; /* 1 = enable, 0 = disable */ +} __attribute__ ((packed)); + +#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, */ +} __attribute__ ((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; +} __attribute__ ((packed)); + +#define AFE_CMD_GET_ACTIVE_PORTS 0x000100d1 + + +#define AFE_CMD_GET_ACTIVE_HANDLES_FOR_PORT 0x000100d2 +struct afe_get_active_handles_command { + struct apr_hdr hdr; + u16 port_id; + u16 reserved; +} __attribute__ ((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; + +#define AFE_PCM_CFG_MODE_PCM 0x0 +#define AFE_PCM_CFG_MODE_AUX 0x1 +#define AFE_PCM_CFG_SYNC_EXT 0x0 +#define AFE_PCM_CFG_SYNC_INT 0x1 +#define AFE_PCM_CFG_FRM_8BPF 0x0 +#define AFE_PCM_CFG_FRM_16BPF 0x1 +#define AFE_PCM_CFG_FRM_32BPF 0x2 +#define AFE_PCM_CFG_FRM_64BPF 0x3 +#define AFE_PCM_CFG_FRM_128BPF 0x4 +#define AFE_PCM_CFG_FRM_256BPF 0x5 +#define AFE_PCM_CFG_QUANT_ALAW_NOPAD 0x0 +#define AFE_PCM_CFG_QUANT_MULAW_NOPAD 0x1 +#define AFE_PCM_CFG_QUANT_LINEAR_NOPAD 0x2 +#define AFE_PCM_CFG_QUANT_ALAW_PAD 0x3 +#define AFE_PCM_CFG_QUANT_MULAW_PAD 0x4 +#define AFE_PCM_CFG_QUANT_LINEAR_PAD 0x5 +#define AFE_PCM_CFG_CDATAOE_MASTER 0x0 +#define AFE_PCM_CFG_CDATAOE_SHARE 0x1 + +struct afe_port_pcm_cfg { + u16 mode; /* PCM (short sync) = 0, AUXPCM (long sync) = 1 */ + u16 sync; /* external = 0 , internal = 1 */ + u16 frame; /* 8 bpf = 0 */ + /* 16 bpf = 1 */ + /* 32 bpf = 2 */ + /* 64 bpf = 3 */ + /* 128 bpf = 4 */ + /* 256 bpf = 5 */ + u16 quant; + u16 slot; /* Slot for PCM stream , 0 - 31 */ + u16 data; /* 0, PCM block is the only master */ + /* 1, PCM block is shares to driver data out signal */ + /* other master */ + u16 reserved; +} __attribute__ ((packed)); + +enum { + AFE_I2S_SD0 = 1, + AFE_I2S_SD1, + AFE_I2S_SD2, + AFE_I2S_SD3, + AFE_I2S_QUAD01, + AFE_I2S_QUAD23, + AFE_I2S_6CHS, + AFE_I2S_8CHS, +}; + +#define AFE_MI2S_MONO 0 +#define AFE_MI2S_STEREO 3 +#define AFE_MI2S_4CHANNELS 4 +#define AFE_MI2S_6CHANNELS 6 +#define AFE_MI2S_8CHANNELS 8 + +struct afe_port_mi2s_cfg { + u16 bitwidth; /* 16,24,32 */ + u16 line; /* Called ChannelMode in documentation */ + /* i2s_sd0 = 1 */ + /* i2s_sd1 = 2 */ + /* i2s_sd2 = 3 */ + /* i2s_sd3 = 4 */ + /* i2s_quad01 = 5 */ + /* i2s_quad23 = 6 */ + /* i2s_6chs = 7 */ + /* i2s_8chs = 8 */ + u16 channel; /* Called MonoStereo in documentation */ + /* i2s mono = 0 */ + /* i2s mono right = 1 */ + /* i2s mono left = 2 */ + /* i2s stereo = 3 */ + u16 ws; /* 0, word select signal from external source */ + /* 1, word select signal from internal source */ + u16 format; /* don't touch this field if it is not for */ + /* AFE_PORT_CMD_I2S_CONFIG opcode */ +} __attribute__ ((packed)); + +struct afe_port_hdmi_cfg { + u16 bitwidth; /* 16,24,32 */ + u16 channel_mode; /* HDMI Stereo = 0 */ + /* HDMI_3Point1 (4-ch) = 1 */ + /* HDMI_5Point1 (6-ch) = 2 */ + /* HDMI_6Point1 (8-ch) = 3 */ + u16 data_type; /* HDMI_Linear = 0 */ + /* HDMI_non_Linear = 1 */ +} __attribute__ ((packed)); + + +struct afe_port_hdmi_multi_ch_cfg { + u16 data_type; /* HDMI_Linear = 0 */ + /* HDMI_non_Linear = 1 */ + u16 channel_allocation; /* The default is 0 (Stereo) */ + u16 reserved; /* must be set to 0 */ +} __packed; + + +/* Slimbus Device Ids */ +#define AFE_SLIMBUS_DEVICE_1 0x0 +#define AFE_SLIMBUS_DEVICE_2 0x1 +#define AFE_PORT_MAX_AUDIO_CHAN_CNT 16 + +struct afe_port_slimbus_cfg { + u16 slimbus_dev_id; /* SLIMBUS Device id.*/ + + u16 slave_dev_pgd_la; /* Slave ported generic device + * logical address. + */ + u16 slave_dev_intfdev_la; /* Slave interface device logical + * address. + */ + u16 bit_width; /** bit width of the samples, 16, 24.*/ + + u16 data_format; /** data format.*/ + + u16 num_channels; /** Number of channels.*/ + + /** Slave port mapping for respective channels.*/ + u16 slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT]; + + u16 reserved; +} __packed; + +struct afe_port_slimbus_sch_cfg { + u16 slimbus_dev_id; /* SLIMBUS Device id.*/ + u16 bit_width; /** bit width of the samples, 16, 24.*/ + u16 data_format; /** data format.*/ + u16 num_channels; /** Number of channels.*/ + u16 reserved; + /** Slave channel mapping for respective channels.*/ + u8 slave_ch_mapping[8]; +} __packed; + +struct afe_port_rtproxy_cfg { + u16 bitwidth; /* 16,24,32 */ + u16 interleaved; /* interleaved = 1 */ + /* Noninterleaved = 0 */ + u16 frame_sz; /* 5ms buffers = 160bytes */ + u16 jitter; /* 10ms of jitter = 320 */ + u16 lw_mark; /* Low watermark in bytes for triggering event*/ + u16 hw_mark; /* High watermark bytes for triggering event*/ + u16 rsvd; + int num_ch; /* 1 to 8 */ +} __packed; + +struct afe_port_pseudo_cfg { + u16 bit_width; + u16 num_channels; + u16 data_format; + u16 timing_mode; + u16 reserved; +} __packed; + +#define AFE_PORT_AUDIO_IF_CONFIG 0x000100d3 +#define AFE_PORT_AUDIO_SLIM_SCH_CONFIG 0x000100e4 +#define AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG 0x000100D9 +#define AFE_PORT_CMD_I2S_CONFIG 0x000100E7 + +union afe_port_config { + struct afe_port_pcm_cfg pcm; + struct afe_port_mi2s_cfg mi2s; + struct afe_port_hdmi_cfg hdmi; + struct afe_port_hdmi_multi_ch_cfg hdmi_multi_ch; + struct afe_port_slimbus_cfg slimbus; + struct afe_port_slimbus_sch_cfg slim_sch; + struct afe_port_rtproxy_cfg rtproxy; + struct afe_port_pseudo_cfg pseudo; +} __attribute__((packed)); + +struct afe_audioif_config_command { + struct apr_hdr hdr; + u16 port_id; + union afe_port_config port; +} __attribute__ ((packed)); + +#define AFE_TEST_CODEC_LOOPBACK_CTL 0x000100d5 +struct afe_codec_loopback_command { + u16 port_inf; /* Primary i2s = 0 */ + /* PCM = 2 */ + /* Secondary i2s = 4 */ + /* Mi2s = 6 */ + u16 enable; /* 0, disable. 1, enable */ +} __attribute__ ((packed)); + + +#define AFE_PARAM_ID_SIDETONE_GAIN 0x00010300 +struct afe_param_sidetone_gain { + u16 gain; + u16 reserved; +} __attribute__ ((packed)); + +#define AFE_PARAM_ID_SAMPLING_RATE 0x00010301 +struct afe_param_sampling_rate { + u32 sampling_rate; +} __attribute__ ((packed)); + + +#define AFE_PARAM_ID_CHANNELS 0x00010302 +struct afe_param_channels { + u16 channels; + u16 reserved; +} __attribute__ ((packed)); + + +#define AFE_PARAM_ID_LOOPBACK_GAIN 0x00010303 +struct afe_param_loopback_gain { + u16 gain; + u16 reserved; +} __attribute__ ((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. +*/ +enum { + /* Regular loopback from source to destination port */ + LB_MODE_DEFAULT = 1, + /* Sidetone feed from Tx source to Rx destination port */ + LB_MODE_SIDETONE, + /* Echo canceller reference, voice + audio + DTMF */ + LB_MODE_EC_REF_VOICE_AUDIO, + /* Echo canceller reference, voice alone */ + LB_MODE_EC_REF_VOICE +}; + +#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B +#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1 +struct afe_param_loopback_cfg { + /* Minor version used for tracking the version of the configuration + * interface. + */ + uint32_t loopback_cfg_minor_version; + + /* Destination Port Id. */ + uint16_t dst_port_id; + + /* 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 + */ + uint16_t routing_mode; + + /* Specifies whether to enable (1) or disable (0) an AFE loopback. */ + uint16_t enable; + + /* Reserved for 32-bit alignment. This field must be set to 0. */ + uint16_t reserved; +} __packed; + +#define AFE_MODULE_ID_PORT_INFO 0x00010200 +/* Module ID for the loopback-related parameters. */ +#define AFE_MODULE_LOOPBACK 0x00010205 +struct afe_param_payload_base { + u32 module_id; + u32 param_id; + u16 param_size; + u16 reserved; +} __packed; + +struct afe_param_payload { + struct afe_param_payload_base base; + union { + struct afe_param_sidetone_gain sidetone_gain; + struct afe_param_sampling_rate sampling_rate; + struct afe_param_channels channels; + struct afe_param_loopback_gain loopback_gain; + struct afe_param_loopback_cfg loopback_cfg; + } __attribute__((packed)) param; +} __attribute__ ((packed)); + +#define AFE_PORT_CMD_SET_PARAM 0x000100dc + +struct afe_port_cmd_set_param { + struct apr_hdr hdr; + u16 port_id; + u16 payload_size; + u32 payload_address; + struct afe_param_payload payload; +} __attribute__ ((packed)); + +struct afe_port_cmd_set_param_no_payload { + struct apr_hdr hdr; + u16 port_id; + u16 payload_size; + u32 payload_address; +} __packed; + +#define AFE_EVENT_GET_ACTIVE_PORTS 0x00010100 +struct afe_get_active_ports_rsp { + u16 num_ports; + u16 port_id; +} __attribute__ ((packed)); + + +#define AFE_EVENT_GET_ACTIVE_HANDLES 0x00010102 +struct afe_get_active_handles_rsp { + u16 port_id; + u16 num_handles; + u16 mode; /* 0, voice rx */ + /* 1, voice tx */ + /* 2, audio rx */ + /* 3, audio tx */ + u16 handle; +} __attribute__ ((packed)); + +#define AFE_SERVICE_CMD_MEMORY_MAP 0x000100DE +struct afe_cmd_memory_map { + struct apr_hdr hdr; + u32 phy_addr; + u32 mem_sz; + u16 mem_id; + u16 rsvd; +} __packed; + +#define AFE_SERVICE_CMD_MEMORY_UNMAP 0x000100DF +struct afe_cmd_memory_unmap { + struct apr_hdr hdr; + u32 phy_addr; +} __packed; + +#define AFE_SERVICE_CMD_REG_RTPORT 0x000100E0 +struct afe_cmd_reg_rtport { + struct apr_hdr hdr; + u16 port_id; + u16 rsvd; +} __packed; + +#define AFE_SERVICE_CMD_UNREG_RTPORT 0x000100E1 +struct afe_cmd_unreg_rtport { + struct apr_hdr hdr; + u16 port_id; + u16 rsvd; +} __packed; + +#define AFE_SERVICE_CMD_RTPORT_WR 0x000100E2 +struct afe_cmd_rtport_wr { + struct apr_hdr hdr; + u16 port_id; + u16 rsvd; + u32 buf_addr; + u32 bytes_avail; +} __packed; + +#define AFE_SERVICE_CMD_RTPORT_RD 0x000100E3 +struct afe_cmd_rtport_rd { + struct apr_hdr hdr; + u16 port_id; + u16 rsvd; + u32 buf_addr; + u32 bytes_avail; +} __packed; + +#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105 + +#define ADM_MAX_COPPS 5 + +#define ADM_SERVICE_CMD_GET_COPP_HANDLES 0x00010300 +struct adm_get_copp_handles_command { + struct apr_hdr hdr; +} __attribute__ ((packed)); + +#define ADM_CMD_MATRIX_MAP_ROUTINGS 0x00010301 +struct adm_routings_session { + u16 id; + u16 num_copps; + u16 copp_id[ADM_MAX_COPPS+1]; /*Padding if numCopps is odd */ +} __packed; + +struct adm_routings_command { + struct apr_hdr hdr; + u32 path; /* 0 = Rx, 1 Tx */ + u32 num_sessions; + struct adm_routings_session session[8]; +} __attribute__ ((packed)); + + +#define ADM_CMD_MATRIX_RAMP_GAINS 0x00010302 +struct adm_ramp_gain { + struct apr_hdr hdr; + u16 session_id; + u16 copp_id; + u16 initial_gain; + u16 gain_increment; + u16 ramp_duration; + u16 reserved; +} __attribute__ ((packed)); + +struct adm_ramp_gains_command { + struct apr_hdr hdr; + u32 id; + u32 num_gains; + struct adm_ramp_gain gains[ADM_MAX_COPPS]; +} __attribute__ ((packed)); + + +#define ADM_CMD_COPP_OPEN 0x00010304 +struct adm_copp_open_command { + struct apr_hdr hdr; + u16 flags; + u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */ + u16 endpoint_id1; + u16 endpoint_id2; + u32 topology_id; + u16 channel_config; + u16 reserved; + u32 rate; +} __attribute__ ((packed)); + +#define ADM_CMD_COPP_CLOSE 0x00010305 + +#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN 0x00010310 +#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN_V3 0x00010333 +struct adm_multi_ch_copp_open_command { + struct apr_hdr hdr; + u16 flags; + u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */ + u16 endpoint_id1; + u16 endpoint_id2; + u32 topology_id; + u16 channel_config; + u16 reserved; + u32 rate; + u8 dev_channel_mapping[8]; +} __packed; + +struct adm_multi_channel_copp_open_v3 { + struct apr_hdr hdr; + u16 flags; + u16 mode; + u16 endpoint_id1; + u16 endpoint_id2; + u32 topology_id; + u16 channel_config; + u16 bit_width; + u32 rate; + u8 dev_channel_mapping[8]; +}; +#define ADM_CMD_MEMORY_MAP 0x00010C30 +struct adm_cmd_memory_map{ + struct apr_hdr hdr; + u32 buf_add; + u32 buf_size; + u16 mempool_id; + u16 reserved; +} __attribute__((packed)); + +#define ADM_CMD_MEMORY_UNMAP 0x00010C31 +struct adm_cmd_memory_unmap{ + struct apr_hdr hdr; + u32 buf_add; +} __attribute__((packed)); + +#define ADM_CMD_MEMORY_MAP_REGIONS 0x00010C47 +struct adm_memory_map_regions{ + u32 phys; + u32 buf_size; +} __attribute__((packed)); + +struct adm_cmd_memory_map_regions{ + struct apr_hdr hdr; + u16 mempool_id; + u16 nregions; +} __attribute__((packed)); + +#define ADM_CMD_MEMORY_UNMAP_REGIONS 0x00010C48 +struct adm_memory_unmap_regions{ + u32 phys; +} __attribute__((packed)); + +struct adm_cmd_memory_unmap_regions{ + struct apr_hdr hdr; + u16 nregions; + u16 reserved; +} __attribute__((packed)); + +#define DEFAULT_COPP_TOPOLOGY 0x00010be3 +#define DEFAULT_POPP_TOPOLOGY 0x00010be4 +#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71 +#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72 +#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75 + +#define LOWLATENCY_POPP_TOPOLOGY 0x00010C68 +#define LOWLATENCY_COPP_TOPOLOGY 0x00010312 +#define PCM_BITS_PER_SAMPLE 16 + +#define ASM_OPEN_WRITE_PERF_MODE_BIT (1<<28) +#define ASM_OPEN_READ_PERF_MODE_BIT (1<<29) +#define ADM_MULTI_CH_COPP_OPEN_PERF_MODE_BIT (1<<13) + + +#define ASM_MAX_EQ_BANDS 12 + +struct asm_eq_band { + u32 band_idx; /* The band index, 0 .. 11 */ + u32 filter_type; /* Filter band type */ + u32 center_freq_hz; /* Filter band center frequency */ + u32 filter_gain; /* Filter band initial gain (dB) */ + /* Range is +12 dB to -12 dB with 1dB increments. */ + u32 q_factor; +} __attribute__ ((packed)); + +struct asm_equalizer_params { + u32 enable; + u32 num_bands; + struct asm_eq_band eq_bands[ASM_MAX_EQ_BANDS]; +} __attribute__ ((packed)); + +struct asm_master_gain_params { + u16 master_gain; + u16 padding; +} __attribute__ ((packed)); + +struct asm_lrchannel_gain_params { + u16 left_gain; + u16 right_gain; +} __attribute__ ((packed)); + +struct asm_mute_params { + u32 muteflag; +} __attribute__ ((packed)); + +struct asm_softvolume_params { + u32 period; + u32 step; + u32 rampingcurve; +} __attribute__ ((packed)); + +struct asm_softpause_params { + u32 enable; + u32 period; + u32 step; + u32 rampingcurve; +} __packed; + +struct asm_pp_param_data_hdr { + u32 module_id; + u32 param_id; + u16 param_size; + u16 reserved; +} __attribute__ ((packed)); + +struct asm_pp_params_command { + struct apr_hdr hdr; + u32 *payload; + u32 payload_size; + struct asm_pp_param_data_hdr params; +} __attribute__ ((packed)); + +#define EQUALIZER_MODULE_ID 0x00010c27 +#define EQUALIZER_PARAM_ID 0x00010c28 + +#define VOLUME_CONTROL_MODULE_ID 0x00010bfe +#define MASTER_GAIN_PARAM_ID 0x00010bff +#define L_R_CHANNEL_GAIN_PARAM_ID 0x00010c00 +#define MUTE_CONFIG_PARAM_ID 0x00010c01 +#define SOFT_PAUSE_PARAM_ID 0x00010D6A +#define SOFT_VOLUME_PARAM_ID 0x00010C29 + +#define IIR_FILTER_ENABLE_PARAM_ID 0x00010c03 +#define IIR_FILTER_PREGAIN_PARAM_ID 0x00010c04 +#define IIR_FILTER_CONFIG_PARAM_ID 0x00010c05 + +#define MBADRC_MODULE_ID 0x00010c06 +#define MBADRC_ENABLE_PARAM_ID 0x00010c07 +#define MBADRC_CONFIG_PARAM_ID 0x00010c08 + + +#define ADM_CMD_SET_PARAMS 0x00010306 +#define ADM_CMD_GET_PARAMS 0x0001030B +#define ADM_CMDRSP_GET_PARAMS 0x0001030C +struct adm_set_params_command { + struct apr_hdr hdr; + u32 payload; + u32 payload_size; +} __attribute__ ((packed)); + + +#define ADM_CMD_TAP_COPP_PCM 0x00010307 +struct adm_tap_copp_pcm_command { + struct apr_hdr hdr; +} __attribute__ ((packed)); + + +/* QDSP6 to Client messages +*/ +#define ADM_SERVICE_CMDRSP_GET_COPP_HANDLES 0x00010308 +struct adm_get_copp_handles_respond { + struct apr_hdr hdr; + u32 handles; + u32 copp_id; +} __attribute__ ((packed)); + +#define ADM_CMDRSP_COPP_OPEN 0x0001030A +struct adm_copp_open_respond { + u32 status; + u16 copp_id; + u16 reserved; +} __attribute__ ((packed)); + +#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN 0x00010311 +#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN_V3 0x00010334 + + +#define ASM_STREAM_PRIORITY_NORMAL 0 +#define ASM_STREAM_PRIORITY_LOW 1 +#define ASM_STREAM_PRIORITY_HIGH 2 +#define ASM_STREAM_PRIORITY_RESERVED 3 + +#define ASM_END_POINT_DEVICE_MATRIX 0 +#define ASM_END_POINT_STREAM 1 + +#define AAC_ENC_MODE_AAC_LC 0x02 +#define AAC_ENC_MODE_AAC_P 0x05 +#define AAC_ENC_MODE_EAAC_P 0x1D + +#define ASM_STREAM_CMD_CLOSE 0x00010BCD +#define ASM_STREAM_CMD_FLUSH 0x00010BCE +#define ASM_STREAM_CMD_SET_PP_PARAMS 0x00010BCF +#define ASM_STREAM_CMD_GET_PP_PARAMS 0x00010BD0 +#define ASM_STREAM_CMDRSP_GET_PP_PARAMS 0x00010BD1 +#define ASM_SESSION_CMD_PAUSE 0x00010BD3 +#define ASM_SESSION_CMD_GET_SESSION_TIME 0x00010BD4 +#define ASM_DATA_CMD_EOS 0x00010BDB +#define ASM_DATA_EVENT_EOS 0x00010BDD + +#define ASM_SERVICE_CMD_GET_STREAM_HANDLES 0x00010C0B +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09 + +#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17 +#define ASM_SESSION_EVENT_TX_OVERFLOW 0x00010C18 +#define ASM_SERVICE_CMD_GET_WALLCLOCK_TIME 0x00010C19 +#define ASM_DATA_CMDRSP_EOS 0x00010C1C + +/* ASM Data structures */ + +/* common declarations */ +struct asm_pcm_cfg { + u16 ch_cfg; + u16 bits_per_sample; + u32 sample_rate; + u16 is_signed; + u16 interleaved; +}; + +#define PCM_CHANNEL_NULL 0 + +/* Front left channel. */ +#define PCM_CHANNEL_FL 1 + +/* Front right channel. */ +#define PCM_CHANNEL_FR 2 + +/* Front center channel. */ +#define PCM_CHANNEL_FC 3 + +/* Left surround channel.*/ +#define PCM_CHANNEL_LS 4 + +/* Right surround channel.*/ +#define PCM_CHANNEL_RS 5 + +/* Low frequency effect channel. */ +#define PCM_CHANNEL_LFE 6 + +/* Center surround channel; Rear center channel. */ +#define PCM_CHANNEL_CS 7 + +/* Left back channel; Rear left channel. */ +#define PCM_CHANNEL_LB 8 + +/* Right back channel; Rear right channel. */ +#define PCM_CHANNEL_RB 9 + +/* Top surround channel. */ +#define PCM_CHANNEL_TS 10 + +/* Center vertical height channel.*/ +#define PCM_CHANNEL_CVH 11 + +/* Mono surround channel.*/ +#define PCM_CHANNEL_MS 12 + +/* Front left of center. */ +#define PCM_CHANNEL_FLC 13 + +/* Front right of center. */ +#define PCM_CHANNEL_FRC 14 + +/* Rear left of center. */ +#define PCM_CHANNEL_RLC 15 + +/* Rear right of center. */ +#define PCM_CHANNEL_RRC 16 + +#define PCM_FORMAT_MAX_NUM_CHANNEL 8 + +/* Maximum number of channels supported + * in ASM_ENCDEC_DEC_CHAN_MAP command + */ +#define MAX_CHAN_MAP_CHANNELS 16 +/* + * Multiple-channel PCM decoder format block structure used in the + * #ASM_STREAM_CMD_OPEN_WRITE command. + * The data must be in little-endian format. + */ +struct asm_multi_channel_pcm_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 is_interleaved; /* + * Flag that indicates whether the channels are + * de-interleaved (0) or interleaved (1). + * Interleaved format means corresponding + * samples from the left and right channels are + * interleaved within the buffer. + * De-interleaved format means samples from + * each channel are contiguous in the buffer. + * The samples from one channel immediately + * follow those of the previous channel. + */ + + u8 channel_mapping[8]; /* + * Supported values: + * PCM_CHANNEL_NULL, PCM_CHANNEL_FL, + * PCM_CHANNEL_FR, PCM_CHANNEL_FC, + * PCM_CHANNEL_LS, PCM_CHANNEL_RS, + * PCM_CHANNEL_LFE, PCM_CHANNEL_CS, + * PCM_CHANNEL_LB, PCM_CHANNEL_RB, + * PCM_CHANNEL_TS, PCM_CHANNEL_CVH, + * PCM_CHANNEL_MS, PCM_CHANNEL_FLC, + * PCM_CHANNEL_FRC, PCM_CHANNEL_RLC, + * PCM_CHANNEL_RRC. + * Channel[i] mapping describes channel I. Each + * element i of the array describes channel I + * inside the buffer where I < num_channels. + * An unused channel is set to zero. + */ +}; +struct asm_dts_enc_cfg { + uint32_t sample_rate; + /* + * Samples at which input is to be encoded. + * Supported values: + * 44100 -- encode at 44.1 Khz + * 48000 -- encode at 48 Khz + */ + + uint32_t num_channels; + /* + * Number of channels for multi-channel encoding. + * Supported values: 1 to 6 + */ + + uint8_t channel_mapping[6]; + /* + * Channel array of size 16. Channel[i] mapping describes channel I. + * Each element i of the array describes channel I inside the buffer + * where num_channels. An unused channel is set to zero. Only first + * num_channels elements are valid + + * Supported values: + * - # PCM_CHANNEL_L + * - # PCM_CHANNEL_R + * - # PCM_CHANNEL_C + * - # PCM_CHANNEL_LS + * - # PCM_CHANNEL_RS + * - # PCM_CHANNEL_LFE + */ + +}; +struct asm_adpcm_cfg { + u16 ch_cfg; + u16 bits_per_sample; + u32 sample_rate; + u32 block_size; +}; + +struct asm_yadpcm_cfg { + u16 ch_cfg; + u16 bits_per_sample; + u32 sample_rate; +}; + +struct asm_midi_cfg { + u32 nMode; +}; + +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; +}; + +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; +}; + +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; +}; + +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; +}; + +struct asm_flac_cfg { + u16 stream_info_present; + u16 min_blk_size; + u16 max_blk_size; + u16 ch_cfg; + u16 sample_size; + u16 sample_rate; + u16 md5_sum; + u32 ext_sample_rate; + u32 min_frame_size; + u32 max_frame_size; +}; + +struct asm_vorbis_cfg { + u32 ch_cfg; + u32 bit_rate; + u32 min_bit_rate; + u32 max_bit_rate; + u16 bit_depth_pcm_sample; + u16 bit_stream_format; +}; + +struct asm_aac_read_cfg { + u32 bitrate; + u32 enc_mode; + u16 format; + u16 ch_cfg; + u32 sample_rate; +}; + +struct asm_amrnb_read_cfg { + u16 mode; + u16 dtx_mode; +}; + +struct asm_amrwb_read_cfg { + u16 mode; + u16 dtx_mode; +}; + +struct asm_evrc_read_cfg { + u16 max_rate; + u16 min_rate; + u16 rate_modulation_cmd; + u16 reserved; +}; + +struct asm_qcelp13_read_cfg { + u16 max_rate; + u16 min_rate; + u16 reduced_rate_level; + u16 rate_modulation_cmd; +}; + +struct asm_sbc_read_cfg { + u32 subband; + u32 block_len; + u32 ch_mode; + u32 alloc_method; + u32 bit_rate; + u32 sample_rate; +}; + +struct asm_sbc_bitrate { + u32 bitrate; +}; + +struct asm_immed_decode { + u32 mode; +}; + +struct asm_sbr_ps { + u32 enable; +}; + +struct asm_dual_mono { + u16 sce_left; + u16 sce_right; +}; + +struct asm_dec_chan_map { + u32 num_channels; /* Number of decoder output + * 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];/* Channel array of size + * num_channels. It can grow + * till MAX_CHAN_MAP_CHANNELS. + * Channel[i] mapping + * describes channel I inside + * the decoder output buffer. + * Valid channel mapping + * values are to be present at + * the beginning of the array. + * All remaining elements of + * the array are to be filled + * with PCM_CHANNEL_NULL. + */ +}; + +struct asm_encode_cfg_blk { + u32 frames_per_buf; + u32 format_id; + u32 cfg_size; + union { + struct asm_pcm_cfg pcm; + struct asm_aac_read_cfg aac; + struct asm_amrnb_read_cfg amrnb; + struct asm_evrc_read_cfg evrc; + struct asm_qcelp13_read_cfg qcelp13; + struct asm_sbc_read_cfg sbc; + struct asm_amrwb_read_cfg amrwb; + struct asm_multi_channel_pcm_fmt_blk mpcm; + struct asm_dts_enc_cfg dts; + } __attribute__((packed)) cfg; +}; + +struct asm_frame_meta_info { + u32 offset_to_frame; + u32 frame_size; + u32 encoded_pcm_samples; + u32 msw_ts; + u32 lsw_ts; + u32 nflags; +}; + +/* Stream level commands */ +#define ASM_STREAM_CMD_OPEN_READ 0x00010BCB +#define ASM_STREAM_CMD_OPEN_READ_V2_1 0x00010DB2 +struct asm_stream_cmd_open_read { + struct apr_hdr hdr; + u32 uMode; + u32 src_endpoint; + u32 pre_proc_top; + u32 format; +} __attribute__((packed)); + +struct asm_stream_cmd_open_read_v2_1 { + struct apr_hdr hdr; + u32 uMode; + u32 src_endpoint; + u32 pre_proc_top; + u32 format; + u16 bits_per_sample; + u16 reserved; +} __packed; + +/* Supported formats */ +#define LINEAR_PCM 0x00010BE5 +#define DTMF 0x00010BE6 +#define ADPCM 0x00010BE7 +#define YADPCM 0x00010BE8 +#define MP3 0x00010BE9 +#define MPEG4_AAC 0x00010BEA +#define AMRNB_FS 0x00010BEB +#define AMRWB_FS 0x00010BEC +#define V13K_FS 0x00010BED +#define EVRC_FS 0x00010BEE +#define EVRCB_FS 0x00010BEF +#define EVRCWB_FS 0x00010BF0 +#define MIDI 0x00010BF1 +#define SBC 0x00010BF2 +#define WMA_V10PRO 0x00010BF3 +#define WMA_V9 0x00010BF4 +#define AMR_WB_PLUS 0x00010BF5 +#define AC3_DECODER 0x00010BF6 +#define EAC3_DECODER 0x00010C3C +#define DTS 0x00010D88 +#define DTS_LBR 0x00010DBB +#define MP2 0x00010DBE +#define ATRAC 0x00010D89 +#define MAT 0x00010D8A +#define G711_ALAW_FS 0x00010BF7 +#define G711_MLAW_FS 0x00010BF8 +#define G711_PCM_FS 0x00010BF9 +#define MPEG4_MULTI_AAC 0x00010D86 +#define US_POINT_EPOS_FORMAT 0x00012310 +#define US_RAW_FORMAT 0x0001127C +#define US_PROX_FORMAT 0x0001272B +#define MULTI_CHANNEL_PCM 0x00010C66 + +#define ASM_ENCDEC_SBCRATE 0x00010C13 +#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14 +#define ASM_ENCDEC_CFG_BLK 0x00010C2C + +#define ASM_ENCDEC_SBCRATE 0x00010C13 +#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14 +#define ASM_ENCDEC_CFG_BLK 0x00010C2C + +#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95 +struct asm_stream_cmd_open_read_compressed { + struct apr_hdr hdr; + u32 uMode; + u32 frame_per_buf; +} __packed; + +#define ASM_STREAM_CMD_OPEN_WRITE 0x00010BCA +#define ASM_STREAM_CMD_OPEN_WRITE_V2_1 0x00010DB1 +struct asm_stream_cmd_open_write { + struct apr_hdr hdr; + u32 uMode; + u16 sink_endpoint; + u16 stream_handle; + u32 post_proc_top; + u32 format; +} __attribute__((packed)); + +#define IEC_61937_MASK 0x00000001 +#define IEC_60958_MASK 0x00000002 + +#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84 +struct asm_stream_cmd_open_write_compressed { + struct apr_hdr hdr; + u32 flags; + u32 format; +} __packed; +#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA +struct asm_stream_cmd_open_transcode_loopback { + struct apr_hdr hdr; + uint32_t mode_flags; + /* + * All bits are reserved. Clients must set them to zero. + */ + + uint32_t src_format_id; + /* + * Specifies the media format of the input audio stream. + + * Supported values: + * - #ASM_MEDIA_FMT_LINEAR_PCM + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM + */ + + uint32_t sink_format_id; + /* + * Specifies the media format of the output stream. + + * Supported values: + * - #ASM_MEDIA_FMT_LINEAR_PCM + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM + * - #ASM_MEDIA_FMT_DTS + */ + + uint32_t 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_NONE + * - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL + */ + + uint16_t 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 zero. Otherwise, an error is returned. + */ + + uint16_t 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 zero. Otherwise, an error is returned. + */ + + uint16_t bits_per_sample; + /* + * Number of bits per sample processed by the ASM modules. + * Supported values: 16, 24 + */ + + uint16_t reserved; + /* + * This field must be set to zero. + */ +} __packed; + +/* +* ID of the DTS mix LFE channel to front channels parameter in the +* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. +* asm_dts_generic_param_t +* ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT +*/ +#define ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT 0x00010DB6 + +/* +* ID of the DTS DRC ratio parameter in the +* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. +* asm_dts_generic_param_t +* ASM_PARAM_ID_DTS_DRC_RATIO +*/ +#define ASM_PARAM_ID_DTS_DRC_RATIO 0x00010DB7 + +/* +* ID of the DTS enable dialog normalization parameter in the +* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. + +* asm_dts_generic_param_t +* ASM_PARAM_ID_DTS_ENABLE_DIALNORM +*/ +#define ASM_PARAM_ID_DTS_ENABLE_DIALNORM 0x00010DB8 + +/* +* ID of the DTS enable parse REV2AUX parameter in the +* #ASM_STREAM_CMD_SET_ENCDEC_PARAM command. +* asm_dts_generic_param_t +* ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX +*/ +#define ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX 0x00010DB9 + +struct asm_dts_generic_param { + int32_t generic_parameter; + /* + * #ASM_PARAM_ID_DTS_MIX_LFE_TO_FRONT: + * - if enabled, mixes LFE channel to front + * while downmixing (if necessary) + * - Supported values: 1-> enable, 0-> disable + * - Default: disabled + + * #ASM_PARAM_ID_DTS_DRC_RATIO: + * - percentage of DRC ratio. + * - Supported values: 0-100 + * - Default: 0, DRC is disabled. + + * #ASM_PARAM_ID_DTS_ENABLE_DIALNORM: + * - flag to enable dialog normalization post processing. + * - Supported values: 1-> enable, 0-> disable. + * - Default: enabled. + + * #ASM_PARAM_ID_DTS_ENABLE_PARSE_REV2AUX: + * - flag to enable parsing of rev2aux chunk in the bitstream. + * This chunk contains broadcast metadata. + * - Supported values: 1-> enable, 0-> disable. + * - Default: disabled. + */ +}; + +struct asm_stream_cmd_dts_dec_param { + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_dts_generic_param generic_param; +} __packed; + + +#define ASM_STREAM_CMD_OPEN_READWRITE 0x00010BCC + +struct asm_stream_cmd_open_read_write { + struct apr_hdr hdr; + u32 uMode; + u32 post_proc_top; + u32 write_format; + u32 read_format; +} __attribute__((packed)); + +#define ASM_STREAM_CMD_OPEN_LOOPBACK 0x00010D6E +struct asm_stream_cmd_open_loopback { + 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. + */ +} __packed; + +#define ADM_CMD_CONNECT_AFE_PORT 0x00010320 +#define ADM_CMD_DISCONNECT_AFE_PORT 0x00010321 + +struct adm_cmd_connect_afe_port { + struct apr_hdr hdr; + u8 mode; /*mode represent the interface is for RX or TX*/ + u8 session_id; /*ASM session ID*/ + u16 afe_port_id; +} __packed; + +#define ADM_CMD_CONNECT_AFE_PORT_V2 0x00010332 + +struct adm_cmd_connect_afe_port_v2 { + struct apr_hdr hdr; + u8 mode; /*mode represent the interface is for RX or TX*/ + u8 session_id; /*ASM session ID*/ + u16 afe_port_id; + u32 num_channels; + u32 sampling_rate; +} __packed; + +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10 +#define ASM_STREAM_CMD_GET_ENCDEC_PARAM 0x00010C11 +#define ASM_ENCDEC_CFG_BLK_ID 0x00010C2C +#define ASM_ENABLE_SBR_PS 0x00010C63 +#define ASM_CONFIGURE_DUAL_MONO 0x00010C64 +struct asm_stream_cmd_encdec_cfg_blk{ + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_encode_cfg_blk enc_blk; +} __attribute__((packed)); + +struct asm_stream_cmd_encdec_sbc_bitrate{ + struct apr_hdr hdr; + u32 param_id; + struct asm_sbc_bitrate sbc_bitrate; +} __attribute__((packed)); + +struct asm_stream_cmd_encdec_immed_decode{ + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_immed_decode dec; +} __attribute__((packed)); + +struct asm_stream_cmd_encdec_sbr{ + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_sbr_ps sbr_ps; +} __attribute__((packed)); + +struct asm_stream_cmd_encdec_dualmono { + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_dual_mono channel_map; +} __packed; + +#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG 0x00010DD8 + +/* Structure for AAC decoder stereo coefficient setting. */ + +struct asm_aac_stereo_mix_coeff_selection_param { + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + u32 aac_stereo_mix_coeff_flag; +} __packed; + +#define ASM_ENCDEC_DEC_CHAN_MAP 0x00010D82 +struct asm_stream_cmd_encdec_channelmap { + struct apr_hdr hdr; + u32 param_id; + u32 param_size; + struct asm_dec_chan_map chan_map; +} __packed; + +#define ASM_STREAM _CMD_ADJUST_SAMPLES 0x00010C0A +struct asm_stream_cmd_adjust_samples{ + struct apr_hdr hdr; + u16 nsamples; + u16 reserved; +} __attribute__((packed)); + +#define ASM_STREAM_CMD_TAP_POPP_PCM 0x00010BF9 +struct asm_stream_cmd_tap_popp_pcm{ + struct apr_hdr hdr; + u16 enable; + u16 reserved; + u32 module_id; +} __attribute__((packed)); + +/* Session Level commands */ +#define ASM_SESSION_CMD_MEMORY_MAP 0x00010C32 +struct asm_stream_cmd_memory_map{ + struct apr_hdr hdr; + u32 buf_add; + u32 buf_size; + u16 mempool_id; + u16 reserved; +} __attribute__((packed)); + +#define ASM_SESSION_CMD_MEMORY_UNMAP 0x00010C33 +struct asm_stream_cmd_memory_unmap{ + struct apr_hdr hdr; + u32 buf_add; +} __attribute__((packed)); + +#define ASM_SESSION_CMD_MEMORY_MAP_REGIONS 0x00010C45 +struct asm_memory_map_regions{ + u32 phys; + u32 buf_size; +} __attribute__((packed)); + +struct asm_stream_cmd_memory_map_regions{ + struct apr_hdr hdr; + u16 mempool_id; + u16 nregions; +} __attribute__((packed)); + +#define ASM_SESSION_CMD_MEMORY_UNMAP_REGIONS 0x00010C46 +struct asm_memory_unmap_regions{ + u32 phys; +} __attribute__((packed)); + +struct asm_stream_cmd_memory_unmap_regions{ + struct apr_hdr hdr; + u16 nregions; + u16 reserved; +} __attribute__((packed)); + +#define ASM_SESSION_CMD_RUN 0x00010BD2 +struct asm_stream_cmd_run{ + struct apr_hdr hdr; + u32 flags; + u32 msw_ts; + u32 lsw_ts; +} __attribute__((packed)); + +/* Session level events */ +#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5 +struct asm_stream_cmd_reg_rx_underflow_event{ + struct apr_hdr hdr; + u16 enable; + u16 reserved; +} __attribute__((packed)); + +#define ASM_SESSION_CMD_REGISTER_FOR_TX_OVERFLOW_EVENTS 0x00010BD6 +struct asm_stream_cmd_reg_tx_overflow_event{ + struct apr_hdr hdr; + u16 enable; + u16 reserved; +} __attribute__((packed)); + +/* Data Path commands */ +#define ASM_DATA_CMD_WRITE 0x00010BD9 +struct asm_stream_cmd_write{ + struct apr_hdr hdr; + u32 buf_add; + u32 avail_bytes; + u32 uid; + u32 msw_ts; + u32 lsw_ts; + u32 uflags; +} __attribute__((packed)); + +#define ASM_DATA_CMD_READ 0x00010BDA +struct asm_stream_cmd_read{ + struct apr_hdr hdr; + u32 buf_add; + u32 buf_size; + u32 uid; +} __attribute__((packed)); + +#define ASM_DATA_CMD_READ_COMPRESSED 0x00010DBF +struct asm_stream_cmd_read_compressed { + struct apr_hdr hdr; + u32 buf_add; + u32 buf_size; + u32 uid; +} __packed; + +#define ASM_DATA_CMD_MEDIA_FORMAT_UPDATE 0x00010BDC +#define ASM_DATA_EVENT_ENC_SR_CM_NOTIFY 0x00010BDE +struct asm_stream_media_format_update{ + struct apr_hdr hdr; + u32 format; + u32 cfg_size; + union { + struct asm_pcm_cfg pcm_cfg; + struct asm_adpcm_cfg adpcm_cfg; + struct asm_yadpcm_cfg yadpcm_cfg; + struct asm_midi_cfg midi_cfg; + struct asm_wma_cfg wma_cfg; + struct asm_wmapro_cfg wmapro_cfg; + struct asm_aac_cfg aac_cfg; + struct asm_flac_cfg flac_cfg; + struct asm_vorbis_cfg vorbis_cfg; + struct asm_multi_channel_pcm_fmt_blk multi_ch_pcm_cfg; + struct asm_amrwbplus_cfg amrwbplus_cfg; + } __attribute__((packed)) write_cfg; +} __attribute__((packed)); + + +/* Command Responses */ +#define ASM_STREAM_CMDRSP_GET_ENCDEC_PARAM 0x00010C12 +struct asm_stream_cmdrsp_get_readwrite_param{ + struct apr_hdr hdr; + u32 status; + u32 param_id; + u16 param_size; + u16 padding; + union { + struct asm_sbc_bitrate sbc_bitrate; + struct asm_immed_decode aac_dec; + } __attribute__((packed)) read_write_cfg; +} __attribute__((packed)); + + +#define ASM_SESSION_CMDRSP_GET_SESSION_TIME 0x00010BD8 +struct asm_stream_cmdrsp_get_session_time{ + struct apr_hdr hdr; + u32 status; + u32 msw_ts; + u32 lsw_ts; +} __attribute__((packed)); + +#define ASM_DATA_EVENT_WRITE_DONE 0x00010BDF +struct asm_data_event_write_done{ + u32 buf_add; + u32 status; +} __attribute__((packed)); + +#define ASM_DATA_EVENT_READ_DONE 0x00010BE0 +struct asm_data_event_read_done{ + u32 status; + u32 buffer_add; + u32 enc_frame_size; + u32 offset; + u32 msw_ts; + u32 lsw_ts; + u32 flags; + u32 num_frames; + u32 id; +} __attribute__((packed)); + +#define ASM_DATA_EVENT_READ_COMPRESSED_DONE 0x00010DC0 +struct asm_data_event_read_compressed_done { + u32 status; + u32 buffer_add; + u32 enc_frame_size; + u32 offset; + u32 msw_ts; + u32 lsw_ts; + u32 flags; + u32 num_frames; + u32 id; +} __packed; + +#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65 +struct asm_data_event_sr_cm_change_notify { + u32 sample_rate; + u16 no_of_channels; + u16 reserved; + u8 channel_map[8]; +} __packed; + +/* service level events */ + +#define ASM_SERVICE_CMDRSP_GET_STREAM_HANDLES 0x00010C1B +struct asm_svc_cmdrsp_get_strm_handles{ + struct apr_hdr hdr; + u32 num_handles; + u32 stream_handles; +} __attribute__((packed)); + + +#define ASM_SERVICE_CMDRSP_GET_WALLCLOCK_TIME 0x00010C1A +struct asm_svc_cmdrsp_get_wallclock_time{ + struct apr_hdr hdr; + u32 status; + u32 msw_ts; + u32 lsw_ts; +} __attribute__((packed)); + +/* + * Error code +*/ +#define ADSP_EOK 0x00000000 /* Success / completed / no errors. */ +#define ADSP_EFAILED 0x00000001 /* General failure. */ +#define ADSP_EBADPARAM 0x00000002 /* Bad operation parameter(s). */ +#define ADSP_EUNSUPPORTED 0x00000003 /* Unsupported routine/operation. */ +#define ADSP_EVERSION 0x00000004 /* Unsupported version. */ +#define ADSP_EUNEXPECTED 0x00000005 /* Unexpected problem encountered. */ +#define ADSP_EPANIC 0x00000006 /* Unhandled problem occurred. */ +#define ADSP_ENORESOURCE 0x00000007 /* Unable to allocate resource(s). */ +#define ADSP_EHANDLE 0x00000008 /* Invalid handle. */ +#define ADSP_EALREADY 0x00000009 /* Operation is already processed. */ +#define ADSP_ENOTREADY 0x0000000A /* Operation not ready to be processed*/ +#define ADSP_EPENDING 0x0000000B /* Operation is pending completion*/ +#define ADSP_EBUSY 0x0000000C /* Operation could not be accepted or + processed. */ +#define ADSP_EABORTED 0x0000000D /* Operation aborted due to an error. */ +#define ADSP_EPREEMPTED 0x0000000E /* Operation preempted by higher priority*/ +#define ADSP_ECONTINUE 0x0000000F /* Operation requests intervention + to complete. */ +#define ADSP_EIMMEDIATE 0x00000010 /* Operation requests immediate + intervention to complete. */ +#define ADSP_ENOTIMPL 0x00000011 /* Operation is not implemented. */ +#define ADSP_ENEEDMORE 0x00000012 /* Operation needs more data or resources*/ + +/* SRS TRUMEDIA GUIDS */ +#define SRS_TRUMEDIA_TOPOLOGY_ID 0x00010D90 +#define SRS_TRUMEDIA_MODULE_ID 0x10005010 +#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_PEQ 0x10005015 +#define SRS_TRUMEDIA_PARAMS_HL 0x10005016 + +/* SRS STUDIO SOUND 3D GUIDS */ +#define SRS_SS3D_TOPOLOGY_ID 0x00010720 +#define SRS_SS3D_MODULE_ID 0x10005020 +#define SRS_SS3D_PARAMS 0x10005021 +#define SRS_SS3D_PARAMS_CTRL 0x10005022 +#define SRS_SS3D_PARAMS_FILTER 0x10005023 + +/* SRS ALSA CMD MASKS */ +#define SRS_CMD_UPLOAD 0x7FFF0000 +#define SRS_PARAM_INDEX_MASK 0x80000000 +#define SRS_PARAM_OFFSET_MASK 0x3FFF0000 +#define SRS_PARAM_VALUE_MASK 0x0000FFFF + +/* SRS TRUMEDIA start */ +#define SRS_ID_GLOBAL 0x00000001 +#define SRS_ID_WOWHD 0x00000002 +#define SRS_ID_CSHP 0x00000003 +#define SRS_ID_HPF 0x00000004 +#define SRS_ID_PEQ 0x00000005 +#define SRS_ID_HL 0x00000006 + +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; +} __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]; +} __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_PEQ { + uint32_t v1; + uint16_t v2; + uint16_t v3; + uint16_t v4; + uint16_t v____A1; + uint32_t v5[26]; + uint32_t v6[26]; +} __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 { + 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_PEQ peq; + struct srs_trumedia_params_HL hl; +} __packed; + +int srs_trumedia_open(int port_id, int srs_tech_id, void *srs_params); +/* SRS TruMedia end */ + +/* SRS Studio Sound 3D start */ +#define SRS_ID_SS3D_GLOBAL 0x00000001 +#define SRS_ID_SS3D_CTRL 0x00000002 +#define SRS_ID_SS3D_FILTER 0x00000003 + +struct srs_SS3D_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; +} __packed; + +struct srs_SS3D_ctrl_params { + uint8_t v[236]; +} __packed; + +struct srs_SS3D_filter_params { + uint8_t v[28 + 2752]; +} __packed; + +struct srs_SS3D_params { + struct srs_SS3D_params_GLOBAL global; + struct srs_SS3D_ctrl_params ss3d; + struct srs_SS3D_filter_params ss3d_f; +} __packed; + +int srs_ss3d_open(int port_id, int srs_tech_id, void *srs_params); +/* SRS Studio Sound 3D end */ +#endif /*_APR_AUDIO_H_*/ diff --git a/include/sound/audio_cal_utils.h b/include/sound/audio_cal_utils.h new file mode 100644 index 000000000000..b28b3bdf4e83 --- /dev/null +++ b/include/sound/audio_cal_utils.h @@ -0,0 +1,102 @@ +/* Copyright (c) 2014, 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 _AUDIO_CAL_UTILS_H +#define _AUDIO_CAL_UTILS_H + +#include <linux/msm_ion.h> +#include <linux/msm_audio_ion.h> +#include <linux/msm_audio_calibration.h> +#include "audio_calibration.h" + +struct cal_data { + size_t size; + void *kvaddr; + phys_addr_t paddr; +}; + +struct mem_map_data { + size_t map_size; + int32_t q6map_handle; + int32_t ion_map_handle; + struct ion_client *ion_client; + struct ion_handle *ion_handle; +}; + +struct cal_block_data { + size_t client_info_size; + void *client_info; + void *cal_info; + struct list_head list; + struct cal_data cal_data; + struct mem_map_data map_data; + int32_t buffer_number; +}; + +struct cal_util_callbacks { + int (*map_cal) + (int32_t cal_type, struct cal_block_data *cal_block); + int (*unmap_cal) + (int32_t cal_type, struct cal_block_data *cal_block); + bool (*match_block) + (struct cal_block_data *cal_block, void *user_data); +}; + +struct cal_type_info { + struct audio_cal_reg reg; + struct cal_util_callbacks cal_util_callbacks; +}; + +struct cal_type_data { + struct cal_type_info info; + struct mutex lock; + struct list_head cal_blocks; +}; + + +/* to register & degregister with cal util driver */ +int cal_utils_create_cal_types(int num_cal_types, + struct cal_type_data **cal_type, + struct cal_type_info *info); +void cal_utils_destroy_cal_types(int num_cal_types, + struct cal_type_data **cal_type); + +/* common functions for callbacks */ +int cal_utils_alloc_cal(size_t data_size, void *data, + struct cal_type_data *cal_type, + size_t client_info_size, void *client_info); +int cal_utils_dealloc_cal(size_t data_size, void *data, + struct cal_type_data *cal_type); +int cal_utils_set_cal(size_t data_size, void *data, + struct cal_type_data *cal_type, + size_t client_info_size, void *client_info); + +/* use for SSR */ +void cal_utils_clear_cal_block_q6maps(int num_cal_types, + struct cal_type_data **cal_type); + + +/* common matching functions used to add blocks */ +bool cal_utils_match_buf_num(struct cal_block_data *cal_block, + void *user_data); + +/* common matching functions to find cal blocks */ +struct cal_block_data *cal_utils_get_only_cal_block( + struct cal_type_data *cal_type); + +/* Size of calibration specific data */ +size_t get_cal_info_size(int32_t cal_type); +size_t get_user_cal_type_size(int32_t cal_type); + +/* Version of the cal type*/ +int32_t cal_utils_get_cal_type_version(void *cal_type_data); +#endif diff --git a/include/sound/audio_calibration.h b/include/sound/audio_calibration.h new file mode 100644 index 000000000000..5f6b5e37aa5c --- /dev/null +++ b/include/sound/audio_calibration.h @@ -0,0 +1,40 @@ +/* Copyright (c) 2014, 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 _AUDIO_CALIBRATION_H +#define _AUDIO_CALIBRATION_H + +#include <linux/msm_audio_calibration.h> + +/* Used by driver in buffer_number field to notify client + * To update all blocks, for example: freeing all memory */ +#define ALL_CAL_BLOCKS -1 + + +struct audio_cal_callbacks { + int (*alloc) (int32_t cal_type, size_t data_size, void *data); + int (*dealloc) (int32_t cal_type, size_t data_size, void *data); + int (*pre_cal) (int32_t cal_type, size_t data_size, void *data); + int (*set_cal) (int32_t cal_type, size_t data_size, void *data); + int (*get_cal) (int32_t cal_type, size_t data_size, void *data); + int (*post_cal) (int32_t cal_type, size_t data_size, void *data); +}; + +struct audio_cal_reg { + int32_t cal_type; + struct audio_cal_callbacks callbacks; +}; + +int audio_cal_register(int num_cal_types, struct audio_cal_reg *reg_data); +int audio_cal_deregister(int num_cal_types, struct audio_cal_reg *reg_data); + +#endif diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 57872c8f1151..07555417fac5 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -84,6 +84,7 @@ struct snd_compr_stream { bool next_track; bool partial_drain; void *private_data; + struct snd_soc_pcm_runtime *be; }; /** @@ -98,6 +99,8 @@ struct snd_compr_stream { * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream * @get_metadata: retrieves the requested metadata values from stream + * @set_next_track_param: send codec specific data of subsequent track + * in gapless * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory @@ -120,6 +123,8 @@ struct snd_compr_ops { struct snd_compr_metadata *metadata); int (*get_metadata)(struct snd_compr_stream *stream, struct snd_compr_metadata *metadata); + int (*set_next_track_param)(struct snd_compr_stream *stream, + union snd_codec_options *codec_options); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); @@ -161,6 +166,7 @@ int snd_compress_register(struct snd_compr *device); int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, struct snd_compr *compr); +void snd_compress_free(struct snd_card *card, struct snd_compr *compr); /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the diff --git a/include/sound/core.h b/include/sound/core.h index cdfecafff0f4..7b05f88cacac 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -134,6 +134,9 @@ struct snd_card { struct device card_dev; /* cardX object for sysfs */ const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */ bool registered; /* card_dev is registered? */ + int offline; /* if this sound card is offline */ + unsigned long offline_change; + wait_queue_head_t offline_poll_wait; #ifdef CONFIG_PM unsigned int power_state; /* power state */ @@ -265,6 +268,8 @@ int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); #define snd_card_unref(card) put_device(&(card)->card_dev) +void snd_card_change_online_state(struct snd_card *card, int online); +bool snd_card_is_online_state(struct snd_card *card); #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/include/sound/cpe_cmi.h b/include/sound/cpe_cmi.h new file mode 100644 index 000000000000..cbf83e7a7e91 --- /dev/null +++ b/include/sound/cpe_cmi.h @@ -0,0 +1,492 @@ +/* + * Copyright (c) 2014-2016, 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 __CPE_CMI_H__ +#define __CPE_CMI_H__ + +#include <linux/types.h> + +#define CPE_AFE_PORT_1_TX 1 +#define CPE_AFE_PORT_3_TX 3 +#define CPE_AFE_PORT_ID_2_OUT 0x02 +#define CMI_INBAND_MESSAGE_SIZE 127 + +/* + * Multiple mad types can be supported at once. + * these values can be OR'ed to form the set of + * supported mad types + */ +#define MAD_TYPE_AUDIO (1 << 0) +#define MAD_TYPE_BEACON (1 << 1) +#define MAD_TYPE_ULTRASND (1 << 2) + +/* Core service command opcodes */ +#define CPE_CORE_SVC_CMD_SHARED_MEM_ALLOC (0x3001) +#define CPE_CORE_SVC_CMDRSP_SHARED_MEM_ALLOC (0x3002) +#define CPE_CORE_SVC_CMD_SHARED_MEM_DEALLOC (0x3003) +#define CPE_CORE_SVC_CMD_DRAM_ACCESS_REQ (0x3004) +#define CPE_CORE_SVC_EVENT_SYSTEM_BOOT (0x3005) +/* core service command opcodes for WCD9335 */ +#define CPE_CORE_SVC_CMD_CFG_CLK_PLAN (0x3006) +#define CPE_CORE_SVC_CMD_CLK_FREQ_REQUEST (0x3007) + +#define CPE_BOOT_SUCCESS 0x00 +#define CPE_BOOT_FAILED 0x01 + +#define CPE_CORE_VERSION_SYSTEM_BOOT_EVENT 0x01 + +/* LSM Service command opcodes */ +#define CPE_LSM_SESSION_CMD_OPEN_TX (0x2000) +#define CPE_LSM_SESSION_CMD_SET_PARAMS (0x2001) +#define CPE_LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x2002) +#define CPE_LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x2003) +#define CPE_LSM_SESSION_CMD_START (0x2004) +#define CPE_LSM_SESSION_CMD_STOP (0x2005) +#define CPE_LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x2006) +#define CPE_LSM_SESSION_CMD_CLOSE_TX (0x2007) +#define CPE_LSM_SESSION_CMD_SHARED_MEM_ALLOC (0x2008) +#define CPE_LSM_SESSION_CMDRSP_SHARED_MEM_ALLOC (0x2009) +#define CPE_LSM_SESSION_CMD_SHARED_MEM_DEALLOC (0x200A) +#define CPE_LSM_SESSION_CMD_TX_BUFF_OUTPUT_CONFIG (0x200f) +#define CPE_LSM_SESSION_CMD_OPEN_TX_V2 (0x200D) +#define CPE_LSM_SESSION_CMD_SET_PARAMS_V2 (0x200E) + +/* LSM Service module and param IDs */ +#define CPE_LSM_MODULE_ID_VOICE_WAKEUP (0x00012C00) +#define CPE_LSM_MODULE_ID_VOICE_WAKEUP_V2 (0x00012C0D) +#define CPE_LSM_MODULE_FRAMEWORK (0x00012C0E) + +#define CPE_LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01) +#define CPE_LSM_PARAM_ID_OPERATION_MODE (0x00012C02) +#define CPE_LSM_PARAM_ID_GAIN (0x00012C03) +#define CPE_LSM_PARAM_ID_CONNECT_TO_PORT (0x00012C04) +#define CPE_LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS (0x00012C07) + +/* LSM LAB command opcodes */ +#define CPE_LSM_SESSION_CMD_EOB 0x0000200B +#define CPE_LSM_MODULE_ID_LAB 0x00012C08 +/* used for enable/disable lab*/ +#define CPE_LSM_PARAM_ID_LAB_ENABLE 0x00012C09 +/* used for T in LAB config DSP internal buffer*/ +#define CPE_LSM_PARAM_ID_LAB_CONFIG 0x00012C0A +#define CPE_LSM_PARAM_ID_REGISTER_SOUND_MODEL (0x00012C14) +#define CPE_LSM_PARAM_ID_DEREGISTER_SOUND_MODEL (0x00012C15) +#define CPE_LSM_PARAM_ID_MEDIA_FMT (0x00012C1E) + +/* AFE Service command opcodes */ +#define CPE_AFE_PORT_CMD_START (0x1001) +#define CPE_AFE_PORT_CMD_STOP (0x1002) +#define CPE_AFE_PORT_CMD_SUSPEND (0x1003) +#define CPE_AFE_PORT_CMD_RESUME (0x1004) +#define CPE_AFE_PORT_CMD_SHARED_MEM_ALLOC (0x1005) +#define CPE_AFE_PORT_CMDRSP_SHARED_MEM_ALLOC (0x1006) +#define CPE_AFE_PORT_CMD_SHARED_MEM_DEALLOC (0x1007) +#define CPE_AFE_PORT_CMD_GENERIC_CONFIG (0x1008) +#define CPE_AFE_SVC_CMD_LAB_MODE (0x1009) + +/* AFE Service module and param IDs */ +#define CPE_AFE_CMD_SET_PARAM (0x1000) +#define CPE_AFE_MODULE_ID_SW_MAD (0x0001022D) +#define CPE_AFE_PARAM_ID_SW_MAD_CFG (0x0001022E) +#define CPE_AFE_PARAM_ID_SVM_MODEL (0x0001022F) + +#define CPE_AFE_MODULE_HW_MAD (0x00010230) +#define CPE_AFE_PARAM_ID_HW_MAD_CTL (0x00010232) +#define CPE_AFE_PARAM_ID_HW_MAD_CFG (0x00010231) + +#define CPE_AFE_MODULE_AUDIO_DEV_INTERFACE (0x0001020C) +#define CPE_AFE_PARAM_ID_GENERIC_PORT_CONFIG (0x00010253) + +#define CPE_CMI_BASIC_RSP_OPCODE (0x0001) +#define CPE_HDR_MAX_PLD_SIZE (0x7F) + +#define CMI_OBM_FLAG_IN_BAND 0 +#define CMI_OBM_FLAG_OUT_BAND 1 + +#define CMI_SHMEM_ALLOC_FAILED 0xff + +/* + * Future Service ID's can be added one line + * before the CMI_CPE_SERVICE_ID_MAX + */ +enum { + CMI_CPE_SERVICE_ID_MIN = 0, + CMI_CPE_CORE_SERVICE_ID, + CMI_CPE_AFE_SERVICE_ID, + CMI_CPE_LSM_SERVICE_ID, + CMI_CPE_SERVICE_ID_MAX, +}; + +#define CPE_LSM_SESSION_ID_MAX 2 + +#define IS_VALID_SESSION_ID(s_id) \ + (s_id <= CPE_LSM_SESSION_ID_MAX) + +#define IS_VALID_SERVICE_ID(s_id) \ + (s_id > CMI_CPE_SERVICE_ID_MIN && \ + s_id < CMI_CPE_SERVICE_ID_MAX) + +#define IS_VALID_PLD_SIZE(p_size) \ + (p_size <= CPE_HDR_MAX_PLD_SIZE) + +#define CMI_HDR_SET_OPCODE(hdr, cmd) (hdr->opcode = cmd) + + +#define CMI_HDR_SET(hdr_info, mask, shift, value) \ + (hdr_info = (((hdr_info) & ~(mask)) | \ + ((value << shift) & mask))) + +#define SVC_ID_SHIFT 4 +#define SVC_ID_MASK (0x07 << SVC_ID_SHIFT) + +#define SESSION_ID_SHIFT 0 +#define SESSION_ID_MASK (0x0F << SESSION_ID_SHIFT) + +#define PAYLD_SIZE_SHIFT 0 +#define PAYLD_SIZE_MASK (0x7F << PAYLD_SIZE_SHIFT) + +#define OBM_FLAG_SHIFT 7 +#define OBM_FLAG_MASK (1 << OBM_FLAG_SHIFT) + +#define VERSION_SHIFT 7 +#define VERSION_MASK (1 << VERSION_SHIFT) + +#define CMI_HDR_SET_SERVICE(hdr, s_id) \ + CMI_HDR_SET(hdr->hdr_info, SVC_ID_MASK,\ + SVC_ID_SHIFT, s_id) +#define CMI_HDR_GET_SERVICE(hdr) \ + ((hdr->hdr_info >> SVC_ID_SHIFT) & \ + (SVC_ID_MASK >> SVC_ID_SHIFT)) + + +#define CMI_HDR_SET_SESSION(hdr, s_id) \ + CMI_HDR_SET(hdr->hdr_info, SESSION_ID_MASK,\ + SESSION_ID_SHIFT, s_id) + +#define CMI_HDR_GET_SESSION_ID(hdr) \ + ((hdr->hdr_info >> SESSION_ID_SHIFT) & \ + (SESSION_ID_MASK >> SESSION_ID_SHIFT)) + +#define CMI_GET_HEADER(msg) ((struct cmi_hdr *)(msg)) +#define CMI_GET_PAYLOAD(msg) ((void *)(CMI_GET_HEADER(msg) + 1)) +#define CMI_GET_OPCODE(msg) (CMI_GET_HEADER(msg)->opcode) + +#define CMI_HDR_SET_VERSION(hdr, ver) \ + CMI_HDR_SET(hdr->hdr_info, VERSION_MASK, \ + VERSION_SHIFT, ver) + +#define CMI_HDR_SET_PAYLOAD_SIZE(hdr, p_size) \ + CMI_HDR_SET(hdr->pld_info, PAYLD_SIZE_MASK, \ + PAYLD_SIZE_SHIFT, p_size) + +#define CMI_HDR_GET_PAYLOAD_SIZE(hdr) \ + ((hdr->pld_info >> PAYLD_SIZE_SHIFT) & \ + (PAYLD_SIZE_MASK >> PAYLD_SIZE_SHIFT)) + +#define CMI_HDR_SET_OBM(hdr, obm_flag) \ + CMI_HDR_SET(hdr->pld_info, OBM_FLAG_MASK, \ + OBM_FLAG_SHIFT, obm_flag) + +#define CMI_HDR_GET_OBM_FLAG(hdr) \ + ((hdr->pld_info >> OBM_FLAG_SHIFT) & \ + (OBM_FLAG_MASK >> OBM_FLAG_SHIFT)) + +struct cmi_hdr { + /* + * bits 0:3 is session id + * bits 4:6 is service id + * bit 7 is the version flag + */ + u8 hdr_info; + + /* + * bits 0:6 is payload size in case of in-band message + * bits 0:6 is size (OBM message size) + * bit 7 is the OBM flag + */ + u8 pld_info; + + /* 16 bit command opcode */ + u16 opcode; +} __packed; + +union cpe_addr { + u64 msw_lsw; + void *kvaddr; +} __packed; + +struct cmi_obm { + u32 version; + u32 size; + union cpe_addr data_ptr; + u32 mem_handle; +} __packed; + +struct cmi_obm_msg { + struct cmi_hdr hdr; + struct cmi_obm pld; +} __packed; + +struct cmi_core_svc_event_system_boot { + u8 status; + u8 version; + u16 sfr_buff_size; + u32 sfr_buff_address; +} __packed; + +struct cmi_core_svc_cmd_shared_mem_alloc { + u32 size; +} __packed; + +struct cmi_core_svc_cmdrsp_shared_mem_alloc { + u32 addr; +} __packed; + +struct cmi_core_svc_cmd_clk_freq_request { + u32 clk_freq; +} __packed; + +struct cmi_msg_transport { + u32 size; + u32 addr; +} __packed; + +struct cmi_basic_rsp_result { + u8 status; +} __packed; + +struct cpe_lsm_cmd_open_tx { + struct cmi_hdr hdr; + u16 app_id; + u16 reserved; + u32 sampling_rate; +} __packed; + +struct cpe_lsm_cmd_open_tx_v2 { + struct cmi_hdr hdr; + u32 topology_id; +} __packed; + +struct cpe_cmd_shmem_alloc { + struct cmi_hdr hdr; + u32 size; +} __packed; + +struct cpe_cmdrsp_shmem_alloc { + struct cmi_hdr hdr; + u32 addr; +} __packed; + +struct cpe_cmd_shmem_dealloc { + struct cmi_hdr hdr; + u32 addr; +} __packed; + +struct cpe_lsm_event_detect_v2 { + struct cmi_hdr hdr; + u8 detection_status; + u8 size; + u8 payload[0]; +} __packed; + +struct cpe_lsm_psize_res { + u16 param_size; + u16 reserved; +} __packed; + +union cpe_lsm_param_size { + u32 param_size; + struct cpe_lsm_psize_res sr; +} __packed; + +struct cpe_param_data { + u32 module_id; + u32 param_id; + union cpe_lsm_param_size p_size; +} __packed; + +struct cpe_lsm_param_epd_thres { + struct cmi_hdr hdr; + struct cpe_param_data param; + u32 minor_version; + u32 epd_begin; + u32 epd_end; +} __packed; + +struct cpe_lsm_param_gain { + struct cmi_hdr hdr; + struct cpe_param_data param; + u32 minor_version; + u16 gain; + u16 reserved; +} __packed; + +struct cpe_afe_hw_mad_ctrl { + struct cpe_param_data param; + u32 minor_version; + u16 mad_type; + u16 mad_enable; +} __packed; + +struct cpe_afe_port_cfg { + struct cpe_param_data param; + u32 minor_version; + u16 bit_width; + u16 num_channels; + u32 sample_rate; +} __packed; + +struct cpe_afe_cmd_port_cfg { + struct cmi_hdr hdr; + u8 bit_width; + u8 num_channels; + u16 buffer_size; + u32 sample_rate; +} __packed; + +struct cpe_afe_params { + struct cmi_hdr hdr; + struct cpe_afe_hw_mad_ctrl hw_mad_ctrl; + struct cpe_afe_port_cfg port_cfg; +} __packed; + +struct cpe_afe_svc_cmd_mode { + struct cmi_hdr hdr; + u8 mode; +} __packed; + +struct cpe_lsm_param_opmode { + struct cmi_hdr hdr; + struct cpe_param_data param; + u32 minor_version; + u16 mode; + u16 reserved; +} __packed; + +struct cpe_lsm_param_connectport { + struct cmi_hdr hdr; + struct cpe_param_data param; + u32 minor_version; + u16 afe_port_id; + u16 reserved; +} __packed; + +/* + * This cannot be sent to CPE as is, + * need to append the conf_levels dynamically + */ +struct cpe_lsm_conf_level { + struct cmi_hdr hdr; + struct cpe_param_data param; + u8 num_active_models; +} __packed; + +struct cpe_lsm_output_format_cfg { + struct cmi_hdr hdr; + u8 format; + u8 packing; + u8 data_path_events; +} __packed; + +struct cpe_lsm_lab_enable { + struct cpe_param_data param; + u16 enable; + u16 reserved; +} __packed; + +struct cpe_lsm_control_lab { + struct cmi_hdr hdr; + struct cpe_lsm_lab_enable lab_enable; +} __packed; + +struct cpe_lsm_lab_config { + struct cpe_param_data param; + u32 minor_ver; + u32 latency; +} __packed; + +struct cpe_lsm_lab_latency_config { + struct cmi_hdr hdr; + struct cpe_lsm_lab_config latency_cfg; +} __packed; + +struct cpe_lsm_media_fmt_param { + struct cmi_hdr hdr; + struct cpe_param_data param; + u32 minor_version; + u32 sample_rate; + u16 num_channels; + u16 bit_width; +} __packed; + + +#define CPE_PARAM_LSM_LAB_LATENCY_SIZE (\ + sizeof(struct cpe_lsm_lab_latency_config) - \ + sizeof(struct cmi_hdr)) +#define PARAM_SIZE_LSM_LATENCY_SIZE (\ + sizeof(struct cpe_lsm_lab_config) - \ + sizeof(struct cpe_param_data)) +#define CPE_PARAM_SIZE_LSM_LAB_CONTROL (\ + sizeof(struct cpe_lsm_control_lab) - \ + sizeof(struct cmi_hdr)) +#define PARAM_SIZE_LSM_CONTROL_SIZE (sizeof(struct cpe_lsm_lab_enable) - \ + sizeof(struct cpe_param_data)) +#define PARAM_SIZE_AFE_HW_MAD_CTRL (sizeof(struct cpe_afe_hw_mad_ctrl) - \ + sizeof(struct cpe_param_data)) +#define PARAM_SIZE_AFE_PORT_CFG (sizeof(struct cpe_afe_port_cfg) - \ + sizeof(struct cpe_param_data)) +#define CPE_AFE_PARAM_PAYLOAD_SIZE (sizeof(struct cpe_afe_params) - \ + sizeof(struct cmi_hdr)) + +#define OPEN_CMD_PAYLOAD_SIZE (sizeof(struct cpe_lsm_cmd_open_tx) - \ + sizeof(struct cmi_hdr)) +#define OPEN_V2_CMD_PAYLOAD_SIZE (sizeof(struct cpe_lsm_cmd_open_tx_v2) - \ + sizeof(struct cmi_hdr)) +#define SHMEM_ALLOC_CMD_PLD_SIZE (sizeof(struct cpe_cmd_shmem_alloc) - \ + sizeof(struct cmi_hdr)) + +#define SHMEM_DEALLOC_CMD_PLD_SIZE (sizeof(struct cpe_cmd_shmem_dealloc) - \ + sizeof(struct cmi_hdr)) +#define OUT_FMT_CFG_CMD_PAYLOAD_SIZE ( \ + sizeof(struct cpe_lsm_output_format_cfg) - \ + sizeof(struct cmi_hdr)) + +#define CPE_AFE_CMD_PORT_CFG_PAYLOAD_SIZE \ + (sizeof(struct cpe_afe_cmd_port_cfg) - \ + sizeof(struct cmi_hdr)) + +#define CPE_AFE_CMD_MODE_PAYLOAD_SIZE \ + (sizeof(struct cpe_afe_svc_cmd_mode) - \ + sizeof(struct cmi_hdr)) +#define CPE_CMD_EPD_THRES_PLD_SIZE (sizeof(struct cpe_lsm_param_epd_thres) - \ + sizeof(struct cmi_hdr)) +#define CPE_EPD_THRES_PARAM_SIZE ((CPE_CMD_EPD_THRES_PLD_SIZE) - \ + sizeof(struct cpe_param_data)) +#define CPE_CMD_OPMODE_PLD_SIZE (sizeof(struct cpe_lsm_param_opmode) - \ + sizeof(struct cmi_hdr)) +#define CPE_OPMODE_PARAM_SIZE ((CPE_CMD_OPMODE_PLD_SIZE) -\ + sizeof(struct cpe_param_data)) +#define CPE_CMD_CONNECTPORT_PLD_SIZE \ + (sizeof(struct cpe_lsm_param_connectport) - \ + sizeof(struct cmi_hdr)) +#define CPE_CONNECTPORT_PARAM_SIZE ((CPE_CMD_CONNECTPORT_PLD_SIZE) - \ + sizeof(struct cpe_param_data)) +#define CPE_CMD_GAIN_PLD_SIZE (sizeof(struct cpe_lsm_param_gain) - \ + sizeof(struct cmi_hdr)) +#define CPE_GAIN_PARAM_SIZE ((CPE_CMD_GAIN_PLD_SIZE) - \ + sizeof(struct cpe_param_data)) +#define CPE_MEDIA_FMT_PLD_SIZE (sizeof(struct cpe_lsm_media_fmt_param) - \ + sizeof(struct cmi_hdr)) +#define CPE_MEDIA_FMT_PARAM_SIZE ((CPE_MEDIA_FMT_PLD_SIZE) - \ + sizeof(struct cpe_param_data)) +#endif /* __CPE_CMI_H__ */ diff --git a/include/sound/cpe_core.h b/include/sound/cpe_core.h new file mode 100644 index 000000000000..846cf819b9e5 --- /dev/null +++ b/include/sound/cpe_core.h @@ -0,0 +1,179 @@ +/* + * Copyright (c) 2013-2017, 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 __CPE_CORE_H__ +#define __CPE_CORE_H__ + +#include <linux/types.h> +#include <linux/wait.h> +#include <linux/dma-mapping.h> +#include <sound/lsm_params.h> + +enum { + CMD_INIT_STATE = 0, + CMD_SENT, + CMD_RESP_RCVD, +}; + +enum wcd_cpe_event { + WCD_CPE_PRE_ENABLE = 1, + WCD_CPE_POST_ENABLE, + WCD_CPE_PRE_DISABLE, + WCD_CPE_POST_DISABLE, +}; + +struct wcd_cpe_afe_port_cfg { + u8 port_id; + u16 bit_width; + u16 num_channels; + u32 sample_rate; +}; + +struct lsm_out_fmt_cfg { + u8 format; + u8 pack_mode; + u8 data_path_events; + u8 transfer_mode; +}; + +struct lsm_hw_params { + u32 sample_rate; + u16 num_chs; + u16 bit_width; +}; + +struct cpe_lsm_session { + /* sound model related */ + void *snd_model_data; + u8 *conf_levels; + void *cmi_reg_handle; + + /* Clients private data */ + void *priv_d; + + void (*event_cb) (void *priv_data, + u8 detect_status, + u8 size, u8 *payload); + + struct completion cmd_comp; + struct wcd_cpe_afe_port_cfg afe_port_cfg; + struct wcd_cpe_afe_port_cfg afe_out_port_cfg; + struct mutex lsm_lock; + + u32 snd_model_size; + u32 lsm_mem_handle; + u16 cmd_err_code; + u8 id; + u8 num_confidence_levels; + u16 afe_out_port_id; + struct task_struct *lsm_lab_thread; + bool started; + + u32 lab_enable; + struct lsm_out_fmt_cfg out_fmt_cfg; + + bool is_topology_used; +}; + +struct wcd_cpe_afe_ops { + int (*afe_set_params) (void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg, + bool afe_mad_ctl); + + int (*afe_port_start) (void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg); + + int (*afe_port_stop) (void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg); + + int (*afe_port_suspend) (void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg); + + int (*afe_port_resume) (void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg); + + int (*afe_port_cmd_cfg)(void *core_handle, + struct wcd_cpe_afe_port_cfg *cfg); +}; + +struct wcd_cpe_lsm_ops { + + struct cpe_lsm_session *(*lsm_alloc_session) + (void *core_handle, void *lsm_priv_d, + void (*event_cb) (void *priv_data, + u8 detect_status, + u8 size, u8 *payload)); + + int (*lsm_dealloc_session) + (void *core_handle, struct cpe_lsm_session *); + + int (*lsm_open_tx) (void *core_handle, + struct cpe_lsm_session *, u16, u16); + + int (*lsm_close_tx) (void *core_handle, + struct cpe_lsm_session *); + + int (*lsm_shmem_alloc) (void *core_handle, + struct cpe_lsm_session *, u32 size); + + int (*lsm_shmem_dealloc) (void *core_handle, + struct cpe_lsm_session *); + + int (*lsm_register_snd_model) (void *core_handle, + struct cpe_lsm_session *, + enum lsm_detection_mode, bool); + + int (*lsm_deregister_snd_model) (void *core_handle, + struct cpe_lsm_session *); + + int (*lsm_get_afe_out_port_id)(void *core_handle, + struct cpe_lsm_session *session); + + int (*lsm_start) (void *core_handle, + struct cpe_lsm_session *); + + int (*lsm_stop) (void *core_handle, + struct cpe_lsm_session *); + + int (*lsm_lab_control)(void *core_handle, + struct cpe_lsm_session *session, + bool enable); + + int (*lab_ch_setup)(void *core_handle, + struct cpe_lsm_session *session, + enum wcd_cpe_event event); + + int (*lsm_set_data) (void *core_handle, + struct cpe_lsm_session *session, + enum lsm_detection_mode detect_mode, + bool detect_failure); + int (*lsm_set_fmt_cfg)(void *core_handle, + struct cpe_lsm_session *session); + int (*lsm_set_one_param)(void *core_handle, + struct cpe_lsm_session *session, + struct lsm_params_info *p_info, + void *data, uint32_t param_type); + void (*lsm_get_snd_model_offset) + (void *core_handle, struct cpe_lsm_session *, + size_t *offset); + int (*lsm_set_media_fmt_params)(void *core_handle, + struct cpe_lsm_session *session, + struct lsm_hw_params *param); + int (*lsm_set_port)(void *core_handle, + struct cpe_lsm_session *session, void *data); +}; + +int wcd_cpe_get_lsm_ops(struct wcd_cpe_lsm_ops *); +int wcd_cpe_get_afe_ops(struct wcd_cpe_afe_ops *); +void *wcd_cpe_get_core_handle(struct snd_soc_codec *); +#endif diff --git a/include/sound/cpe_err.h b/include/sound/cpe_err.h new file mode 100644 index 000000000000..e31fa757aa4d --- /dev/null +++ b/include/sound/cpe_err.h @@ -0,0 +1,166 @@ +/* + * Copyright (c) 2015, 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 __CPE_ERR__ +#define __CPE_ERR__ + +#include <linux/errno.h> + +/* ERROR CODES */ +/* Success. The operation completed with no errors. */ +#define CPE_EOK 0x00000000 +/* General failure. */ +#define CPE_EFAILED 0x00000001 +/* Bad operation parameter. */ +#define CPE_EBADPARAM 0x00000002 +/* Unsupported routine or operation. */ +#define CPE_EUNSUPPORTED 0x00000003 +/* Unsupported version. */ +#define CPE_EVERSION 0x00000004 +/* Unexpected problem encountered. */ +#define CPE_EUNEXPECTED 0x00000005 +/* Unhandled problem occurred. */ +#define CPE_EPANIC 0x00000006 +/* Unable to allocate resource. */ +#define CPE_ENORESOURCE 0x00000007 +/* Invalid handle. */ +#define CPE_EHANDLE 0x00000008 +/* Operation is already processed. */ +#define CPE_EALREADY 0x00000009 +/* Operation is not ready to be processed. */ +#define CPE_ENOTREADY 0x0000000A +/* Operation is pending completion. */ +#define CPE_EPENDING 0x0000000B +/* Operation could not be accepted or processed. */ +#define CPE_EBUSY 0x0000000C +/* Operation aborted due to an error. */ +#define CPE_EABORTED 0x0000000D +/* Operation preempted by a higher priority. */ +#define CPE_EPREEMPTED 0x0000000E +/* Operation requests intervention to complete. */ +#define CPE_ECONTINUE 0x0000000F +/* Operation requests immediate intervention to complete. */ +#define CPE_EIMMEDIATE 0x00000010 +/* Operation is not implemented. */ +#define CPE_ENOTIMPL 0x00000011 +/* Operation needs more data or resources. */ +#define CPE_ENEEDMORE 0x00000012 +/* Operation does not have memory. */ +#define CPE_ENOMEMORY 0x00000014 +/* Item does not exist. */ +#define CPE_ENOTEXIST 0x00000015 +/* Operation is finished. */ +#define CPE_ETERMINATED 0x00000016 +/* Max count for adsp error code sent to HLOS*/ +#define CPE_ERR_MAX (CPE_ETERMINATED + 1) + + +/* ERROR STRING */ +/* Success. The operation completed with no errors. */ +#define CPE_EOK_STR "CPE_EOK" +/* General failure. */ +#define CPE_EFAILED_STR "CPE_EFAILED" +/* Bad operation parameter. */ +#define CPE_EBADPARAM_STR "CPE_EBADPARAM" +/* Unsupported routine or operation. */ +#define CPE_EUNSUPPORTED_STR "CPE_EUNSUPPORTED" +/* Unsupported version. */ +#define CPE_EVERSION_STR "CPE_EVERSION" +/* Unexpected problem encountered. */ +#define CPE_EUNEXPECTED_STR "CPE_EUNEXPECTED" +/* Unhandled problem occurred. */ +#define CPE_EPANIC_STR "CPE_EPANIC" +/* Unable to allocate resource. */ +#define CPE_ENORESOURCE_STR "CPE_ENORESOURCE" +/* Invalid handle. */ +#define CPE_EHANDLE_STR "CPE_EHANDLE" +/* Operation is already processed. */ +#define CPE_EALREADY_STR "CPE_EALREADY" +/* Operation is not ready to be processed. */ +#define CPE_ENOTREADY_STR "CPE_ENOTREADY" +/* Operation is pending completion. */ +#define CPE_EPENDING_STR "CPE_EPENDING" +/* Operation could not be accepted or processed. */ +#define CPE_EBUSY_STR "CPE_EBUSY" +/* Operation aborted due to an error. */ +#define CPE_EABORTED_STR "CPE_EABORTED" +/* Operation preempted by a higher priority. */ +#define CPE_EPREEMPTED_STR "CPE_EPREEMPTED" +/* Operation requests intervention to complete. */ +#define CPE_ECONTINUE_STR "CPE_ECONTINUE" +/* Operation requests immediate intervention to complete. */ +#define CPE_EIMMEDIATE_STR "CPE_EIMMEDIATE" +/* Operation is not implemented. */ +#define CPE_ENOTIMPL_STR "CPE_ENOTIMPL" +/* Operation needs more data or resources. */ +#define CPE_ENEEDMORE_STR "CPE_ENEEDMORE" +/* Operation does not have memory. */ +#define CPE_ENOMEMORY_STR "CPE_ENOMEMORY" +/* Item does not exist. */ +#define CPE_ENOTEXIST_STR "CPE_ENOTEXIST" +/* Operation is finished. */ +#define CPE_ETERMINATED_STR "CPE_ETERMINATED" +/* Unexpected error code. */ +#define CPE_ERR_MAX_STR "CPE_ERR_MAX" + + +struct cpe_err_code { + int lnx_err_code; + char *cpe_err_str; +}; + + +static struct cpe_err_code cpe_err_code_info[CPE_ERR_MAX+1] = { + { 0, CPE_EOK_STR}, + { -ENOTRECOVERABLE, CPE_EFAILED_STR}, + { -EINVAL, CPE_EBADPARAM_STR}, + { -ENOSYS, CPE_EUNSUPPORTED_STR}, + { -ENOPROTOOPT, CPE_EVERSION_STR}, + { -ENOTRECOVERABLE, CPE_EUNEXPECTED_STR}, + { -ENOTRECOVERABLE, CPE_EPANIC_STR}, + { -ENOSPC, CPE_ENORESOURCE_STR}, + { -EBADR, CPE_EHANDLE_STR}, + { -EALREADY, CPE_EALREADY_STR}, + { -EPERM, CPE_ENOTREADY_STR}, + { -EINPROGRESS, CPE_EPENDING_STR}, + { -EBUSY, CPE_EBUSY_STR}, + { -ECANCELED, CPE_EABORTED_STR}, + { -EAGAIN, CPE_EPREEMPTED_STR}, + { -EAGAIN, CPE_ECONTINUE_STR}, + { -EAGAIN, CPE_EIMMEDIATE_STR}, + { -EAGAIN, CPE_ENOTIMPL_STR}, + { -ENODATA, CPE_ENEEDMORE_STR}, + { -EADV, CPE_ERR_MAX_STR}, + { -ENOMEM, CPE_ENOMEMORY_STR}, + { -ENODEV, CPE_ENOTEXIST_STR}, + { -EADV, CPE_ETERMINATED_STR}, + { -EADV, CPE_ERR_MAX_STR}, +}; + +static inline int cpe_err_get_lnx_err_code(u32 cpe_error) +{ + if (cpe_error > CPE_ERR_MAX) + return cpe_err_code_info[CPE_ERR_MAX].lnx_err_code; + else + return cpe_err_code_info[cpe_error].lnx_err_code; +} + +static inline char *cpe_err_get_err_str(u32 cpe_error) +{ + if (cpe_error > CPE_ERR_MAX) + return cpe_err_code_info[CPE_ERR_MAX].cpe_err_str; + else + return cpe_err_code_info[cpe_error].cpe_err_str; +} + +#endif diff --git a/include/sound/info.h b/include/sound/info.h index 67390ee846aa..3f0ceb511bff 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -161,7 +161,9 @@ static inline void snd_info_set_text_ops(struct snd_info_entry *entry, } int snd_info_check_reserved_words(const char *str); - +struct snd_info_entry *snd_register_module_info(struct module *module, + const char *name, + struct snd_info_entry *parent); #else #define snd_seq_root NULL @@ -190,7 +192,9 @@ static inline void snd_info_set_text_ops(struct snd_info_entry *entry __attribut void *private_data, void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) {} static inline int snd_info_check_reserved_words(const char *str) { return 1; } - +static inline struct snd_info_entry *snd_register_module_info( + struct module *module, const char *name, + struct snd_info_entry *parent) { return NULL; } #endif /* diff --git a/include/sound/jack.h b/include/sound/jack.h index 23bede121c78..0d2a334fbeaa 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -58,14 +58,20 @@ enum snd_jack_types { SND_JACK_VIDEOOUT = 0x0010, SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, SND_JACK_LINEIN = 0x0020, + SND_JACK_OC_HPHL = 0x0040, + SND_JACK_OC_HPHR = 0x0080, + SND_JACK_UNSUPPORTED = 0x0100, + SND_JACK_MICROPHONE2 = 0x0200, + SND_JACK_ANC_HEADPHONE = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_MICROPHONE2, /* Kept separate from switches to facilitate implementation */ - SND_JACK_BTN_0 = 0x4000, - SND_JACK_BTN_1 = 0x2000, - SND_JACK_BTN_2 = 0x1000, - SND_JACK_BTN_3 = 0x0800, - SND_JACK_BTN_4 = 0x0400, - SND_JACK_BTN_5 = 0x0200, + SND_JACK_BTN_0 = 0x8000, + SND_JACK_BTN_1 = 0x4000, + SND_JACK_BTN_2 = 0x2000, + SND_JACK_BTN_3 = 0x1000, + SND_JACK_BTN_4 = 0x0800, + SND_JACK_BTN_5 = 0x0400, }; /* Keep in sync with definitions above */ diff --git a/include/sound/msm-audio-effects-q6-v2.h b/include/sound/msm-audio-effects-q6-v2.h new file mode 100644 index 000000000000..6bc2338bcf55 --- /dev/null +++ b/include/sound/msm-audio-effects-q6-v2.h @@ -0,0 +1,55 @@ +/* + * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef _MSM_AUDIO_EFFECTS_H +#define _MSM_AUDIO_EFFECTS_H + +#include <sound/audio_effects.h> + +#define MAX_PP_PARAMS_SZ 128 + +bool msm_audio_effects_is_effmodule_supp_in_top(int effect_module, + int topology); + +int msm_audio_effects_enable_extn(struct audio_client *ac, + struct msm_nt_eff_all_config *effects, + bool flag); + +int msm_audio_effects_reverb_handler(struct audio_client *ac, + struct reverb_params *reverb, + long *values); + +int msm_audio_effects_bass_boost_handler(struct audio_client *ac, + struct bass_boost_params *bass_boost, + long *values); + +int msm_audio_effects_pbe_handler(struct audio_client *ac, + struct pbe_params *pbe, + long *values); + +int msm_audio_effects_virtualizer_handler(struct audio_client *ac, + struct virtualizer_params *virtualizer, + long *values); + +int msm_audio_effects_popless_eq_handler(struct audio_client *ac, + struct eq_params *eq, + long *values); + +int msm_audio_effects_volume_handler(struct audio_client *ac, + struct soft_volume_params *vol, + long *values); + +int msm_audio_effects_volume_handler_v2(struct audio_client *ac, + struct soft_volume_params *vol, + long *values, int instance); +#endif /*_MSM_AUDIO_EFFECTS_H*/ diff --git a/include/sound/msm-dai-q6-v2.h b/include/sound/msm-dai-q6-v2.h new file mode 100644 index 000000000000..b1d76bf73f51 --- /dev/null +++ b/include/sound/msm-dai-q6-v2.h @@ -0,0 +1,92 @@ +/* Copyright (c) 2012-2016, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __MSM_DAI_Q6_PDATA_H__ + +#define __MSM_DAI_Q6_PDATA_H__ + +#define MSM_MI2S_SD0 (1 << 0) +#define MSM_MI2S_SD1 (1 << 1) +#define MSM_MI2S_SD2 (1 << 2) +#define MSM_MI2S_SD3 (1 << 3) +#define MSM_MI2S_CAP_RX 0 +#define MSM_MI2S_CAP_TX 1 + +#define MSM_PRIM_MI2S 0 +#define MSM_SEC_MI2S 1 +#define MSM_TERT_MI2S 2 +#define MSM_QUAT_MI2S 3 +#define MSM_SEC_MI2S_SD1 4 +#define MSM_QUIN_MI2S 5 +#define MSM_SENARY_MI2S 6 +#define MSM_INT0_MI2S 7 +#define MSM_INT1_MI2S 8 +#define MSM_INT2_MI2S 9 +#define MSM_INT3_MI2S 10 +#define MSM_INT4_MI2S 11 +#define MSM_INT5_MI2S 12 +#define MSM_INT6_MI2S 13 +#define MSM_MI2S_MIN MSM_PRIM_MI2S +#define MSM_MI2S_MAX MSM_INT6_MI2S + +struct msm_dai_auxpcm_config { + u16 mode; + u16 sync; + u16 frame; + u16 quant; + u16 num_slots; + u16 *slot_mapping; + u16 data; + u32 pcm_clk_rate; +}; + +struct msm_dai_auxpcm_pdata { + struct msm_dai_auxpcm_config mode_8k; + struct msm_dai_auxpcm_config mode_16k; +}; + +struct msm_mi2s_pdata { + u16 rx_sd_lines; + u16 tx_sd_lines; + u16 intf_id; +}; + +struct msm_i2s_data { + u32 capability; /* RX or TX */ + u16 sd_lines; +}; + +struct msm_dai_tdm_group_config { + u16 group_id; + u16 num_ports; + u16 *port_id; + u32 clk_rate; +}; + +struct msm_dai_tdm_config { + u16 sync_mode; + u16 sync_src; + u16 data_out; + u16 invert_sync; + u16 data_delay; + u32 data_align; + u16 header_start_offset; + u16 header_width; + u16 header_num_frame_repeat; +}; + +struct msm_dai_tdm_pdata { + struct msm_dai_tdm_group_config group_config; + struct msm_dai_tdm_config config; +}; + +#endif diff --git a/include/sound/msm-slim-dma.h b/include/sound/msm-slim-dma.h new file mode 100644 index 000000000000..6bbdbe563edc --- /dev/null +++ b/include/sound/msm-slim-dma.h @@ -0,0 +1,44 @@ +/* + * Copyright (c) 2014, 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 _MSM_SLIMBUS_DMA_H +#define _MSM_SLIMBUS_DMA_H + +#include <linux/slimbus/slimbus.h> + +/* + * struct msm_slim_dma_data - DMA data for slimbus data transfer + * + * @sdev: Handle to the slim_device instance associated with the + * data transfer. + * @ph: Port handle for the slimbus ports. + * @dai_channel_ctl: callback function into the CPU dai driver + * to setup the data path. + * + * This structure is used to share the slimbus port handles and + * other data path setup related handles with other drivers. + */ +struct msm_slim_dma_data { + + /* Handle to slimbus device */ + struct slim_device *sdev; + + /* Port Handle */ + u32 ph; + + /* Callback for data channel control */ + int (*dai_channel_ctl) (struct msm_slim_dma_data *dma_data, + struct snd_soc_dai *dai, bool enable); +}; + +#endif diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ffc161906d36..147e448ed405 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -68,6 +68,8 @@ struct snd_pcm_ops { int (*close)(struct snd_pcm_substream *substream); int (*ioctl)(struct snd_pcm_substream * substream, unsigned int cmd, void *arg); + int (*compat_ioctl)(struct snd_pcm_substream *substream, + unsigned int cmd, void *arg); int (*hw_params)(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); int (*hw_free)(struct snd_pcm_substream *substream); @@ -78,15 +80,19 @@ struct snd_pcm_ops { struct timespec *system_ts, struct timespec *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, struct snd_pcm_audio_tstamp_report *audio_tstamp_report); + int (*delay_blk)(struct snd_pcm_substream *substream); + int (*wall_clock)(struct snd_pcm_substream *substream, + struct timespec *audio_ts); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); - int (*silence)(struct snd_pcm_substream *substream, int channel, + int (*silence)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count); struct page *(*page)(struct snd_pcm_substream *substream, unsigned long offset); int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int (*ack)(struct snd_pcm_substream *substream); + int (*restart)(struct snd_pcm_substream *substream); }; /* @@ -115,6 +121,12 @@ struct snd_pcm_ops { #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) +#define SNDRV_DMA_MODE (0) +#define SNDRV_NON_DMA_MODE (1 << 0) +#define SNDRV_RENDER_STOPPED (1 << 1) +#define SNDRV_RENDER_RUNNING (1 << 2) + + /* If you change this don't forget to change rates[] table in pcm_native.c */ #define SNDRV_PCM_RATE_5512 (1<<0) /* 5512Hz */ #define SNDRV_PCM_RATE_8000 (1<<1) /* 8000Hz */ @@ -129,6 +141,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -141,6 +155,9 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) +#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\ + SNDRV_PCM_RATE_352800|\ + SNDRV_PCM_RATE_384000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) @@ -368,6 +385,7 @@ struct snd_pcm_runtime { unsigned int rate_num; unsigned int rate_den; unsigned int no_period_wakeup: 1; + unsigned int render_flag; /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ @@ -448,6 +466,7 @@ struct snd_pcm_substream { const struct snd_pcm_ops *ops; /* -- runtime information -- */ struct snd_pcm_runtime *runtime; + spinlock_t runtime_lock; /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ @@ -482,6 +501,7 @@ struct snd_pcm_substream { #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; + unsigned int hw_no_buffer: 1; /* substream may not have a buffer */ }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) @@ -507,6 +527,8 @@ struct snd_pcm_str { #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ + struct snd_kcontrol *vol_kctl; /* volume controls */ + struct snd_kcontrol *usr_kctl; /* user controls */ struct device dev; }; @@ -1399,6 +1421,54 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) return 1ULL << (__force int) pcm_format; } +/* + * PCM Volume control API + */ +/* array element of volume */ +struct snd_pcm_volume_elem { + int volume; +}; + +/* pp information; retrieved via snd_kcontrol_chip() */ +struct snd_pcm_volume { + struct snd_pcm *pcm; /* assigned PCM instance */ + int stream; /* PLAYBACK or CAPTURE */ + struct snd_kcontrol *kctl; + const struct snd_pcm_volume_elem *volume; + int max_length; + void *private_data; /* optional: private data pointer */ +}; + +int snd_pcm_add_volume_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_volume_elem *volume, + int max_length, + unsigned long private_value, + struct snd_pcm_volume **info_ret); + +/* + * PCM User control API + */ +/* array element of usr elem */ +struct snd_pcm_usr_elem { + int val[128]; +}; + +/* pp information; retrieved via snd_kcontrol_chip() */ +struct snd_pcm_usr { + struct snd_pcm *pcm; /* assigned PCM instance */ + int stream; /* PLAYBACK or CAPTURE */ + struct snd_kcontrol *kctl; + const struct snd_pcm_usr_elem *usr; + int max_length; + void *private_data; /* optional: private data pointer */ +}; + +int snd_pcm_add_usr_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_usr_elem *usr, + int max_length, int max_control_str_len, + unsigned long private_value, + struct snd_pcm_usr **info_ret); + /* printk helpers */ #define pcm_err(pcm, fmt, args...) \ dev_err((pcm)->card->dev, fmt, ##args) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 2af7bb3ee57d..91f6abfb2ce0 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -110,10 +110,14 @@ static inline void snd_mask_reset_range(struct snd_mask *mask, static inline void snd_mask_leave(struct snd_mask *mask, unsigned int val) { - unsigned int v; - v = mask->bits[MASK_OFS(val)] & MASK_BIT(val); - snd_mask_none(mask); - mask->bits[MASK_OFS(val)] = v; + unsigned int v, bits_index; + + bits_index = MASK_OFS(val); + if (bits_index < ((SNDRV_MASK_MAX+31)/32)) { + v = mask->bits[bits_index] & MASK_BIT(val); + snd_mask_none(mask); + mask->bits[bits_index] = v; + } } static inline void snd_mask_intersect(struct snd_mask *mask, diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h new file mode 100644 index 000000000000..f04daf310182 --- /dev/null +++ b/include/sound/q6adm-v2.h @@ -0,0 +1,227 @@ +/* Copyright (c) 2012-2018, 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 __Q6_ADM_V2_H__ +#define __Q6_ADM_V2_H__ + + +#define ADM_PATH_PLAYBACK 0x1 +#define ADM_PATH_LIVE_REC 0x2 +#define ADM_PATH_NONLIVE_REC 0x3 +#define ADM_PATH_COMPRESSED_RX 0x5 +#define ADM_PATH_COMPRESSED_TX 0x6 +#include <linux/qdsp6v2/rtac.h> +#include <sound/q6afe-v2.h> +#include <sound/q6audio-v2.h> + +#define MAX_MODULES_IN_TOPO 16 +#define ADM_GET_TOPO_MODULE_LIST_LENGTH\ + ((MAX_MODULES_IN_TOPO + 1) * sizeof(uint32_t)) +#define ADM_GET_TOPO_MODULE_INSTANCE_LIST_LENGTH \ + ((MAX_MODULES_IN_TOPO + 1) * 2 * sizeof(uint32_t)) +#define AUD_PROC_BLOCK_SIZE 4096 +#define AUD_VOL_BLOCK_SIZE 4096 +#define AUDIO_RX_CALIBRATION_SIZE (AUD_PROC_BLOCK_SIZE + \ + AUD_VOL_BLOCK_SIZE) +enum { + ADM_CUSTOM_TOP_CAL = 0, + ADM_AUDPROC_CAL, + ADM_AUDVOL_CAL, + ADM_RTAC_INFO_CAL, + ADM_RTAC_APR_CAL, + ADM_SRS_TRUMEDIA, + ADM_RTAC_AUDVOL_CAL, + ADM_MAX_CAL_TYPES +}; + +enum { + ADM_MEM_MAP_INDEX_SOURCE_TRACKING = ADM_MAX_CAL_TYPES, + ADM_MEM_MAP_INDEX_MAX +}; + +enum { + ADM_CLIENT_ID_DEFAULT = 0, + ADM_CLIENT_ID_SOURCE_TRACKING, + ADM_CLIENT_ID_MAX, +}; + +/* ENUM for adm_status & route_status */ +enum adm_status_flags { + ADM_STATUS_CALIBRATION_REQUIRED = 0, + ADM_STATUS_LIMITER, + ADM_STATUS_MAX, +}; + +#define MAX_COPPS_PER_PORT 0x8 +#define ADM_MAX_CHANNELS 32 + +/* multiple copp per stream. */ +struct route_payload { + unsigned int copp_idx[MAX_COPPS_PER_PORT]; + unsigned int port_id[MAX_COPPS_PER_PORT]; + int app_type[MAX_COPPS_PER_PORT]; + int acdb_dev_id[MAX_COPPS_PER_PORT]; + int sample_rate[MAX_COPPS_PER_PORT]; + unsigned long route_status[MAX_COPPS_PER_PORT]; + unsigned short num_copps; + unsigned int session_id; +}; + +struct default_chmixer_param_id_coeff { + uint32_t index; + uint16_t num_output_channels; + uint16_t num_input_channels; +}; + +struct msm_pcm_channel_mixer { + int output_channel; + int input_channels[ADM_MAX_CHANNELS]; + bool enable; + int rule; + int channel_weight[ADM_MAX_CHANNELS][ADM_MAX_CHANNELS]; + int port_idx; + int input_channel; + uint16_t in_ch_map[ADM_MAX_CHANNELS]; + uint16_t out_ch_map[ADM_MAX_CHANNELS]; + int override_cfg; +}; + +int srs_trumedia_open(int port_id, int copp_idx, __s32 srs_tech_id, + void *srs_params); + +int adm_dts_eagle_set(int port_id, int copp_idx, int param_id, + void *data, uint32_t size); + +int adm_dts_eagle_get(int port_id, int copp_idx, int param_id, + void *data, uint32_t size); + +void adm_copp_mfc_cfg(int port_id, int copp_idx, int dst_sample_rate); + +int adm_get_params(int port_id, int copp_idx, uint32_t module_id, + uint32_t param_id, uint32_t params_length, char *params); + +int adm_get_pp_params(int port_id, int copp_idx, uint32_t client_id, + struct mem_mapping_hdr *mem_hdr, + struct param_hdr_v3 *param_hdr, u8 *returned_param_data); + +int adm_send_params_v5(int port_id, int copp_idx, char *params, + uint32_t params_length); + +int adm_dolby_dap_send_params(int port_id, int copp_idx, char *params, + uint32_t params_length); + +int adm_set_pp_params(int port_id, int copp_idx, + struct mem_mapping_hdr *mem_hdr, u8 *param_data, + u32 params_size); + +int adm_pack_and_set_one_pp_param(int port_id, int copp_idx, + struct param_hdr_v3 param_hdr, + u8 *param_data); + +int adm_open(int port, int path, int rate, int mode, int topology, + int perf_mode, uint16_t bits_per_sample, + int app_type, int acdbdev_id, u32 copp_token); + +int adm_map_rtac_block(struct rtac_cal_block_data *cal_block); + +int adm_unmap_rtac_block(uint32_t *mem_map_handle); + +int adm_close(int port, int topology, int perf_mode); + +int adm_matrix_map(int path, struct route_payload payload_map, + int perf_mode, uint32_t passthr_mode); + +int adm_connect_afe_port(int mode, int session_id, int port_id); + +void adm_ec_ref_rx_id(int port_id); + +void adm_num_ec_ref_rx_chans(int num_chans); + +void adm_ec_ref_rx_bit_width(int bit_width); + +void adm_ec_ref_rx_sampling_rate(int sampling_rate); + +int adm_get_lowlatency_copp_id(int port_id); + +int adm_set_multi_ch_map(char *channel_map, int path); + +int adm_get_multi_ch_map(char *channel_map, int path); + +int adm_validate_and_get_port_index(int port_id); + +int adm_get_default_copp_idx(int port_id); + +int adm_get_topology_for_port_from_copp_id(int port_id, int copp_id); + +int adm_get_topology_for_port_copp_idx(int port_id, int copp_idx); + +int adm_get_indexes_from_copp_id(int copp_id, int *port_idx, int *copp_idx); + +int adm_set_pspd_matrix_params(int port_id, int copp_idx, + unsigned int session_id, + char *params, uint32_t params_length, + int session_type); + +int adm_set_downmix_params(int port_id, int copp_idx, + unsigned int session_id, char *params, + uint32_t params_length); + +int adm_get_pp_topo_module_list(int port_id, int copp_idx, int32_t param_length, + char *params); + +int adm_get_pp_topo_module_list_v2(int port_id, int copp_idx, + int32_t param_length, + int32_t *returned_params); + +int adm_set_volume(int port_id, int copp_idx, int volume); + +int adm_set_softvolume(int port_id, int copp_idx, + struct audproc_softvolume_params *softvol_param); + +int adm_set_mic_gain(int port_id, int copp_idx, int volume); + +int adm_send_set_multichannel_ec_primary_mic_ch(int port_id, int copp_idx, + int primary_mic_ch); + +int adm_param_enable(int port_id, int copp_idx, int module_id, int enable); + +int adm_param_enable_v2(int port_id, int copp_idx, + struct module_instance_info mod_inst_info, int enable); + +int adm_send_calibration(int port_id, int copp_idx, int path, int perf_mode, + int cal_type, char *params, int size); + +int adm_set_wait_parameters(int port_id, int copp_idx); + +int adm_reset_wait_parameters(int port_id, int copp_idx); + +int adm_wait_timeout(int port_id, int copp_idx, int wait_time); + +int adm_store_cal_data(int port_id, int copp_idx, int path, int perf_mode, + int cal_type, char *params, int *size); + +int adm_send_compressed_device_mute(int port_id, int copp_idx, bool mute_on); + +int adm_send_compressed_device_latency(int port_id, int copp_idx, int latency); +int adm_set_sound_focus(int port_id, int copp_idx, + struct sound_focus_param soundFocusData); +int adm_get_sound_focus(int port_id, int copp_idx, + struct sound_focus_param *soundFocusData); +int adm_get_source_tracking(int port_id, int copp_idx, + struct source_tracking_param *sourceTrackingData); +int adm_swap_speaker_channels(int port_id, int copp_idx, int sample_rate, + bool spk_swap); +int adm_programable_channel_mixer(int port_id, int copp_idx, int session_id, + int session_type, + struct msm_pcm_channel_mixer *ch_mixer, + int channel_index); +void adm_set_native_mode(int mode); +#endif /* __Q6_ADM_V2_H__ */ diff --git a/include/sound/q6afe-v2.h b/include/sound/q6afe-v2.h new file mode 100644 index 000000000000..e171028839f7 --- /dev/null +++ b/include/sound/q6afe-v2.h @@ -0,0 +1,534 @@ +/* Copyright (c) 2012-2018, 2020, 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 __Q6AFE_V2_H__ +#define __Q6AFE_V2_H__ +#include <sound/apr_audio-v2.h> +#include <linux/qdsp6v2/rtac.h> + +#define IN 0x000 +#define OUT 0x001 +#define MSM_AFE_MONO 0 +#define MSM_AFE_CH_STEREO 1 +#define MSM_AFE_MONO_RIGHT 1 +#define MSM_AFE_MONO_LEFT 2 +#define MSM_AFE_STEREO 3 +#define MSM_AFE_4CHANNELS 4 +#define MSM_AFE_6CHANNELS 6 +#define MSM_AFE_8CHANNELS 8 + +#define MSM_AFE_I2S_FORMAT_LPCM 0 +#define MSM_AFE_I2S_FORMAT_COMPR 1 +#define MSM_AFE_I2S_FORMAT_IEC60958_LPCM 2 +#define MSM_AFE_I2S_FORMAT_IEC60958_COMPR 3 + +#define MSM_AFE_PORT_TYPE_RX 0 +#define MSM_AFE_PORT_TYPE_TX 1 + +#define RT_PROXY_DAI_001_RX 0xE0 +#define RT_PROXY_DAI_001_TX 0xF0 +#define RT_PROXY_DAI_002_RX 0xF1 +#define RT_PROXY_DAI_002_TX 0xE1 +#define VIRTUAL_ID_TO_PORTID(val) ((val & 0xF) | 0x2000) + +#define AFE_CLK_VERSION_V1 1 +#define AFE_CLK_VERSION_V2 2 + +#define AFE_MAX_RDDMA 10 +#define AFE_MAX_WRDMA 10 + +typedef int (*routing_cb)(int port); + +enum { + /* IDX 0->4 */ + IDX_PRIMARY_I2S_RX, + IDX_PRIMARY_I2S_TX, + IDX_AFE_PORT_ID_PRIMARY_PCM_RX, + IDX_AFE_PORT_ID_PRIMARY_PCM_TX, + IDX_SECONDARY_I2S_RX, + /* IDX 5->9 */ + IDX_SECONDARY_I2S_TX, + IDX_MI2S_RX, + IDX_MI2S_TX, + IDX_HDMI_RX, + IDX_RSVD_2, + /* IDX 10->14 */ + IDX_RSVD_3, + IDX_DIGI_MIC_TX, + IDX_VOICE_RECORD_RX, + IDX_VOICE_RECORD_TX, + IDX_VOICE_PLAYBACK_TX, + /* IDX 15->19 */ + IDX_SLIMBUS_0_RX, + IDX_SLIMBUS_0_TX, + IDX_SLIMBUS_1_RX, + IDX_SLIMBUS_1_TX, + IDX_SLIMBUS_2_RX, + /* IDX 20->24 */ + IDX_SLIMBUS_2_TX, + IDX_SLIMBUS_3_RX, + IDX_SLIMBUS_3_TX, + IDX_SLIMBUS_4_RX, + IDX_SLIMBUS_4_TX, + /* IDX 25->29 */ + IDX_SLIMBUS_5_RX, + IDX_SLIMBUS_5_TX, + IDX_INT_BT_SCO_RX, + IDX_INT_BT_SCO_TX, + IDX_INT_BT_A2DP_RX, + /* IDX 30->34 */ + IDX_INT_FM_RX, + IDX_INT_FM_TX, + IDX_RT_PROXY_PORT_001_RX, + IDX_RT_PROXY_PORT_001_TX, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX, + /* IDX 35->39 */ + IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX, + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX, + IDX_AFE_PORT_ID_SECONDARY_MI2S_TX, + IDX_AFE_PORT_ID_TERTIARY_MI2S_RX, + IDX_AFE_PORT_ID_TERTIARY_MI2S_TX, + /* IDX 40->44 */ + IDX_AFE_PORT_ID_PRIMARY_MI2S_RX, + IDX_AFE_PORT_ID_PRIMARY_MI2S_TX, + IDX_AFE_PORT_ID_SECONDARY_PCM_RX, + IDX_AFE_PORT_ID_SECONDARY_PCM_TX, + IDX_VOICE2_PLAYBACK_TX, + /* IDX 45->49 */ + IDX_SLIMBUS_6_RX, + IDX_SLIMBUS_6_TX, + IDX_SPDIF_RX, + IDX_GLOBAL_CFG, + IDX_AUDIO_PORT_ID_I2S_RX, + /* IDX 50->53 */ + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_SD1, + IDX_AFE_PORT_ID_QUINARY_MI2S_RX, + IDX_AFE_PORT_ID_QUINARY_MI2S_TX, + IDX_AFE_PORT_ID_SENARY_MI2S_TX, + /* IDX 54->117 */ + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_0, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_0, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_1, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_1, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_2, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_2, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_3, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_3, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_4, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_4, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_5, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_5, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_6, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_6, + IDX_AFE_PORT_ID_PRIMARY_TDM_RX_7, + IDX_AFE_PORT_ID_PRIMARY_TDM_TX_7, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_0, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_0, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_1, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_1, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_2, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_2, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_3, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_3, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_4, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_4, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_5, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_5, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_6, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_6, + IDX_AFE_PORT_ID_SECONDARY_TDM_RX_7, + IDX_AFE_PORT_ID_SECONDARY_TDM_TX_7, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_0, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_0, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_1, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_1, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_2, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_2, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_3, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_3, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_4, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_4, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_5, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_5, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_6, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_6, + IDX_AFE_PORT_ID_TERTIARY_TDM_RX_7, + IDX_AFE_PORT_ID_TERTIARY_TDM_TX_7, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_0, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_0, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_1, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_1, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_2, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_2, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_3, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_3, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_4, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_4, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_5, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_5, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_6, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_6, + IDX_AFE_PORT_ID_QUATERNARY_TDM_RX_7, + IDX_AFE_PORT_ID_QUATERNARY_TDM_TX_7, + /* IDX 118->121 */ + IDX_SLIMBUS_7_RX, + IDX_SLIMBUS_7_TX, + IDX_SLIMBUS_8_RX, + IDX_SLIMBUS_8_TX, + /* IDX 122-> 123 */ + IDX_AFE_PORT_ID_USB_RX, + IDX_AFE_PORT_ID_USB_TX, + /* IDX 124 */ + IDX_DISPLAY_PORT_RX, + /* IDX 125-> 128 */ + IDX_AFE_PORT_ID_TERTIARY_PCM_RX, + IDX_AFE_PORT_ID_TERTIARY_PCM_TX, + IDX_AFE_PORT_ID_QUATERNARY_PCM_RX, + IDX_AFE_PORT_ID_QUATERNARY_PCM_TX, + /* IDX 129-> 142 */ + IDX_AFE_PORT_ID_INT0_MI2S_RX, + IDX_AFE_PORT_ID_INT0_MI2S_TX, + IDX_AFE_PORT_ID_INT1_MI2S_RX, + IDX_AFE_PORT_ID_INT1_MI2S_TX, + IDX_AFE_PORT_ID_INT2_MI2S_RX, + IDX_AFE_PORT_ID_INT2_MI2S_TX, + IDX_AFE_PORT_ID_INT3_MI2S_RX, + IDX_AFE_PORT_ID_INT3_MI2S_TX, + IDX_AFE_PORT_ID_INT4_MI2S_RX, + IDX_AFE_PORT_ID_INT4_MI2S_TX, + IDX_AFE_PORT_ID_INT5_MI2S_RX, + IDX_AFE_PORT_ID_INT5_MI2S_TX, + IDX_AFE_PORT_ID_INT6_MI2S_RX, + IDX_AFE_PORT_ID_INT6_MI2S_TX, + /* IDX 143 -> 150 */ + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_1, + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_2, + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_3, + IDX_AFE_PORT_ID_SECONDARY_MI2S_RX_4, + IDX_AFE_PORT_ID_SECONDARY_MI2S_TX_1, + IDX_AFE_PORT_ID_SECONDARY_MI2S_TX_2, + IDX_AFE_PORT_ID_SECONDARY_MI2S_TX_3, + IDX_AFE_PORT_ID_SECONDARY_MI2S_TX_4, + /* IDX 151 -> 158 */ + IDX_AFE_PORT_ID_TERTIARY_MI2S_RX_1, + IDX_AFE_PORT_ID_TERTIARY_MI2S_RX_2, + IDX_AFE_PORT_ID_TERTIARY_MI2S_RX_3, + IDX_AFE_PORT_ID_TERTIARY_MI2S_RX_4, + IDX_AFE_PORT_ID_TERTIARY_MI2S_TX_1, + IDX_AFE_PORT_ID_TERTIARY_MI2S_TX_2, + IDX_AFE_PORT_ID_TERTIARY_MI2S_TX_3, + IDX_AFE_PORT_ID_TERTIARY_MI2S_TX_4, + /* IDX 159 -> 166 */ + IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX_1, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX_2, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX_3, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_RX_4, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX_1, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX_2, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX_3, + IDX_AFE_PORT_ID_QUATERNARY_MI2S_TX_4, + /* IDX 167 -> 168 */ + IDX_RT_PROXY_PORT_002_RX, + IDX_RT_PROXY_PORT_002_TX, + AFE_MAX_PORTS +}; + + +enum { + IDX_PRIMARY_TDM_RX_0, + IDX_PRIMARY_TDM_RX_1, + IDX_PRIMARY_TDM_RX_2, + IDX_PRIMARY_TDM_RX_3, + IDX_PRIMARY_TDM_RX_4, + IDX_PRIMARY_TDM_RX_5, + IDX_PRIMARY_TDM_RX_6, + IDX_PRIMARY_TDM_RX_7, + IDX_PRIMARY_TDM_TX_0, + IDX_PRIMARY_TDM_TX_1, + IDX_PRIMARY_TDM_TX_2, + IDX_PRIMARY_TDM_TX_3, + IDX_PRIMARY_TDM_TX_4, + IDX_PRIMARY_TDM_TX_5, + IDX_PRIMARY_TDM_TX_6, + IDX_PRIMARY_TDM_TX_7, + IDX_SECONDARY_TDM_RX_0, + IDX_SECONDARY_TDM_RX_1, + IDX_SECONDARY_TDM_RX_2, + IDX_SECONDARY_TDM_RX_3, + IDX_SECONDARY_TDM_RX_4, + IDX_SECONDARY_TDM_RX_5, + IDX_SECONDARY_TDM_RX_6, + IDX_SECONDARY_TDM_RX_7, + IDX_SECONDARY_TDM_TX_0, + IDX_SECONDARY_TDM_TX_1, + IDX_SECONDARY_TDM_TX_2, + IDX_SECONDARY_TDM_TX_3, + IDX_SECONDARY_TDM_TX_4, + IDX_SECONDARY_TDM_TX_5, + IDX_SECONDARY_TDM_TX_6, + IDX_SECONDARY_TDM_TX_7, + IDX_TERTIARY_TDM_RX_0, + IDX_TERTIARY_TDM_RX_1, + IDX_TERTIARY_TDM_RX_2, + IDX_TERTIARY_TDM_RX_3, + IDX_TERTIARY_TDM_RX_4, + IDX_TERTIARY_TDM_RX_5, + IDX_TERTIARY_TDM_RX_6, + IDX_TERTIARY_TDM_RX_7, + IDX_TERTIARY_TDM_TX_0, + IDX_TERTIARY_TDM_TX_1, + IDX_TERTIARY_TDM_TX_2, + IDX_TERTIARY_TDM_TX_3, + IDX_TERTIARY_TDM_TX_4, + IDX_TERTIARY_TDM_TX_5, + IDX_TERTIARY_TDM_TX_6, + IDX_TERTIARY_TDM_TX_7, + IDX_QUATERNARY_TDM_RX_0, + IDX_QUATERNARY_TDM_RX_1, + IDX_QUATERNARY_TDM_RX_2, + IDX_QUATERNARY_TDM_RX_3, + IDX_QUATERNARY_TDM_RX_4, + IDX_QUATERNARY_TDM_RX_5, + IDX_QUATERNARY_TDM_RX_6, + IDX_QUATERNARY_TDM_RX_7, + IDX_QUATERNARY_TDM_TX_0, + IDX_QUATERNARY_TDM_TX_1, + IDX_QUATERNARY_TDM_TX_2, + IDX_QUATERNARY_TDM_TX_3, + IDX_QUATERNARY_TDM_TX_4, + IDX_QUATERNARY_TDM_TX_5, + IDX_QUATERNARY_TDM_TX_6, + IDX_QUATERNARY_TDM_TX_7, + IDX_TDM_MAX, +}; + +enum { + IDX_GROUP_PRIMARY_TDM_RX, + IDX_GROUP_PRIMARY_TDM_TX, + IDX_GROUP_SECONDARY_TDM_RX, + IDX_GROUP_SECONDARY_TDM_TX, + IDX_GROUP_TERTIARY_TDM_RX, + IDX_GROUP_TERTIARY_TDM_TX, + IDX_GROUP_QUATERNARY_TDM_RX, + IDX_GROUP_QUATERNARY_TDM_TX, + IDX_GROUP_TDM_MAX, +}; + +enum { + IDX_SECONDARY_MI2S_RX_1, + IDX_SECONDARY_MI2S_RX_2, + IDX_SECONDARY_MI2S_RX_3, + IDX_SECONDARY_MI2S_RX_4, + IDX_SECONDARY_MI2S_TX_1, + IDX_SECONDARY_MI2S_TX_2, + IDX_SECONDARY_MI2S_TX_3, + IDX_SECONDARY_MI2S_TX_4, + IDX_TERTIARY_MI2S_RX_1, + IDX_TERTIARY_MI2S_RX_2, + IDX_TERTIARY_MI2S_RX_3, + IDX_TERTIARY_MI2S_RX_4, + IDX_TERTIARY_MI2S_TX_1, + IDX_TERTIARY_MI2S_TX_2, + IDX_TERTIARY_MI2S_TX_3, + IDX_TERTIARY_MI2S_TX_4, + IDX_QUATERNARY_MI2S_RX_1, + IDX_QUATERNARY_MI2S_RX_2, + IDX_QUATERNARY_MI2S_RX_3, + IDX_QUATERNARY_MI2S_RX_4, + IDX_QUATERNARY_MI2S_TX_1, + IDX_QUATERNARY_MI2S_TX_2, + IDX_QUATERNARY_MI2S_TX_3, + IDX_QUATERNARY_MI2S_TX_4, + IDX_GROUP_MI2S_PORT_MAX, +}; + +enum { + IDX_GROUP_SECONDARY_MI2S_RX, + IDX_GROUP_SECONDARY_MI2S_TX, + IDX_GROUP_TERTIARY_MI2S_RX, + IDX_GROUP_TERTIARY_MI2S_TX, + IDX_GROUP_QUATERNARY_MI2S_RX, + IDX_GROUP_QUATERNARY_MI2S_TX, + IDX_GROUP_MI2S_MAX, +}; + +enum afe_mad_type { + MAD_HW_NONE = 0x00, + MAD_HW_AUDIO = 0x01, + MAD_HW_BEACON = 0x02, + MAD_HW_ULTRASOUND = 0x04, + MAD_SW_AUDIO = 0x05, +}; + +enum afe_cal_mode { + AFE_CAL_MODE_DEFAULT = 0x00, + AFE_CAL_MODE_NONE, +}; + +struct afe_audio_buffer { + dma_addr_t phys; + void *data; + uint32_t used; + uint32_t size;/* size of buffer */ + uint32_t actual_size; /* actual number of bytes read by DSP */ + struct ion_handle *handle; + struct ion_client *client; +}; + +struct afe_audio_port_data { + struct afe_audio_buffer *buf; + uint32_t max_buf_cnt; + uint32_t dsp_buf; + uint32_t cpu_buf; + struct list_head mem_map_handle; + uint32_t tmp_hdl; + /* read or write locks */ + struct mutex lock; + spinlock_t dsp_lock; +}; + +struct afe_audio_client { + atomic_t cmd_state; + /* Relative or absolute TS */ + uint32_t time_flag; + void *priv; + uint64_t time_stamp; + struct mutex cmd_lock; + /* idx:1 out port, 0: in port*/ + struct afe_audio_port_data port[2]; + wait_queue_head_t cmd_wait; + uint32_t mem_map_handle; +}; + +struct aanc_data { + bool aanc_active; + uint16_t aanc_rx_port; + uint16_t aanc_tx_port; + uint32_t aanc_rx_port_sample_rate; + uint32_t aanc_tx_port_sample_rate; +}; + +int afe_open(u16 port_id, union afe_port_config *afe_config, int rate); +int afe_close(int port_id); +int afe_loopback(u16 enable, u16 rx_port, u16 tx_port); +int afe_sidetone_enable(u16 tx_port_id, u16 rx_port_id, bool enable); +int afe_loopback_gain(u16 port_id, u16 volume); +int afe_validate_port(u16 port_id); +int afe_get_port_index(u16 port_id); +int afe_get_topology(int port_id); +int afe_start_pseudo_port(u16 port_id); +int afe_stop_pseudo_port(u16 port_id); +uint32_t afe_req_mmap_handle(struct afe_audio_client *ac); +int afe_memory_map(phys_addr_t dma_addr_p, u32 dma_buf_sz, + struct afe_audio_client *ac); +int afe_cmd_memory_map(phys_addr_t dma_addr_p, u32 dma_buf_sz); +int afe_cmd_memory_map_nowait(int port_id, phys_addr_t dma_addr_p, + u32 dma_buf_sz); +int afe_cmd_memory_unmap(u32 dma_addr_p); +int afe_cmd_memory_unmap_nowait(u32 dma_addr_p); +void afe_set_dtmf_gen_rx_portid(u16 rx_port_id, int set); +int afe_dtmf_generate_rx(int64_t duration_in_ms, + uint16_t high_freq, + uint16_t low_freq, uint16_t gain); +int afe_register_get_events(u16 port_id, + void (*cb) (uint32_t opcode, + uint32_t token, uint32_t *payload, void *priv), + void *private_data); +int afe_unregister_get_events(u16 port_id); +int afe_rt_proxy_port_write(phys_addr_t buf_addr_p, + u32 mem_map_handle, int bytes); +int afe_rt_proxy_port_read(phys_addr_t buf_addr_p, + u32 mem_map_handle, int bytes); +void afe_set_cal_mode(u16 port_id, enum afe_cal_mode afe_cal_mode); +int afe_port_start(u16 port_id, union afe_port_config *afe_config, + u32 rate); +int afe_port_start_v2(u16 port_id, union afe_port_config *afe_config, + u32 rate, u16 afe_in_channels, u16 afe_in_bit_width, + struct afe_enc_config *enc_config); +int afe_spk_prot_feed_back_cfg(int src_port, int dst_port, + int l_ch, int r_ch, u32 enable); +int afe_spk_prot_get_calib_data(struct afe_spkr_prot_get_vi_calib *calib); +int afe_port_stop_nowait(int port_id); +int afe_apply_gain(u16 port_id, u16 gain); +int afe_q6_interface_prepare(void); +int afe_get_port_type(u16 port_id); +int q6afe_audio_client_buf_alloc_contiguous(unsigned int dir, + struct afe_audio_client *ac, + unsigned int bufsz, + unsigned int bufcnt); +struct afe_audio_client *q6afe_audio_client_alloc(void *priv); +int q6afe_audio_client_buf_free_contiguous(unsigned int dir, + struct afe_audio_client *ac); +void q6afe_audio_client_free(struct afe_audio_client *ac); +/* if port_id is virtual, convert to physical.. + * if port_id is already physical, return physical + */ +int afe_convert_virtual_to_portid(u16 port_id); + +int afe_pseudo_port_start_nowait(u16 port_id); +int afe_pseudo_port_stop_nowait(u16 port_id); +int afe_set_lpass_clock(u16 port_id, struct afe_clk_cfg *cfg); +int afe_set_lpass_clock_v2(u16 port_id, struct afe_clk_set *cfg); +int afe_set_lpass_clk_cfg(int index, struct afe_clk_set *cfg); +int afe_set_digital_codec_core_clock(u16 port_id, + struct afe_digital_clk_cfg *cfg); +int afe_set_lpass_internal_digital_codec_clock(u16 port_id, + struct afe_digital_clk_cfg *cfg); +int afe_enable_lpass_core_shared_clock(u16 port_id, u32 enable); + +int q6afe_check_osr_clk_freq(u32 freq); + +int afe_send_spdif_clk_cfg(struct afe_param_id_spdif_clk_cfg *cfg, + u16 port_id); +int afe_send_spdif_ch_status_cfg(struct afe_param_id_spdif_ch_status_cfg + *ch_status_cfg, u16 port_id); + +int afe_spdif_port_start(u16 port_id, struct afe_spdif_port_config *spdif_port, + u32 rate); + +int afe_turn_onoff_hw_mad(u16 mad_type, u16 mad_enable); +int afe_port_set_mad_type(u16 port_id, enum afe_mad_type mad_type); +enum afe_mad_type afe_port_get_mad_type(u16 port_id); +int afe_set_config(enum afe_config_type config_type, void *config_data, + int arg); +void afe_clear_config(enum afe_config_type config); +bool afe_has_config(enum afe_config_type config); + +void afe_set_aanc_info(struct aanc_data *aanc_info); +int afe_port_group_set_param(u16 group_id, + union afe_port_group_config *afe_group_config); +int afe_port_group_enable(u16 group_id, + union afe_port_group_config *afe_group_config, u16 enable); +int afe_unmap_rtac_block(uint32_t *mem_map_handle); +int afe_map_rtac_block(struct rtac_cal_block_data *cal_block); +int afe_send_slot_mapping_cfg( + struct afe_param_id_slot_mapping_cfg *slot_mapping_cfg, + u16 port_id); +int afe_send_custom_tdm_header_cfg( + struct afe_param_id_custom_tdm_header_cfg *custom_tdm_header_cfg, + u16 port_id); +int afe_tdm_port_start(u16 port_id, struct afe_tdm_port_config *tdm_port, + u32 rate, u16 num_groups); +void afe_set_routing_callback(routing_cb); +int afe_get_av_dev_drift(struct afe_param_id_dev_timing_stats *timing_stats, + u16 port); +int afe_get_svc_version(uint32_t service_id); +int afe_request_dma_resources(uint8_t dma_type, uint8_t num_read_dma_channels, + uint8_t num_write_dma_channels); +int afe_get_dma_idx(bool **ret_rddma_idx, + bool **ret_wrdma_idx); +int afe_release_all_dma_resources(void); +int afe_i2s_port_start(u16 port_id, struct afe_i2s_port_config *i2s_port, + u32 rate, u16 num_groups); +int afe_port_group_mi2s_enable(u16 group_id, + union afe_port_group_mi2s_config *afe_group_config, + u16 enable); +#endif /* __Q6AFE_V2_H__ */ diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h new file mode 100644 index 000000000000..9a6c1c8eefd2 --- /dev/null +++ b/include/sound/q6asm-v2.h @@ -0,0 +1,741 @@ +/* Copyright (c) 2012-2018, 2020 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 __Q6_ASM_V2_H__ +#define __Q6_ASM_V2_H__ + +#include <linux/qdsp6v2/apr.h> +#include <linux/qdsp6v2/rtac.h> +#include <sound/apr_audio-v2.h> +#include <linux/list.h> +#include <linux/msm_ion.h> + +#define IN 0x000 +#define OUT 0x001 +#define CH_MODE_MONO 0x001 +#define CH_MODE_STEREO 0x002 + +#define FORMAT_LINEAR_PCM 0x0000 +#define FORMAT_DTMF 0x0001 +#define FORMAT_ADPCM 0x0002 +#define FORMAT_YADPCM 0x0003 +#define FORMAT_MP3 0x0004 +#define FORMAT_MPEG4_AAC 0x0005 +#define FORMAT_AMRNB 0x0006 +#define FORMAT_AMRWB 0x0007 +#define FORMAT_V13K 0x0008 +#define FORMAT_EVRC 0x0009 +#define FORMAT_EVRCB 0x000a +#define FORMAT_EVRCWB 0x000b +#define FORMAT_MIDI 0x000c +#define FORMAT_SBC 0x000d +#define FORMAT_WMA_V10PRO 0x000e +#define FORMAT_WMA_V9 0x000f +#define FORMAT_AMR_WB_PLUS 0x0010 +#define FORMAT_MPEG4_MULTI_AAC 0x0011 +#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012 +#define FORMAT_AC3 0x0013 +#define FORMAT_EAC3 0x0014 +#define FORMAT_MP2 0x0015 +#define FORMAT_FLAC 0x0016 +#define FORMAT_ALAC 0x0017 +#define FORMAT_VORBIS 0x0018 +#define FORMAT_APE 0x0019 +#define FORMAT_G711_ALAW_FS 0x001a +#define FORMAT_G711_MLAW_FS 0x001b +#define FORMAT_DTS 0x001c +#define FORMAT_DSD 0x001d +#define FORMAT_APTX 0x001e +#define FORMAT_GEN_COMPR 0x001f +#define FORMAT_TRUEHD 0x0020 +#define FORMAT_IEC61937 0x0021 +#define FORMAT_APTXHD 0x0022 + +#define ENCDEC_SBCBITRATE 0x0001 +#define ENCDEC_IMMEDIATE_DECODE 0x0002 +#define ENCDEC_CFG_BLK 0x0003 + +#define CMD_PAUSE 0x0001 +#define CMD_FLUSH 0x0002 +#define CMD_EOS 0x0003 +#define CMD_CLOSE 0x0004 +#define CMD_OUT_FLUSH 0x0005 +#define CMD_SUSPEND 0x0006 + +/* bit 0:1 represents priority of stream */ +#define STREAM_PRIORITY_NORMAL 0x0000 +#define STREAM_PRIORITY_LOW 0x0001 +#define STREAM_PRIORITY_HIGH 0x0002 + +/* bit 4 represents META enable of encoded data buffer */ +#define BUFFER_META_ENABLE 0x0010 + +/* bit 5 represents timestamp */ +/* bit 5 - 0 -- ASM_DATA_EVENT_READ_DONE will have relative time-stamp*/ +/* bit 5 - 1 -- ASM_DATA_EVENT_READ_DONE will have absolute time-stamp*/ +#define ABSOLUTE_TIMESTAMP_ENABLE 0x0020 + +/* Enable Sample_Rate/Channel_Mode notification event from Decoder */ +#define SR_CM_NOTIFY_ENABLE 0x0004 + +#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */ +#define TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ +#define SYNC_IO_MODE 0x0001 +#define ASYNC_IO_MODE 0x0002 +#define COMPRESSED_IO 0x0040 +#define COMPRESSED_STREAM_IO 0x0080 +#define NT_MODE 0x0400 + +#define NO_TIMESTAMP 0xFF00 +#define SET_TIMESTAMP 0x0000 + +#define SOFT_PAUSE_ENABLE 1 +#define SOFT_PAUSE_DISABLE 0 + +#define ASM_ACTIVE_STREAMS_ALLOWED 0xF +/* Control session is used for mapping calibration memory */ +#define ASM_CONTROL_SESSION (ASM_ACTIVE_STREAMS_ALLOWED + 1) + +#define ASM_SHIFT_GAPLESS_MODE_FLAG 31 +#define ASM_SHIFT_LAST_BUFFER_FLAG 30 + +#define ASM_LITTLE_ENDIAN 0 +#define ASM_BIG_ENDIAN 1 + +/* PCM_MEDIA_FORMAT_Version */ +enum { + PCM_MEDIA_FORMAT_V2 = 0, + PCM_MEDIA_FORMAT_V3, + PCM_MEDIA_FORMAT_V4, + PCM_MEDIA_FORMAT_V5, +}; + +/* PCM format modes in DSP */ +enum { + DEFAULT_QF = 0, + Q15 = 15, + Q23 = 23, + Q31 = 31, +}; + +/* payload structure bytes */ +#define READDONE_IDX_STATUS 0 +#define READDONE_IDX_BUFADD_LSW 1 +#define READDONE_IDX_BUFADD_MSW 2 +#define READDONE_IDX_MEMMAP_HDL 3 +#define READDONE_IDX_SIZE 4 +#define READDONE_IDX_OFFSET 5 +#define READDONE_IDX_LSW_TS 6 +#define READDONE_IDX_MSW_TS 7 +#define READDONE_IDX_FLAGS 8 +#define READDONE_IDX_NUMFRAMES 9 +#define READDONE_IDX_SEQ_ID 10 + +#define SOFT_PAUSE_PERIOD 30 /* ramp up/down for 30ms */ +#define SOFT_PAUSE_STEP 0 /* Step value 0ms or 0us */ +enum { + SOFT_PAUSE_CURVE_LINEAR = 0, + SOFT_PAUSE_CURVE_EXP, + SOFT_PAUSE_CURVE_LOG, +}; + +#define SOFT_VOLUME_PERIOD 30 /* ramp up/down for 30ms */ +#define SOFT_VOLUME_STEP 0 /* Step value 0ms or 0us */ +enum { + SOFT_VOLUME_CURVE_LINEAR = 0, + SOFT_VOLUME_CURVE_EXP, + SOFT_VOLUME_CURVE_LOG, +}; + +#define SOFT_VOLUME_INSTANCE_1 1 +#define SOFT_VOLUME_INSTANCE_2 2 + +typedef void (*app_cb)(uint32_t opcode, uint32_t token, + uint32_t *payload, void *priv); + +struct audio_buffer { + dma_addr_t phys; + void *data; + uint32_t used; + uint32_t size;/* size of buffer */ + uint32_t actual_size; /* actual number of bytes read by DSP */ + struct ion_handle *handle; + struct ion_client *client; +}; + +struct audio_aio_write_param { + phys_addr_t paddr; + uint32_t len; + uint32_t uid; + uint32_t lsw_ts; + uint32_t msw_ts; + uint32_t flags; + uint32_t metadata_len; + uint32_t last_buffer; +}; + +struct audio_aio_read_param { + phys_addr_t paddr; + uint32_t len; + uint32_t uid; + uint32_t flags;/*meta data flags*/ +}; + +struct audio_port_data { + struct audio_buffer *buf; + uint32_t max_buf_cnt; + uint32_t dsp_buf; + uint32_t cpu_buf; + struct list_head mem_map_handle; + uint32_t tmp_hdl; + /* read or write locks */ + struct mutex lock; + spinlock_t dsp_lock; +}; + +struct shared_io_config { + uint32_t format; + uint16_t bits_per_sample; + uint32_t rate; + uint32_t channels; + uint16_t sample_word_size; + uint32_t bufsz; + uint32_t bufcnt; +}; + +struct audio_client { + int session; + app_cb cb; + atomic_t cmd_state; + atomic_t cmd_state_pp; + /* Relative or absolute TS */ + atomic_t time_flag; + atomic_t nowait_cmd_cnt; + atomic_t mem_state; + void *priv; + uint32_t io_mode; + uint64_t time_stamp; + struct apr_svc *apr; + struct apr_svc *mmap_apr; + struct apr_svc *apr2; + struct mutex cmd_lock; + /* idx:1 out port, 0: in port*/ + struct audio_port_data port[2]; + wait_queue_head_t cmd_wait; + wait_queue_head_t time_wait; + wait_queue_head_t mem_wait; + int perf_mode; + int stream_id; + struct device *dev; + int topology; + int app_type; + /* audio cache operations fptr*/ + int (*fptr_cache_ops)(struct audio_buffer *abuff, int cache_op); + atomic_t unmap_cb_success; + atomic_t reset; + /* holds latest DSP pipeline delay */ + uint32_t path_delay; + /* shared io */ + struct audio_buffer shared_pos_buf; + struct shared_io_config config; +}; + +void q6asm_audio_client_free(struct audio_client *ac); + +struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv); + +struct audio_client *q6asm_get_audio_client(int session_id); + +int q6asm_audio_client_buf_alloc(unsigned int dir/* 1:Out,0:In */, + struct audio_client *ac, + unsigned int bufsz, + uint32_t bufcnt); +int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir + /* 1:Out,0:In */, + struct audio_client *ac, + unsigned int bufsz, + unsigned int bufcnt); + +int q6asm_audio_client_buf_free_contiguous(unsigned int dir, + struct audio_client *ac); + +int q6asm_set_pp_params(struct audio_client *ac, + struct mem_mapping_hdr *mem_hdr, u8 *param_data, + u32 param_size); + +int q6asm_pack_and_set_pp_param_in_band(struct audio_client *ac, + struct param_hdr_v3 param_hdr, + u8 *param_data); + +int q6asm_set_soft_volume_module_instance_ids(int instance, + struct param_hdr_v3 *param_hdr); + +int q6asm_open_read(struct audio_client *ac, uint32_t format + /*, uint16_t bits_per_sample*/); + +int q6asm_open_read_v2(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_read_v3(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_read_v4(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, bool ts_mode); + +int q6asm_open_read_v5(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, bool ts_mode); + +int q6asm_open_read_with_retry(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, bool ts_mode); + +int q6asm_open_write(struct audio_client *ac, uint32_t format + /*, uint16_t bits_per_sample*/); + +int q6asm_open_write_v2(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_shared_io(struct audio_client *ac, + struct shared_io_config *c, int dir); + +int q6asm_open_write_v3(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_write_v4(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_write_v5(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_open_write_with_retry(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); + +int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, int32_t stream_id, + bool is_gapless_mode); + +int q6asm_stream_open_write_v3(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, int32_t stream_id, + bool is_gapless_mode); + +int q6asm_stream_open_write_v4(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, int32_t stream_id, + bool is_gapless_mode); + +int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format, + uint32_t passthrough_flag); + +int q6asm_open_read_write(struct audio_client *ac, + uint32_t rd_format, + uint32_t wr_format); + +int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format, + uint32_t wr_format, bool is_meta_data_mode, + uint32_t bits_per_sample, bool overwrite_topology, + int topology); + +int q6asm_open_loopback_v2(struct audio_client *ac, + uint16_t bits_per_sample); + +int q6asm_open_loopback_with_retry(struct audio_client *ac, + uint16_t bits_per_sample); + +int q6asm_open_transcode_loopback(struct audio_client *ac, + uint16_t bits_per_sample, uint32_t source_format, + uint32_t sink_format); + +int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts, + uint32_t lsw_ts, uint32_t flags); +int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts, + uint32_t lsw_ts, uint32_t flags); + +int q6asm_async_write(struct audio_client *ac, + struct audio_aio_write_param *param); + +int q6asm_async_read(struct audio_client *ac, + struct audio_aio_read_param *param); + +int q6asm_read(struct audio_client *ac); +int q6asm_read_v2(struct audio_client *ac, uint32_t len); +int q6asm_read_nolock(struct audio_client *ac); + +int q6asm_memory_map(struct audio_client *ac, phys_addr_t buf_add, + int dir, uint32_t bufsz, uint32_t bufcnt); + +int q6asm_memory_unmap(struct audio_client *ac, phys_addr_t buf_add, + int dir); + +struct audio_buffer *q6asm_shared_io_buf(struct audio_client *ac, int dir); + +int q6asm_shared_io_free(struct audio_client *ac, int dir); + +int q6asm_get_shared_pos(struct audio_client *ac, uint32_t *si, uint32_t *msw, + uint32_t *lsw); + +int q6asm_map_rtac_block(struct rtac_cal_block_data *cal_block); + +int q6asm_unmap_rtac_block(uint32_t *mem_map_handle); + +int q6asm_send_cal(struct audio_client *ac); + +int q6asm_run(struct audio_client *ac, uint32_t flags, + uint32_t msw_ts, uint32_t lsw_ts); + +int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, + uint32_t msw_ts, uint32_t lsw_ts); + +int q6asm_stream_run_nowait(struct audio_client *ac, uint32_t flags, + uint32_t msw_ts, uint32_t lsw_ts, uint32_t stream_id); + +int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable); + +int q6asm_reg_rx_underflow(struct audio_client *ac, uint16_t enable); + +int q6asm_cmd(struct audio_client *ac, int cmd); + +int q6asm_stream_cmd(struct audio_client *ac, int cmd, uint32_t stream_id); + +int q6asm_cmd_nowait(struct audio_client *ac, int cmd); + +int q6asm_stream_cmd_nowait(struct audio_client *ac, int cmd, + uint32_t stream_id); + +void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac, + uint32_t *size, uint32_t *idx); + +int q6asm_cpu_buf_release(int dir, struct audio_client *ac); + +void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac, + uint32_t *size, uint32_t *idx); + +int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac); + +/* File format specific configurations to be added below */ + +int q6asm_enc_cfg_blk_aac(struct audio_client *ac, + uint32_t frames_per_buf, + uint32_t sample_rate, uint32_t channels, + uint32_t bit_rate, + uint32_t mode, uint32_t format); + +int q6asm_enc_cfg_blk_g711(struct audio_client *ac, + uint32_t frames_per_buf, + uint32_t sample_rate); + +int q6asm_enc_cfg_blk_pcm(struct audio_client *ac, + uint32_t rate, uint32_t channels); + +int q6asm_enc_cfg_blk_pcm_v2(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, + bool use_default_chmap, bool use_back_flavor, + u8 *channel_map); + +int q6asm_enc_cfg_blk_pcm_v3(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, bool use_default_chmap, + bool use_back_flavor, u8 *channel_map, + uint16_t sample_word_size); + +int q6asm_enc_cfg_blk_pcm_v4(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, bool use_default_chmap, + bool use_back_flavor, u8 *channel_map, + uint16_t sample_word_size, uint16_t endianness, + uint16_t mode); + +int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample); + +int q6asm_enc_cfg_blk_pcm_format_support_v3(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, + uint16_t sample_word_size); + +int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, + uint16_t sample_word_size, + uint16_t endianness, + uint16_t mode); + +int q6asm_enc_cfg_blk_pcm_format_support_v5(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, + uint16_t sample_word_size, + uint16_t endianness, + uint16_t mode); + +int q6asm_set_encdec_chan_map(struct audio_client *ac, + uint32_t num_channels); + +int q6asm_enc_cfg_blk_pcm_native(struct audio_client *ac, + uint32_t rate, uint32_t channels); + +int q6asm_enable_sbrps(struct audio_client *ac, + uint32_t sbr_ps); + +int q6asm_cfg_dual_mono_aac(struct audio_client *ac, + uint16_t sce_left, uint16_t sce_right); + +int q6asm_cfg_aac_sel_mix_coef(struct audio_client *ac, uint32_t mix_coeff); + +int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf, + uint16_t min_rate, uint16_t max_rate, + uint16_t reduced_rate_level, uint16_t rate_modulation_cmd); + +int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf, + uint16_t min_rate, uint16_t max_rate, + uint16_t rate_modulation_cmd); + +int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf, + uint16_t band_mode, uint16_t dtx_enable); + +int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf, + uint16_t band_mode, uint16_t dtx_enable); + +int q6asm_media_format_block_pcm(struct audio_client *ac, + uint32_t rate, uint32_t channels); + +int q6asm_media_format_block_pcm_format_support(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample); + +int q6asm_media_format_block_pcm_format_support_v2(struct audio_client *ac, + uint32_t rate, uint32_t channels, + uint16_t bits_per_sample, int stream_id, + bool use_default_chmap, char *channel_map); + +int q6asm_media_format_block_pcm_format_support_v3(struct audio_client *ac, + uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample, + int stream_id, + bool use_default_chmap, + char *channel_map, + uint16_t sample_word_size); + +int q6asm_media_format_block_pcm_format_support_v4(struct audio_client *ac, + uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample, + int stream_id, + bool use_default_chmap, + char *channel_map, + uint16_t sample_word_size, + uint16_t endianness, + uint16_t mode); + +int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, char *channel_map); + +int q6asm_media_format_block_multi_ch_pcm_v2( + struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, char *channel_map, + uint16_t bits_per_sample); +int q6asm_media_format_block_gen_compr( + struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, char *channel_map, + uint16_t bits_per_sample); + +int q6asm_media_format_block_iec( + struct audio_client *ac, + uint32_t rate, uint32_t channels); + +int q6asm_media_format_block_multi_ch_pcm_v3(struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, + char *channel_map, + uint16_t bits_per_sample, + uint16_t sample_word_size); + +int q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, + char *channel_map, + uint16_t bits_per_sample, + uint16_t sample_word_size, + uint16_t endianness, + uint16_t mode); + +int q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac, + uint32_t rate, uint32_t channels, + bool use_default_chmap, + char *channel_map, + uint16_t bits_per_sample, + uint16_t sample_word_size, + uint16_t endianness, + uint16_t mode); + +int q6asm_media_format_block_aac(struct audio_client *ac, + struct asm_aac_cfg *cfg); + +int q6asm_stream_media_format_block_aac(struct audio_client *ac, + struct asm_aac_cfg *cfg, int stream_id); + +int q6asm_media_format_block_multi_aac(struct audio_client *ac, + struct asm_aac_cfg *cfg); + +int q6asm_media_format_block_wma(struct audio_client *ac, + void *cfg, int stream_id); + +int q6asm_media_format_block_wmapro(struct audio_client *ac, + void *cfg, int stream_id); + +int q6asm_media_format_block_amrwbplus(struct audio_client *ac, + struct asm_amrwbplus_cfg *cfg); + +int q6asm_stream_media_format_block_flac(struct audio_client *ac, + struct asm_flac_cfg *cfg, int stream_id); + +int q6asm_media_format_block_alac(struct audio_client *ac, + struct asm_alac_cfg *cfg, int stream_id); + +int q6asm_media_format_block_g711(struct audio_client *ac, + struct asm_g711_dec_cfg *cfg, int stream_id); + +int q6asm_stream_media_format_block_vorbis(struct audio_client *ac, + struct asm_vorbis_cfg *cfg, int stream_id); + +int q6asm_media_format_block_ape(struct audio_client *ac, + struct asm_ape_cfg *cfg, int stream_id); + +int q6asm_media_format_block_dsd(struct audio_client *ac, + struct asm_dsd_cfg *cfg, int stream_id); + +int q6asm_stream_media_format_block_aptx_dec(struct audio_client *ac, + uint32_t sr, int stream_id); + +int q6asm_ds1_set_endp_params(struct audio_client *ac, + int param_id, int param_value); + +/* Send stream based end params */ +int q6asm_ds1_set_stream_endp_params(struct audio_client *ac, int param_id, + int param_value, int stream_id); + +/* PP specific */ +int q6asm_equalizer(struct audio_client *ac, void *eq); + +/* Send Volume Command */ +int q6asm_set_volume(struct audio_client *ac, int volume); + +/* Send Volume Command */ +int q6asm_set_volume_v2(struct audio_client *ac, int volume, int instance); + +/* DTS Eagle Params */ +int q6asm_dts_eagle_set(struct audio_client *ac, int param_id, uint32_t size, + void *data, struct param_outband *po, int m_id); +int q6asm_dts_eagle_get(struct audio_client *ac, int param_id, uint32_t size, + void *data, struct param_outband *po, int m_id); + +/* Send aptx decoder BT address */ +int q6asm_set_aptx_dec_bt_addr(struct audio_client *ac, + struct aptx_dec_bt_addr_cfg *cfg); + +/* Set SoftPause Params */ +int q6asm_set_softpause(struct audio_client *ac, + struct asm_softpause_params *param); + +/* Set Softvolume Params */ +int q6asm_set_softvolume(struct audio_client *ac, + struct asm_softvolume_params *param); + +/* Set Softvolume Params */ +int q6asm_set_softvolume_v2(struct audio_client *ac, + struct asm_softvolume_params *param, int instance); + +/* Set panning and MFC params */ +int q6asm_set_mfc_panning_params(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param); + +/* Set vol gain pair */ +int q6asm_set_vol_ctrl_gain_pair(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param); + +/* Send left-right channel gain */ +int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain); + +/* Send multi channel gain */ +int q6asm_set_multich_gain(struct audio_client *ac, uint32_t channels, + uint32_t *gains, uint8_t *ch_map, bool use_default); + +/* Enable Mute/unmute flag */ +int q6asm_set_mute(struct audio_client *ac, int muteflag); + +int q6asm_get_session_time(struct audio_client *ac, uint64_t *tstamp); + +int q6asm_get_session_time_legacy(struct audio_client *ac, uint64_t *tstamp); + +int q6asm_send_audio_effects_params(struct audio_client *ac, char *params, + uint32_t params_length); + +int q6asm_send_stream_cmd(struct audio_client *ac, + struct msm_adsp_event_data *data); + +int q6asm_audio_map_shm_fd(struct audio_client *ac, struct ion_client **client, + struct ion_handle **handle, int fd); + +int q6asm_send_rtic_event_ack(struct audio_client *ac, + void *param, uint32_t params_length); + +/* Client can set the IO mode to either AIO/SIO mode */ +int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode); + +/* Get Service ID for APR communication */ +int q6asm_get_apr_service_id(int session_id); + +/* Common format block without any payload +*/ +int q6asm_media_format_block(struct audio_client *ac, uint32_t format); + +/* Send the meta data to remove initial and trailing silence */ +int q6asm_send_meta_data(struct audio_client *ac, uint32_t initial_samples, + uint32_t trailing_samples); + +/* Send the stream meta data to remove initial and trailing silence */ +int q6asm_stream_send_meta_data(struct audio_client *ac, uint32_t stream_id, + uint32_t initial_samples, uint32_t trailing_samples); + +int q6asm_get_asm_topology(int session_id); +int q6asm_get_asm_app_type(int session_id); + +int q6asm_send_mtmx_strtr_window(struct audio_client *ac, + struct asm_session_mtmx_strtr_param_window_v2_t *window_param, + uint32_t param_id); + +/* Configure DSP render mode */ +int q6asm_send_mtmx_strtr_render_mode(struct audio_client *ac, + uint32_t render_mode); + +/* Configure DSP clock recovery mode */ +int q6asm_send_mtmx_strtr_clk_rec_mode(struct audio_client *ac, + uint32_t clk_rec_mode); + +/* Enable adjust session clock in DSP */ +int q6asm_send_mtmx_strtr_enable_adjust_session_clock(struct audio_client *ac, + bool enable); + +/* Retrieve the current DSP path delay */ +int q6asm_get_path_delay(struct audio_client *ac); + +/* Helper functions to retrieve data from token */ +uint8_t q6asm_get_buf_index_from_token(uint32_t token); +uint8_t q6asm_get_stream_id_from_token(uint32_t token); + +/* Adjust session clock in DSP */ +int q6asm_adjust_session_clock(struct audio_client *ac, + uint32_t adjust_time_lsw, + uint32_t adjust_time_msw); +int q6asm_get_svc_version(uint32_t service_id); +#endif /* __Q6_ASM_H__ */ diff --git a/include/sound/q6audio-v2.h b/include/sound/q6audio-v2.h new file mode 100644 index 000000000000..fd14f330d1d5 --- /dev/null +++ b/include/sound/q6audio-v2.h @@ -0,0 +1,36 @@ +/* Copyright (c) 2012-2013, 2015 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 _Q6_AUDIO_H_ +#define _Q6_AUDIO_H_ + +#include <linux/qdsp6v2/apr.h> + +enum { + LEGACY_PCM_MODE = 0, + LOW_LATENCY_PCM_MODE, + ULTRA_LOW_LATENCY_PCM_MODE, + ULL_POST_PROCESSING_PCM_MODE, +}; + + +int q6audio_get_port_index(u16 port_id); + +int q6audio_convert_virtual_to_portid(u16 port_id); + +int q6audio_validate_port(u16 port_id); + +int q6audio_is_digital_pcm_interface(u16 port_id); + +int q6audio_get_port_id(u16 port_id); + +#endif diff --git a/include/sound/q6common.h b/include/sound/q6common.h new file mode 100644 index 000000000000..b6208f756cd9 --- /dev/null +++ b/include/sound/q6common.h @@ -0,0 +1,23 @@ +/* Copyright (c) 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 __Q6COMMON_H__ +#define __Q6COMMON_H__ + +#include <sound/apr_audio-v2.h> + +void q6common_update_instance_id_support(bool supported); +bool q6common_is_instance_id_supported(void); +int q6common_pack_pp_params(u8 *dest, struct param_hdr_v3 *v3_hdr, + u8 *param_data, u32 *total_size); + +#endif /* __Q6COMMON_H__ */ diff --git a/include/sound/q6core.h b/include/sound/q6core.h new file mode 100644 index 000000000000..773fc45a6734 --- /dev/null +++ b/include/sound/q6core.h @@ -0,0 +1,201 @@ +/* Copyright (c) 2012-2019, 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 __Q6CORE_H__ +#define __Q6CORE_H__ +#include <linux/qdsp6v2/apr.h> +#include <sound/apr_audio-v2.h> + + + +#define AVCS_CMD_ADSP_EVENT_GET_STATE 0x0001290C +#define AVCS_CMDRSP_ADSP_EVENT_GET_STATE 0x0001290D +#define AVCS_SERVICES_AND_STATIC_MODULES_READY 0x1 +#define AVCS_SERVICE_AND_ALL_MODULES_READY 0x5 + +int q6core_is_adsp_ready(void); +int q6core_add_remove_pool_pages(phys_addr_t buf_add, uint32_t bufsz, + uint32_t mempool_id, bool add_pages); + +int q6core_get_service_version(uint32_t service_id, + struct avcs_fwk_ver_info *ver_info, + size_t size); +size_t q6core_get_fwk_version_size(uint32_t service_id); + +#define ADSP_CMD_SET_DTS_EAGLE_DATA_ID 0x00012919 +#define DTS_EAGLE_LICENSE_ID 0x00028346 +struct adsp_dts_eagle { + struct apr_hdr hdr; + uint32_t id; + uint32_t overwrite; + uint32_t size; + char data[]; +}; +int core_dts_eagle_set(int size, char *data); +int core_dts_eagle_get(int id, int size, char *data); + +#define ADSP_CMD_SET_DOLBY_MANUFACTURER_ID 0x00012918 + +struct adsp_dolby_manufacturer_id { + struct apr_hdr hdr; + int manufacturer_id; +}; + +uint32_t core_set_dolby_manufacturer_id(int manufacturer_id); + +/* Dolby Surround1 Module License ID. This ID is used as an identifier + for DS1 license via ADSP generic license mechanism. + Please refer AVCS_CMD_SET_LICENSE for more details. +*/ +#define DOLBY_DS1_LICENSE_ID 0x00000001 + +#define AVCS_CMD_SET_LICENSE 0x00012919 +struct avcs_cmd_set_license { + struct apr_hdr hdr; + uint32_t id; /**< A unique ID used to refer to this license */ + uint32_t overwrite; + /**< 0 = do not overwrite an existing license with this id. + 1 = overwrite an existing license with this id. */ + uint32_t size; + /**< Size in bytes of the license data following this header. */ + /* uint8_t* data , data and padding follows this structure + total packet size needs to be multiple of 4 Bytes*/ + +}; + +#define AVCS_CMD_GET_LICENSE_VALIDATION_RESULT 0x0001291A +struct avcs_cmd_get_license_validation_result { + struct apr_hdr hdr; + uint32_t id; /**< A unique ID used to refer to this license */ +}; + +#define AVCS_CMDRSP_GET_LICENSE_VALIDATION_RESULT 0x0001291B +struct avcs_cmdrsp_get_license_validation_result { + uint32_t result; + /* ADSP_EOK if the license validation result was successfully retrieved. + ADSP_ENOTEXIST if there is no license with the given id. + ADSP_ENOTIMPL if there is no validation function for a license + with this id. */ + uint32_t size; + /* Length in bytes of the result that follows this structure*/ +}; + +/* Set Q6 topologies */ +/* + * Registers custom topologies in the aDSP for + * use in audio, voice, AFE and LSM. + */ + + +#define AVCS_CMD_SHARED_MEM_MAP_REGIONS 0x00012924 +#define AVCS_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00012925 +#define AVCS_CMD_SHARED_MEM_UNMAP_REGIONS 0x00012926 + + +#define AVCS_CMD_REGISTER_TOPOLOGIES 0x00012923 + +/* The payload for the AVCS_CMD_REGISTER_TOPOLOGIES command */ +struct avcs_cmd_register_topologies { + struct apr_hdr hdr; + uint32_t payload_addr_lsw; + /* Lower 32 bits of the topology buffer address. */ + + uint32_t payload_addr_msw; + /* Upper 32 bits of the topology buffer address. */ + + uint32_t mem_map_handle; + /* Unique identifier for an address. + * -This memory map handle is returned by the aDSP through the + * memory map command. + * -NULL mem_map_handle is interpreted as in-band parameter + * passing. + * -Client has the flexibility to choose in-band or out-of-band. + * -Out-of-band is recommended in this case. + */ + + uint32_t payload_size; + /* Size in bytes of the valid data in the topology buffer. */ +} __packed; + + +#define AVCS_CMD_DEREGISTER_TOPOLOGIES 0x0001292a + +/* The payload for the AVCS_CMD_DEREGISTER_TOPOLOGIES command */ +struct avcs_cmd_deregister_topologies { + struct apr_hdr hdr; + uint32_t payload_addr_lsw; + /* Lower 32 bits of the topology buffer address. */ + + uint32_t payload_addr_msw; + /* Upper 32 bits of the topology buffer address. */ + + uint32_t mem_map_handle; + /* Unique identifier for an address. + * -This memory map handle is returned by the aDSP through the + * memory map command. + * -NULL mem_map_handle is interpreted as in-band parameter + * passing. + * -Client has the flexibility to choose in-band or out-of-band. + * -Out-of-band is recommended in this case. + */ + + uint32_t payload_size; + /* Size in bytes of the valid data in the topology buffer. */ + + uint32_t mode; + /* 1: Deregister selected topologies + * 2: Deregister all topologies + */ +} __packed; + +#define AVCS_MODE_DEREGISTER_ALL_CUSTOM_TOPOLOGIES 2 + + +int32_t core_set_license(uint32_t key, uint32_t module_id); +int32_t core_get_license_status(uint32_t module_id); + +#define ADSP_MEMORY_MAP_HLOS_PHYSPOOL 4 +#define AVCS_CMD_ADD_POOL_PAGES 0x0001292E +#define AVCS_CMD_REMOVE_POOL_PAGES 0x0001292F + +struct avs_mem_assign_region { + struct apr_hdr hdr; + u32 pool_id; + u32 size; + u32 addr_lsw; + u32 addr_msw; +} __packed; + +#define AVCS_GET_VERSIONS 0x00012905 +struct avcs_cmd_get_version_result { + struct apr_hdr hdr; + uint32_t id; +}; +#define AVCS_GET_VERSIONS_RSP 0x00012906 + +#define AVCS_CMDRSP_Q6_ID_2_6 0x00040000 +#define AVCS_CMDRSP_Q6_ID_2_7 0x00040001 +#define AVCS_CMDRSP_Q6_ID_2_8 0x00040002 +#define AVCS_CMDRSP_Q6_ID_2_9 0x00040003 + +enum q6_subsys_image { + Q6_SUBSYS_AVS2_6 = 1, + Q6_SUBSYS_AVS2_7, + Q6_SUBSYS_AVS2_8, + Q6_SUBSYS_AVS2_9, + Q6_SUBSYS_INVALID, +}; + +enum q6_subsys_image q6core_get_avs_version(void); + +int core_get_adsp_ver(void); +#endif /* __Q6CORE_H__ */ diff --git a/include/sound/q6lsm.h b/include/sound/q6lsm.h new file mode 100644 index 000000000000..4600b0445955 --- /dev/null +++ b/include/sound/q6lsm.h @@ -0,0 +1,248 @@ +/* + * Copyright (c) 2013-2017, 2019 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 __Q6LSM_H__ +#define __Q6LSM_H__ + +#include <linux/list.h> +#include <linux/msm_ion.h> +#include <sound/apr_audio-v2.h> +#include <sound/lsm_params.h> +#include <linux/qdsp6v2/apr.h> + +#define MAX_NUM_CONFIDENCE 20 + +#define ADM_LSM_PORT_ID 0xADCB + +#define LSM_MAX_NUM_CHANNELS 8 + +typedef void (*lsm_app_cb)(uint32_t opcode, uint32_t token, + uint32_t *payload, uint16_t client_size, void *priv); + +struct lsm_sound_model { + dma_addr_t phys; + void *data; + size_t size; /* size of buffer */ + uint32_t actual_size; /* actual number of bytes read by DSP */ + struct ion_handle *handle; + struct ion_client *client; + uint32_t mem_map_handle; +}; + +struct snd_lsm_event_status_v2 { + uint16_t status; + uint16_t payload_size; + uint8_t confidence_value[0]; +}; + +struct lsm_lab_buffer { + dma_addr_t phys; + void *data; + size_t size; + struct ion_handle *handle; + struct ion_client *client; + uint32_t mem_map_handle; +}; + +struct lsm_hw_params { + u16 sample_rate; + u16 sample_size; + u32 buf_sz; + u32 period_count; + u16 num_chs; +}; + +struct lsm_client { + int session; + lsm_app_cb cb; + atomic_t cmd_state; + void *priv; + struct apr_svc *apr; + struct apr_svc *mmap_apr; + struct mutex cmd_lock; + struct lsm_sound_model sound_model; + wait_queue_head_t cmd_wait; + uint32_t cmd_err_code; + uint16_t mode; + uint16_t connect_to_port; + uint8_t num_confidence_levels; + uint8_t *confidence_levels; + bool opened; + bool started; + dma_addr_t lsm_cal_phy_addr; + uint32_t lsm_cal_size; + uint32_t app_id; + bool lab_enable; + bool lab_started; + struct lsm_lab_buffer *lab_buffer; + struct lsm_hw_params hw_params; + bool use_topology; + int session_state; + bool poll_enable; + int perf_mode; + uint32_t event_mode; +}; + +struct lsm_stream_cmd_open_tx { + struct apr_hdr hdr; + uint16_t app_id; + uint16_t reserved; + uint32_t sampling_rate; +} __packed; + +struct lsm_stream_cmd_open_tx_v2 { + struct apr_hdr hdr; + uint32_t topology_id; +} __packed; + +struct lsm_custom_topologies { + struct apr_hdr hdr; + uint32_t data_payload_addr_lsw; + uint32_t data_payload_addr_msw; + uint32_t mem_map_handle; + uint32_t buffer_size; +} __packed; + +struct lsm_session_cmd_set_params_v2 { + struct apr_hdr apr_hdr; + uint32_t payload_size; + struct mem_mapping_hdr mem_hdr; + u32 param_data[0]; +} __packed; + +struct lsm_session_cmd_set_params_v3 { + struct apr_hdr apr_hdr; + struct mem_mapping_hdr mem_hdr; + uint32_t payload_size; + u32 param_data[0]; +} __packed; + +struct lsm_param_op_mode { + uint32_t minor_version; + uint16_t mode; + uint16_t reserved; +} __packed; + +struct lsm_param_connect_to_port { + uint32_t minor_version; + /* AFE port id that receives voice wake up data */ + uint16_t port_id; + uint16_t reserved; +} __packed; + +struct lsm_param_poll_enable { + uint32_t minor_version; + /* indicates to voice wakeup that HW MAD/SW polling is enabled or not */ + uint32_t polling_enable; +} __packed; + +struct lsm_param_fwk_mode_cfg { + uint32_t minor_version; + uint32_t mode; +} __packed; + +struct lsm_param_media_fmt { + uint32_t minor_version; + uint32_t sample_rate; + uint16_t num_channels; + uint16_t bit_width; + uint8_t channel_mapping[LSM_MAX_NUM_CHANNELS]; +} __packed; + +struct lsm_param_confidence_levels { + uint8_t num_confidence_levels; + uint8_t confidence_levels[0]; +} __packed; + +struct lsm_param_epd_thres { + uint32_t minor_version; + uint32_t epd_begin; + uint32_t epd_end; +} __packed; + +struct lsm_param_gain { + uint32_t minor_version; + uint16_t gain; + uint16_t reserved; +} __packed; + +struct lsm_cmd_reg_snd_model { + struct apr_hdr hdr; + uint32_t model_size; + uint32_t model_addr_lsw; + uint32_t model_addr_msw; + uint32_t mem_map_handle; +} __packed; + +struct lsm_param_lab_enable { + uint16_t enable; + uint16_t reserved; +} __packed; + +struct lsm_param_lab_config { + uint32_t minor_version; + uint32_t wake_up_latency_ms; +} __packed; + +struct lsm_cmd_read { + struct apr_hdr hdr; + uint32_t buf_addr_lsw; + uint32_t buf_addr_msw; + uint32_t mem_map_handle; + uint32_t buf_size; +} __packed; + +struct lsm_cmd_read_done { + struct apr_hdr hdr; + uint32_t status; + uint32_t buf_addr_lsw; + uint32_t buf_addr_msw; + uint32_t mem_map_handle; + uint32_t total_size; + uint32_t offset; + uint32_t timestamp_lsw; + uint32_t timestamp_msw; + uint32_t flags; +} __packed; + +struct lsm_client *q6lsm_client_alloc(lsm_app_cb cb, void *priv); +void q6lsm_client_free(struct lsm_client *client); +int q6lsm_open(struct lsm_client *client, uint16_t app_id); +int q6lsm_start(struct lsm_client *client, bool wait); +int q6lsm_stop(struct lsm_client *client, bool wait); +int q6lsm_snd_model_buf_alloc(struct lsm_client *client, size_t len, + bool allocate_module_data); +int q6lsm_snd_model_buf_free(struct lsm_client *client); +int q6lsm_close(struct lsm_client *client); +int q6lsm_register_sound_model(struct lsm_client *client, + enum lsm_detection_mode mode, + bool detectfailure); +int q6lsm_set_data(struct lsm_client *client, + enum lsm_detection_mode mode, + bool detectfailure); +int q6lsm_deregister_sound_model(struct lsm_client *client); +void set_lsm_port(int); +int get_lsm_port(void); +int q6lsm_lab_control(struct lsm_client *client, u32 enable); +int q6lsm_stop_lab(struct lsm_client *client); +int q6lsm_read(struct lsm_client *client, struct lsm_cmd_read *read); +int q6lsm_lab_buffer_alloc(struct lsm_client *client, bool alloc); +int q6lsm_set_one_param(struct lsm_client *client, + struct lsm_params_info *p_info, void *data, + uint32_t param_type); +void q6lsm_sm_set_param_data(struct lsm_client *client, + struct lsm_params_info *p_info, + size_t *offset); +int q6lsm_set_port_connected(struct lsm_client *client); +int q6lsm_set_fwk_mode_cfg(struct lsm_client *client, uint32_t event_mode); +int q6lsm_set_media_fmt_params(struct lsm_client *client); +#endif /* __Q6LSM_H__ */ diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index fb36e8a706fb..afffa756357a 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -79,6 +79,7 @@ struct snd_rawmidi_runtime { int buffer_ref; /* buffer reference count */ /* misc */ spinlock_t lock; + struct mutex realloc_mutex; wait_queue_head_t sleep; /* event handler (new bytes, input only) */ void (*event)(struct snd_rawmidi_substream *substream); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 212eaaf172ed..4cbe6a37d121 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -138,6 +138,10 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); + int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); struct snd_soc_dai_ops { @@ -166,6 +170,9 @@ struct snd_soc_dai_ops { unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); + int (*get_channel_map)(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); /* * DAI digital mute - optional. @@ -262,8 +269,8 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active:1; /* stream is in use */ - unsigned int playback_active:1; /* stream is in use */ + unsigned int capture_active; /* stream is in use */ + unsigned int playback_active; /* stream is in use */ unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2fc28324351d..4e1931b7c7bf 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -314,6 +314,11 @@ struct device; .get = snd_soc_dapm_get_pin_switch, \ .put = snd_soc_dapm_put_pin_switch, \ .private_value = (unsigned long)xname } +#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ +{ .id = snd_soc_dapm_micbias, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .event = wevent, .event_flags = wflags} /* dapm stream operations */ #define SND_SOC_DAPM_STREAM_NOP 0x0 @@ -453,6 +458,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_widget_list *dapm_kcontrol_get_wlist( + const struct snd_kcontrol *kcontrol); struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 806059052bfc..2ed3a25233c1 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -17,6 +17,7 @@ struct snd_soc_pcm_runtime; +#define DPCM_MAX_BE_USERS 8 /* * Types of runtime_update to perform. e.g. originated from FE PCM ops * or audio route changes triggered by muxes/mixers. @@ -86,6 +87,7 @@ struct snd_soc_dpcm { #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_state; #endif + int stream; }; /* @@ -148,8 +150,13 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_fe_dai_hw_params_be(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, struct snd_pcm_hw_params *hw_params, + int stream); int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_fe_dai_prepare_be(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event); diff --git a/include/sound/soc.h b/include/sound/soc.h index fb955e69a78e..229c23815eff 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -22,6 +22,7 @@ #include <linux/kernel.h> #include <linux/regmap.h> #include <linux/log2.h> +#include <linux/async.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/compress_driver.h> @@ -224,6 +225,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_SINGLE_MULTI_EXT(xname, xreg, xshift, xmax, xinvert, xcount,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_multi_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_multi_mixer_control) \ + {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \ + .count = xcount, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -354,6 +363,10 @@ #define SND_SOC_COMP_ORDER_LATE 1 #define SND_SOC_COMP_ORDER_LAST 2 +/* DAI Link Host Mode Support */ +#define SND_SOC_DAI_LINK_NO_HOST 0x1 +#define SND_SOC_DAI_LINK_OPT_HOST 0x2 + /* * Bias levels * @@ -545,12 +558,13 @@ int snd_soc_update_bits_locked(struct snd_soc_codec *codec, int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); +void snd_soc_card_change_online_state(struct snd_soc_card *soc_card, + int online); #ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec); struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_codec(struct snd_ac97 *ac97); - int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); @@ -636,6 +650,8 @@ int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_multi_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection @@ -727,6 +743,7 @@ struct snd_soc_pcm_stream { unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ unsigned int sig_bits; /* number of bits of content */ + const char *aif_name; /* DAPM AIF widget name */ }; /* SoC audio ops */ @@ -925,6 +942,16 @@ struct snd_soc_platform_driver { snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *); + /* + * For platform-caused delay reporting, where the thread blocks waiting + * for the delay amount to be determined. Defining this will cause the + * ASoC core to skip calling the delay callbacks for all components in + * the runtime. + * Optional. + */ + snd_pcm_sframes_t (*delay_blk)(struct snd_pcm_substream *, + struct snd_soc_dai *); + /* platform stream pcm ops */ const struct snd_pcm_ops *ops; @@ -949,6 +976,14 @@ struct snd_soc_platform { struct snd_soc_component component; }; +enum snd_soc_async_ops { + ASYNC_DPCM_SND_SOC_OPEN = 1 << 0, + ASYNC_DPCM_SND_SOC_CLOSE = 1 << 1, + ASYNC_DPCM_SND_SOC_PREPARE = 1 << 2, + ASYNC_DPCM_SND_SOC_HW_PARAMS = 1 << 3, + ASYNC_DPCM_SND_SOC_FREE = 1 << 4, +}; + struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ @@ -1028,6 +1063,9 @@ struct snd_soc_dai_link { /* This DAI link can route to other DAI links at runtime (Frontend)*/ unsigned int dynamic:1; + /* This DAI can support no host IO (no pcm data is copied to from host) */ + unsigned int no_host_mode:2; + /* DPCM capture and Playback support */ unsigned int dpcm_capture:1; unsigned int dpcm_playback:1; @@ -1037,6 +1075,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + + /* this value determines what all ops can be started asynchronously */ + enum snd_soc_async_ops async_ops; }; struct snd_soc_codec_conf { @@ -1079,6 +1120,7 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + struct mutex dapm_power_mutex; bool instantiated; @@ -1184,6 +1226,8 @@ struct snd_soc_pcm_runtime { long pmdown_time; unsigned char pop_wait:1; + /* err in case of ops failed */ + int err_ops; /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; @@ -1235,6 +1279,11 @@ struct soc_mreg_control { unsigned int regbase, regcount, nbits, invert; }; +struct soc_multi_mixer_control { + int min, max, platform_max, count; + unsigned int reg, rreg, shift, rshift, invert; +}; + /* enumerated kcontrol */ struct soc_enum { int reg; @@ -1420,6 +1469,8 @@ int snd_soc_component_update_bits_async(struct snd_soc_component *component, void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name); #ifdef CONFIG_REGMAP diff --git a/include/sound/wcd-dsp-mgr.h b/include/sound/wcd-dsp-mgr.h new file mode 100644 index 000000000000..2beb9b38a46a --- /dev/null +++ b/include/sound/wcd-dsp-mgr.h @@ -0,0 +1,136 @@ +/* + * Copyright (c) 2016, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __WCD_DSP_MGR_H__ +#define __WCD_DSP_MGR_H__ + +#include <linux/types.h> + +/* + * These enums correspond to the component types + * that wcd-dsp-manager driver will use. The order + * of the enums specifies the order in which the + * manager driver will perform the sequencing. + * Changing this will cause the sequencing order + * to be changed as well. + */ +enum wdsp_cmpnt_type { + /* Component to control the DSP */ + WDSP_CMPNT_CONTROL = 0, + /* Component to perform data transfer to/from DSP */ + WDSP_CMPNT_TRANSPORT, + /* Component that performs high level IPC */ + WDSP_CMPNT_IPC, + + WDSP_CMPNT_TYPE_MAX, +}; + +enum wdsp_event_type { + /* Initialization related */ + WDSP_EVENT_POST_INIT, + + /* Image download related */ + WDSP_EVENT_PRE_DLOAD_CODE, + WDSP_EVENT_DLOAD_SECTION, + WDSP_EVENT_POST_DLOAD_CODE, + WDSP_EVENT_PRE_DLOAD_DATA, + WDSP_EVENT_POST_DLOAD_DATA, + WDSP_EVENT_DLOAD_FAILED, + + WDSP_EVENT_READ_SECTION, + + /* DSP boot related */ + WDSP_EVENT_PRE_BOOTUP, + WDSP_EVENT_DO_BOOT, + WDSP_EVENT_POST_BOOTUP, + WDSP_EVENT_PRE_SHUTDOWN, + WDSP_EVENT_DO_SHUTDOWN, + WDSP_EVENT_POST_SHUTDOWN, + + /* IRQ handling related */ + WDSP_EVENT_IPC1_INTR, + + /* Suspend/Resume related */ + WDSP_EVENT_SUSPEND, + WDSP_EVENT_RESUME, +}; + +enum wdsp_signal { + /* Hardware generated interrupts signalled to manager */ + WDSP_IPC1_INTR, + WDSP_ERR_INTR, + + /* Other signals */ + WDSP_CDC_DOWN_SIGNAL, + WDSP_CDC_UP_SIGNAL, +}; + +/* + * wdsp_cmpnt_ops: ops/function callbacks for components + * @init: called by manager driver, component is expected + * to initialize itself in this callback + * @deinit: called by manager driver, component should + * de-initialize itself in this callback + * @event_handler: Event handler for each component, called + * by the manager as per sequence + */ +struct wdsp_cmpnt_ops { + int (*init)(struct device *, void *priv_data); + int (*deinit)(struct device *, void *priv_data); + int (*event_handler)(struct device *, void *priv_data, + enum wdsp_event_type, void *data); +}; + +struct wdsp_img_section { + u32 addr; + size_t size; + u8 *data; +}; + +struct wdsp_err_signal_arg { + bool mem_dumps_enabled; + u32 remote_start_addr; + size_t dump_size; +}; + +/* + * wdsp_ops: ops/function callbacks for manager driver + * @register_cmpnt_ops: components will use this to register + * their own ops to manager driver + * @get_dev_for_cmpnt: components can use this to get handle + * to struct device * of any other component + * @signal_handler: callback to notify manager driver that signal + * has occurred. Cannot be called from interrupt + * context as this can sleep + * @vote_for_dsp: notifies manager that dsp should be booted up + * @suspend: notifies manager that one component wants to suspend. + * Manager will make sure to suspend all components in order + * @resume: notifies manager that one component wants to resume. + * Manager will make sure to resume all components in order + */ + +struct wdsp_mgr_ops { + int (*register_cmpnt_ops)(struct device *wdsp_dev, + struct device *cdev, + void *priv_data, + struct wdsp_cmpnt_ops *ops); + struct device *(*get_dev_for_cmpnt)(struct device *wdsp_dev, + enum wdsp_cmpnt_type type); + int (*signal_handler)(struct device *wdsp_dev, + enum wdsp_signal signal, void *arg); + int (*vote_for_dsp)(struct device *wdsp_dev, bool vote); + int (*suspend)(struct device *wdsp_dev); + int (*resume)(struct device *wdsp_dev); +}; + +#endif /* end of __WCD_DSP_MGR_H__ */ diff --git a/include/sound/wcd-spi.h b/include/sound/wcd-spi.h new file mode 100644 index 000000000000..1fff58d727a1 --- /dev/null +++ b/include/sound/wcd-spi.h @@ -0,0 +1,57 @@ +/* + * Copyright (c) 2016, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __WCD_SPI_H__ +#define __WCD_SPI_H__ + +struct wcd_spi_msg { + /* + * Caller's buffer pointer that holds data to + * be transmitted in case of data_write and + * data to be copied to in case of data_read. + */ + void *data; + + /* Length of data to write/read */ + size_t len; + + /* + * Address in remote memory to write to + * or read from. + */ + u32 remote_addr; + + /* Bitmask of flags, currently unused */ + u32 flags; +}; + +#ifdef CONFIG_SND_SOC_WCD_SPI + +int wcd_spi_data_write(struct spi_device *spi, struct wcd_spi_msg *msg); +int wcd_spi_data_read(struct spi_device *spi, struct wcd_spi_msg *msg); + +#else + +int wcd_spi_data_write(struct spi_device *spi, struct wcd_spi_msg *msg) +{ + return -ENODEV; +} + +int wcd_spi_data_read(struct spi_device *spi, struct wcd_spi_msg *msg) +{ + return -ENODEV; +} + +#endif /* End of CONFIG_SND_SOC_WCD_SPI */ + +#endif /* End of __WCD_SPI_H__ */ |