audio-lnx: Initial change for techpack of audio drivers.
Add snapshot for audio drivers for SDM targets. The code is
migrated from msm-4.9 kernel at the below cutoff -
(74ff856e8d6: "net: ipc_router: Add dynamic enable/disable
wakeup source feature")
This changes are done for new techpack addition
for audio kernel. Migrate all audio kernel drivers
to this techpack.
Change-Id: I33d580af3ba86a5cb777583efc5d4cdaf2882d93
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
new file mode 100644
index 0000000..14f6445
--- /dev/null
+++ b/include/sound/apr_audio-v2.h
@@ -0,0 +1,10683 @@
+/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#ifndef _APR_AUDIO_V2_H_
+#define _APR_AUDIO_V2_H_
+
+#include <linux/qdsp6v2/apr.h>
+#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;
+};
+
+#define ADSP_ADM_VERSION 0x00070000
+
+#define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322
+#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323
+#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324
+
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D
+/* Enumeration for an audio Rx matrix ID.*/
+#define ADM_MATRIX_ID_AUDIO_RX 0
+
+#define ADM_MATRIX_ID_AUDIO_TX 1
+
+#define ADM_MATRIX_ID_COMPRESSED_AUDIO_RX 2
+
+#define ADM_MATRIX_ID_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
+
+/* Definition for a low latency stream session. */
+#define ADM_LOW_LATENCY_DEVICE_SESSION 0x2000
+
+/* Definition for a ultra low latency stream session. */
+#define ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION 0x4000
+
+/* Definition for a ultra low latency with Post Processing stream session. */
+#define ADM_ULL_POST_PROCESSING_DEVICE_SESSION 0x8000
+
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION 0
+
+/* Indicates that endpoint_id_2 is to be ignored.*/
+#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3
+
+/* Indicates that an audio COPP is to send/receive a mono PCM
+ * stream to/from
+ * END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1
+
+/* Indicates that an audio COPP is to send/receive a
+ * stereo PCM stream to/from END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2
+
+/* Sample rate is 8000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000
+
+/* Sample rate is 16000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000
+
+/* Sample rate is 48000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000
+
+/* Definition for a COPP live input flag bitmask.*/
+#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U)
+
+/* Definition for a COPP live shift value bitmask.*/
+#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0
+
+/* Definition for the COPP ID bitmask.*/
+#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL)
+
+/* Definition for the COPP ID shift value.*/
+#define ADM_SHIFT_COPP_ID 0
+
+/* Definition for the service ID bitmask.*/
+#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
+
+/* Definition for the service ID shift value.*/
+#define ADM_SHIFT_SERVICE_ID 16
+
+/* Definition for the domain ID bitmask.*/
+#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
+
+/* Definition for the domain ID shift value.*/
+#define ADM_SHIFT_DOMAIN_ID 24
+
+/* ADM device open command payload of the
+ * #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+struct adm_cmd_device_open_v5 {
+ struct apr_hdr hdr;
+ u16 flags;
+/* Reserved for future use. Clients must set this field
+ * to zero.
+ */
+
+ u16 mode_of_operation;
+/* Specifies whether the COPP must be opened on the Tx or Rx
+ * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
+ * supported values and interpretation.
+ * Supported values:
+ * - 0x1 -- Rx path COPP
+ * - 0x2 -- Tx path live COPP
+ * - 0x3 -- Tx path nonlive COPP
+ * Live connections cause sample discarding in the Tx device
+ * matrix if the destination output ports do not pull them
+ * fast enough. Nonlive connections queue the samples
+ * indefinitely.
+ */
+
+ u16 endpoint_id_1;
+/* Logical and physical endpoint ID of the audio path.
+ * If the ID is a voice processor Tx block, it receives near
+ * samples. Supported values: Any pseudoport, AFE Rx port,
+ * or AFE Tx port For a list of valid IDs, refer to
+ * @xhyperref{Q4,[Q4]}.
+ * Q4 = Hexagon Multimedia: AFE Interface Specification
+ */
+
+ u16 endpoint_id_2;
+/* Logical and physical endpoint ID 2 for a voice processor
+ * Tx block.
+ * This is not applicable to audio COPP.
+ * Supported values:
+ * - AFE Rx port
+ * - 0xFFFF -- Endpoint 2 is unavailable and the voice
+ * processor Tx
+ * block ignores this endpoint
+ * When the voice processor Tx block is created on the audio
+ * record path,
+ * it can receive far-end samples from an AFE Rx port if the
+ * voice call
+ * is active. The ID of the AFE port is provided in this
+ * field.
+ * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
+ */
+
+ u32 topology_id;
+/* Audio COPP topology ID; 32-bit GUID. */
+
+ u16 dev_num_channel;
+/* Number of channels the audio COPP sends to/receives from
+ * the endpoint.
+ * Supported values: 1 to 8.
+ * The value is ignored for the voice processor Tx block,
+ * where channel
+ * configuration is derived from the topology ID.
+ */
+
+ u16 bit_width;
+/* Bit width (in bits) that the audio COPP sends to/receives
+ * from the
+ * endpoint. The value is ignored for the voice processing
+ * Tx block,
+ * where the PCM width is 16 bits.
+ */
+
+ u32 sample_rate;
+/* Sampling rate at which the audio COPP/voice processor
+ * Tx block
+ * interfaces with the endpoint.
+ * Supported values for voice processor Tx: 8000, 16000,
+ * 48000 Hz
+ * Supported values for audio COPP: >0 and <=192 kHz
+ */
+
+ u8 dev_channel_mapping[8];
+/* Array of channel mapping of buffers that the audio COPP
+ * sends to the endpoint. Channel[i] mapping describes channel
+ * I inside the buffer, where 0 < i < dev_num_channel.
+ * This value is relevant only for an audio Rx COPP.
+ * For the voice processor block and Tx audio block, this field
+ * is set to zero and is ignored.
+ */
+} __packed;
+
+/* ADM device open command payload of the
+ * #ADM_CMD_DEVICE_OPEN_V6 command.
+ */
+struct adm_cmd_device_open_v6 {
+ struct apr_hdr hdr;
+ u16 flags;
+/* Reserved for future use. Clients must set this field
+ * to zero.
+ */
+
+ u16 mode_of_operation;
+/* Specifies whether the COPP must be opened on the Tx or Rx
+ * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for
+ * supported values and interpretation.
+ * Supported values:
+ * - 0x1 -- Rx path COPP
+ * - 0x2 -- Tx path live COPP
+ * - 0x3 -- Tx path nonlive COPP
+ * Live connections cause sample discarding in the Tx device
+ * matrix if the destination output ports do not pull them
+ * fast enough. Nonlive connections queue the samples
+ * indefinitely.
+ */
+
+ u16 endpoint_id_1;
+/* Logical and physical endpoint ID of the audio path.
+ * If the ID is a voice processor Tx block, it receives near
+ * samples. Supported values: Any pseudoport, AFE Rx port,
+ * or AFE Tx port For a list of valid IDs, refer to
+ * @xhyperref{Q4,[Q4]}.
+ * Q4 = Hexagon Multimedia: AFE Interface Specification
+ */
+
+ u16 endpoint_id_2;
+/* Logical and physical endpoint ID 2 for a voice processor
+ * Tx block.
+ * This is not applicable to audio COPP.
+ * Supported values:
+ * - AFE Rx port
+ * - 0xFFFF -- Endpoint 2 is unavailable and the voice
+ * processor Tx
+ * block ignores this endpoint
+ * When the voice processor Tx block is created on the audio
+ * record path,
+ * it can receive far-end samples from an AFE Rx port if the
+ * voice call
+ * is active. The ID of the AFE port is provided in this
+ * field.
+ * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.
+ */
+
+ u32 topology_id;
+/* Audio COPP topology ID; 32-bit GUID. */
+
+ u16 dev_num_channel;
+/* Number of channels the audio COPP sends to/receives from
+ * the endpoint.
+ * Supported values: 1 to 8.
+ * The value is ignored for the voice processor Tx block,
+ * where channel
+ * configuration is derived from the topology ID.
+ */
+
+ u16 bit_width;
+/* Bit width (in bits) that the audio COPP sends to/receives
+ * from the
+ * endpoint. The value is ignored for the voice processing
+ * Tx block,
+ * where the PCM width is 16 bits.
+ */
+
+ u32 sample_rate;
+/* Sampling rate at which the audio COPP/voice processor
+ * Tx block
+ * interfaces with the endpoint.
+ * Supported values for voice processor Tx: 8000, 16000,
+ * 48000 Hz
+ * Supported values for audio COPP: >0 and <=192 kHz
+ */
+
+ u8 dev_channel_mapping[8];
+/* Array of channel mapping of buffers that the audio COPP
+ * sends to the endpoint. Channel[i] mapping describes channel
+ * I inside the buffer, where 0 < i < dev_num_channel.
+ * This value is relevant only for an audio Rx COPP.
+ * For the voice processor block and Tx audio block, this field
+ * is set to zero and is ignored.
+ */
+
+ u16 dev_num_channel_eid2;
+/* Number of channels the voice processor block sends
+ * to/receives from the endpoint2.
+ * Supported values: 1 to 8.
+ * The value is ignored for audio COPP or if endpoint_id_2 is
+ * set to 0xFFFF.
+ */
+
+ u16 bit_width_eid2;
+/* Bit width (in bits) that the voice processor sends
+ * to/receives from the endpoint2.
+ * Supported values: 16 and 24.
+ * The value is ignored for audio COPP or if endpoint_id_2 is
+ * set to 0xFFFF.
+ */
+
+ u32 sample_rate_eid2;
+/* Sampling rate at which the voice processor Tx block
+ * interfaces with the endpoint2.
+ * Supported values for Tx voice processor: >0 and <=384 kHz
+ * The value is ignored for audio COPP or if endpoint_id_2 is
+ * set to 0xFFFF.
+ */
+
+ u8 dev_channel_mapping_eid2[8];
+/* Array of channel mapping of buffers that the voice processor
+ * sends to the endpoint. Channel[i] mapping describes channel
+ * I inside the buffer, where 0 < i < dev_num_channel.
+ * This value is relevant only for the Tx voice processor.
+ * The values are ignored for audio COPP or if endpoint_id_2 is
+ * set to 0xFFFF.
+ */
+} __packed;
+
+/*
+ * This command allows the client to close a COPP and disconnect
+ * the device session.
+ */
+#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
+
+/* Sets one or more parameters to a COPP. */
+#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328
+
+/* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command.
+ * If the data_payload_addr_lsw and data_payload_addr_msw element
+ * are NULL, a series of adm_param_datastructures immediately
+ * follows, whose total size is data_payload_size bytes.
+ */
+struct adm_cmd_set_pp_params_v5 {
+ struct apr_hdr hdr;
+ u32 payload_addr_lsw;
+/* LSW of parameter data payload address. */
+ u32 payload_addr_msw;
+/* MSW of parameter data payload address. */
+
+ u32 mem_map_handle;
+/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * command
+ *
+ * If mem_map_handle is zero implies the message is in
+ * the payload
+ */
+
+ u32 payload_size;
+/* Size in bytes of the variable payload accompanying this
+ * message or
+ * in shared memory. This is used for parsing the parameter
+ * payload.
+ */
+} __packed;
+
+/* Payload format for COPP parameter data.
+ * Immediately following this structure are param_size bytes
+ * of parameter
+ * data.
+ */
+struct adm_param_data_v5 {
+ u32 module_id;
+ /* Unique ID of the module. */
+ u32 param_id;
+ /* Unique ID of the parameter. */
+ u16 param_size;
+ /* Data size of the param_id/module_id combination.
+ * This value is a
+ * multiple of 4 bytes.
+ */
+ u16 reserved;
+ /* Reserved for future enhancements.
+ * This field must be set to zero.
+ */
+} __packed;
+
+#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;
+
+/* Defined specifically for in-band use, includes params */
+struct adm_cmd_set_pp_params_inband_v5 {
+ struct apr_hdr hdr;
+ /* LSW of parameter data payload address.*/
+ u32 payload_addr_lsw;
+ /* MSW of parameter data payload address.*/
+ u32 payload_addr_msw;
+ /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS */
+ /* command. If mem_map_handle is zero implies the message is in */
+ /* the payload */
+ u32 mem_map_handle;
+ /* Size in bytes of the variable payload accompanying this */
+ /* message or in shared memory. This is used for parsing the */
+ /* parameter payload. */
+ u32 payload_size;
+ /* Parameters passed for in band payload */
+ struct adm_param_data_v5 params;
+} __packed;
+
+/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
+
+/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message,
+ * which returns the
+ * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+struct adm_cmd_rsp_device_open_v5 {
+ u32 status;
+ /* Status message (error code).*/
+
+ u16 copp_id;
+ /* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/
+
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command. */
+#define ADM_CMDRSP_DEVICE_OPEN_V6 0x00010357
+
+/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message,
+ * which returns the
+ * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command
+ * is the exact same as ADM_CMDRSP_DEVICE_OPEN_V5.
+ */
+
+/* This command allows a query of one COPP parameter. */
+#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A
+
+/* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command. */
+struct adm_cmd_get_pp_params_v5 {
+ struct apr_hdr hdr;
+ u32 data_payload_addr_lsw;
+ /* LSW of parameter data payload address.*/
+
+ u32 data_payload_addr_msw;
+ /* MSW of parameter data payload address.*/
+
+ /* If the mem_map_handle is non zero,
+ * on ACK, the ParamData payloads begin at
+ * the address specified (out-of-band).
+ */
+
+ u32 mem_map_handle;
+ /* Memory map handle returned
+ * by ADM_CMD_SHARED_MEM_MAP_REGIONS command.
+ * If the mem_map_handle is 0, it implies that
+ * the ACK's payload will contain the ParamData (in-band).
+ */
+
+ u32 module_id;
+ /* Unique ID of the module. */
+
+ u32 param_id;
+ /* Unique ID of the parameter. */
+
+ u16 param_max_size;
+ /* Maximum data size of the parameter
+ *ID/module ID combination. This
+ * field is a multiple of 4 bytes.
+ */
+ u16 reserved;
+ /* Reserved for future enhancements.
+ * This field must be set to zero.
+ */
+} __packed;
+
+/* Returns parameter values
+ * in response to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ */
+#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B
+
+/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message,
+ * which returns parameter values in response
+ * to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ * Immediately following this
+ * structure is the adm_param_data_v5
+ * structure containing the pre/postprocessing
+ * parameter data. For an in-band
+ * scenario, the variable payload depends
+ * on the size of the parameter.
+ */
+struct adm_cmd_rsp_get_pp_params_v5 {
+ u32 status;
+ /* Status message (error code).*/
+} __packed;
+
+/* Structure for holding soft stepping volume parameters. */
+
+/*
+ * Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+
+struct audproc_softvolume_params {
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+/*
+ * ID of the Media Format Converter (MFC) module.
+ * This module supports the following parameter IDs:
+ * #AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT
+ * #AUDPROC_CHMIXER_PARAM_ID_COEFF
+ */
+#define AUDPROC_MODULE_ID_MFC 0x00010912
+
+/* ID of the Output Media Format parameters used by AUDPROC_MODULE_ID_MFC.
+ *
+ */
+#define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913
+
+
+struct audproc_mfc_output_media_fmt {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+ uint32_t sampling_rate;
+ uint16_t bits_per_sample;
+ uint16_t num_channels;
+ uint16_t channel_type[8];
+} __packed;
+
+struct audproc_volume_ctrl_master_gain {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+ /* Linear gain in Q13 format. */
+ uint16_t master_gain;
+ /* Clients must set this field to zero. */
+ uint16_t reserved;
+} __packed;
+
+struct audproc_soft_step_volume_params {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+/*
+ * Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+ uint32_t period;
+/*
+ * Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+ uint32_t step;
+/*
+ * Ramping curve type.
+ * Supported values:
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #AUDPROC_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+ uint32_t ramping_curve;
+} __packed;
+
+struct audproc_enable_param_t {
+ struct adm_cmd_set_pp_params_inband_v5 pp_params;
+ /*
+ * Specifies whether the Audio processing module is enabled.
+ * This parameter is generic/common parameter to configure or
+ * determine the state of any audio processing module.
+
+ * @values 0 : Disable 1: Enable
+ */
+ uint32_t enable;
+};
+
+/*
+ * Allows a client to control the gains on various session-to-COPP paths.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C
+
+/* Indicates that the target gain in the
+ * current adm_session_copp_gain_v5
+ * structure is to be applied to all
+ * the session-to-COPP paths that exist for
+ * the specified session.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
+
+/* Indicates that the target gain is
+ * to be immediately applied to the
+ * specified session-to-COPP path,
+ * without a ramping fashion.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000
+
+/* Enumeration for a linear ramping curve.*/
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000
+
+/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
+ * Immediately following this structure are num_gains of the
+ * adm_session_copp_gain_v5structure.
+ */
+struct adm_cmd_matrix_ramp_gains_v5 {
+ u32 matrix_id;
+/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
+ * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
+ * macros to set this field.
+ */
+
+ u16 num_gains;
+ /* Number of gains being applied. */
+
+ u16 reserved_for_align;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+/* Session-to-COPP path gain structure, used by the
+ * #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
+ * This structure specifies the target
+ * gain (per channel) that must be applied
+ * to a particular session-to-COPP path in
+ * the audio matrix. The structure can
+ * also be used to apply the gain globally
+ * to all session-to-COPP paths that
+ * exist for the given session.
+ * The aDSP uses device channel mapping to
+ * determine which channel gains to
+ * use from this command. For example,
+ * if the device is configured as stereo,
+ * the aDSP uses only target_gain_ch_1 and
+ * target_gain_ch_2, and it ignores
+ * the others.
+ */
+struct adm_session_copp_gain_v5 {
+ u16 session_id;
+/* Handle of the ASM session.
+ * Supported values: 1 to 8.
+ */
+
+ u16 copp_id;
+/* Handle of the COPP. Gain will be applied on the Session ID
+ * COPP ID path.
+ */
+
+ u16 ramp_duration;
+/* Duration (in milliseconds) of the ramp over
+ * which target gains are
+ * to be applied. Use
+ * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE
+ * to indicate that gain must be applied immediately.
+ */
+
+ u16 step_duration;
+/* Duration (in milliseconds) of each step in the ramp.
+ * This parameter is ignored if ramp_duration is equal to
+ * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE.
+ * Supported value: 1
+ */
+
+ u16 ramp_curve;
+/* Type of ramping curve.
+ * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR
+ */
+
+ u16 reserved_for_align;
+ /* Reserved. This field must be set to zero. */
+
+ u16 target_gain_ch_1;
+ /* Target linear gain for channel 1 in Q13 format; */
+
+ u16 target_gain_ch_2;
+ /* Target linear gain for channel 2 in Q13 format; */
+
+ u16 target_gain_ch_3;
+ /* Target linear gain for channel 3 in Q13 format; */
+
+ u16 target_gain_ch_4;
+ /* Target linear gain for channel 4 in Q13 format; */
+
+ u16 target_gain_ch_5;
+ /* Target linear gain for channel 5 in Q13 format; */
+
+ u16 target_gain_ch_6;
+ /* Target linear gain for channel 6 in Q13 format; */
+
+ u16 target_gain_ch_7;
+ /* Target linear gain for channel 7 in Q13 format; */
+
+ u16 target_gain_ch_8;
+ /* Target linear gain for channel 8 in Q13 format; */
+} __packed;
+
+/* Allows to set mute/unmute on various session-to-COPP paths.
+ * For every session-to-COPP path (stream-device interconnection),
+ * mute/unmute can be set individually on the output channels.
+ */
+#define ADM_CMD_MATRIX_MUTE_V5 0x0001032D
+
+/* Indicates that mute/unmute in the
+ * current adm_session_copp_mute_v5structure
+ * is to be applied to all the session-to-COPP
+ * paths that exist for the specified session.
+ */
+#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
+
+/* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/
+struct adm_cmd_matrix_mute_v5 {
+ u32 matrix_id;
+/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).
+ * Use the ADM_MATRIX_ID_AUDIO_RX or ADM_MATRIX_ID_AUDIOX
+ * macros to set this field.
+ */
+
+ u16 session_id;
+/* Handle of the ASM session.
+ * Supported values: 1 to 8.
+ */
+
+ u16 copp_id;
+/* Handle of the COPP.
+ * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS
+ * to indicate that mute/unmute must be applied to
+ * all the COPPs connected to session_id.
+ * Supported values:
+ * - 0xFFFF -- Apply mute/unmute to all connected COPPs
+ * - Other values -- Valid COPP ID
+ */
+
+ u8 mute_flag_ch_1;
+ /* Mute flag for channel 1 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_2;
+ /* Mute flag for channel 2 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_3;
+ /* Mute flag for channel 3 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_4;
+ /* Mute flag for channel 4 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_5;
+ /* Mute flag for channel 5 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_6;
+ /* Mute flag for channel 6 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_7;
+ /* Mute flag for channel 7 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_8;
+ /* Mute flag for channel 8 is set to unmute (0) or mute (1). */
+
+ u16 ramp_duration;
+/* Period (in milliseconds) over which the soft mute/unmute will be
+ * applied.
+ * Supported values: 0 (Default) to 0xFFFF
+ * The default of 0 means mute/unmute will be applied immediately.
+ */
+
+ u16 reserved_for_align;
+ /* Clients must set this field to zero.*/
+} __packed;
+
+#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8)
+
+struct asm_aac_stereo_mix_coeff_selection_param_v2 {
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ u32 aac_stereo_mix_coeff_flag;
+} __packed;
+
+/* Allows a client to connect the desired stream to
+ * the desired AFE port through the stream router
+ *
+ * This command allows the client to connect specified session to
+ * specified AFE port. This is used for compressed streams only
+ * opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command.
+ *
+ * @prerequisites
+ * Session ID and AFE Port ID must be valid.
+ * #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED
+ * must have been called on this session.
+ */
+
+#define ADM_CMD_CONNECT_AFE_PORT_V5 0x0001032E
+#define ADM_CMD_DISCONNECT_AFE_PORT_V5 0x0001032F
+/* Enumeration for the Rx stream router ID.*/
+#define ADM_STRTR_ID_RX 0
+/* Enumeration for the Tx stream router ID.*/
+#define ADM_STRTR_IDX 1
+
+/* Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/
+struct adm_cmd_connect_afe_port_v5 {
+ struct apr_hdr hdr;
+ u8 mode;
+/* ID of the stream router (RX/TX). Use the
+ * ADM_STRTR_ID_RX or ADM_STRTR_IDX macros
+ * to set this field.
+ */
+
+ u8 session_id;
+ /* Session ID of the stream to connect */
+
+ u16 afe_port_id;
+ /* Port ID of the AFE port to connect to.*/
+ u32 num_channels;
+/* Number of device channels
+ * Supported values: 2(Audio Sample Packet),
+ * 8 (HBR Audio Stream Sample Packet)
+ */
+
+ u32 sampling_rate;
+/* Device sampling rate
+ * Supported values: Any
+ */
+} __packed;
+
+
+/* adsp_adm_api.h */
+
+
+/* Port ID. Update afe_get_port_index
+ * when a new port is added here.
+ */
+#define PRIMARY_I2S_RX 0
+#define PRIMARY_I2S_TX 1
+#define SECONDARY_I2S_RX 4
+#define SECONDARY_I2S_TX 5
+#define MI2S_RX 6
+#define MI2S_TX 7
+#define HDMI_RX 8
+#define RSVD_2 9
+#define RSVD_3 10
+#define DIGI_MIC_TX 11
+#define VOICE2_PLAYBACK_TX 0x8002
+#define VOICE_RECORD_RX 0x8003
+#define VOICE_RECORD_TX 0x8004
+#define VOICE_PLAYBACK_TX 0x8005
+
+/* Slimbus Multi channel port id pool */
+#define SLIMBUS_0_RX 0x4000
+#define SLIMBUS_0_TX 0x4001
+#define SLIMBUS_1_RX 0x4002
+#define SLIMBUS_1_TX 0x4003
+#define SLIMBUS_2_RX 0x4004
+#define SLIMBUS_2_TX 0x4005
+#define SLIMBUS_3_RX 0x4006
+#define SLIMBUS_3_TX 0x4007
+#define SLIMBUS_4_RX 0x4008
+#define SLIMBUS_4_TX 0x4009
+#define SLIMBUS_5_RX 0x400a
+#define SLIMBUS_5_TX 0x400b
+#define SLIMBUS_6_RX 0x400c
+#define SLIMBUS_6_TX 0x400d
+#define SLIMBUS_7_RX 0x400e
+#define SLIMBUS_7_TX 0x400f
+#define SLIMBUS_8_RX 0x4010
+#define SLIMBUS_8_TX 0x4011
+#define SLIMBUS_PORT_LAST SLIMBUS_8_TX
+#define INT_BT_SCO_RX 0x3000
+#define INT_BT_SCO_TX 0x3001
+#define INT_BT_A2DP_RX 0x3002
+#define INT_FM_RX 0x3004
+#define INT_FM_TX 0x3005
+#define RT_PROXY_PORT_001_RX 0x2000
+#define RT_PROXY_PORT_001_TX 0x2001
+#define DISPLAY_PORT_RX 0x6020
+
+#define AFE_PORT_INVALID 0xFFFF
+#define SLIMBUS_INVALID AFE_PORT_INVALID
+
+#define AFE_PORT_CMD_START 0x000100ca
+
+#define AFE_EVENT_RTPORT_START 0
+#define AFE_EVENT_RTPORT_STOP 1
+#define AFE_EVENT_RTPORT_LOW_WM 2
+#define AFE_EVENT_RTPORT_HI_WM 3
+
+#define ADSP_AFE_VERSION 0x00200000
+
+/* Size of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE 0xF
+
+/* Size of the range of port IDs for internal BT-FM ports. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE 0x6
+
+/* Size of the range of port IDs for SLIMbus<sup>®
+ * </sup> multichannel
+ * ports.
+ */
+#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE 0xA
+
+/* Size of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE 0x2
+
+/* Size of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE 0x5
+
+/* Start of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START 0x1000
+
+/* End of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \
+ (AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\
+ AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1)
+
+/* Start of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_START 0x2000
+
+/* End of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \
+ (AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\
+ AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START 0x3000
+
+/* End of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \
+ (AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\
+ AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_START 0x4000
+
+/* End of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_END \
+ (AFE_PORT_ID_SLIMBUS_RANGE_START +\
+ AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_START 0x8001
+
+/* End of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \
+ (AFE_PORT_ID_PSEUDOPORT_RANGE_START +\
+ AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for TDM devices. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_START 0x9000
+
+/* End of the range of port IDs for TDM devices. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_END \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START+0x40-1)
+
+/* Size of the range of port IDs for TDM ports. */
+#define AFE_PORT_ID_TDM_PORT_RANGE_SIZE \
+ (AFE_PORT_ID_TDM_PORT_RANGE_END - \
+ AFE_PORT_ID_TDM_PORT_RANGE_START+1)
+
+#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
+#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004
+#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+#define AUDIO_PORT_ID_I2S_RX 0x1008
+#define AFE_PORT_ID_DIGITAL_MIC_TX 0x1009
+#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A
+#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
+#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C
+#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define AFE_PORT_ID_SECONDARY_MI2S_RX_SD1 0x1010
+#define AFE_PORT_ID_TERTIARY_PCM_RX 0x1012
+#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013
+#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014
+#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015
+#define AFE_PORT_ID_QUINARY_MI2S_RX 0x1016
+#define AFE_PORT_ID_QUINARY_MI2S_TX 0x1017
+/* ID of the senary MI2S Rx port. */
+#define AFE_PORT_ID_SENARY_MI2S_RX 0x1018
+/* ID of the senary MI2S Tx port. */
+#define AFE_PORT_ID_SENARY_MI2S_TX 0x1019
+/* ID of the Internal 0 MI2S Rx port */
+#define AFE_PORT_ID_INT0_MI2S_RX 0x102E
+/* ID of the Internal 0 MI2S Tx port */
+#define AFE_PORT_ID_INT0_MI2S_TX 0x102F
+/* ID of the Internal 1 MI2S Rx port */
+#define AFE_PORT_ID_INT1_MI2S_RX 0x1030
+/* ID of the Internal 1 MI2S Tx port */
+#define AFE_PORT_ID_INT1_MI2S_TX 0x1031
+/* ID of the Internal 2 MI2S Rx port */
+#define AFE_PORT_ID_INT2_MI2S_RX 0x1032
+/* ID of the Internal 2 MI2S Tx port */
+#define AFE_PORT_ID_INT2_MI2S_TX 0x1033
+/* ID of the Internal 3 MI2S Rx port */
+#define AFE_PORT_ID_INT3_MI2S_RX 0x1034
+/* ID of the Internal 3 MI2S Tx port */
+#define AFE_PORT_ID_INT3_MI2S_TX 0x1035
+/* ID of the Internal 4 MI2S Rx port */
+#define AFE_PORT_ID_INT4_MI2S_RX 0x1036
+/* ID of the Internal 4 MI2S Tx port */
+#define AFE_PORT_ID_INT4_MI2S_TX 0x1037
+/* ID of the Internal 5 MI2S Rx port */
+#define AFE_PORT_ID_INT5_MI2S_RX 0x1038
+/* ID of the Internal 5 MI2S Tx port */
+#define AFE_PORT_ID_INT5_MI2S_TX 0x1039
+/* ID of the Internal 6 MI2S Rx port */
+#define AFE_PORT_ID_INT6_MI2S_RX 0x103A
+/* ID of the Internal 6 MI2S Tx port */
+#define AFE_PORT_ID_INT6_MI2S_TX 0x103B
+#define AFE_PORT_ID_SPDIF_RX 0x5000
+#define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000
+#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001
+#define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000
+#define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001
+#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002
+#define AFE_PORT_ID_INTERNAL_FM_RX 0x3004
+#define AFE_PORT_ID_INTERNAL_FM_TX 0x3005
+/* SLIMbus Rx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
+/* SLIMbus Tx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
+/* SLIMbus Rx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
+/* SLIMbus Tx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
+/* SLIMbus Rx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
+/* SLIMbus Tx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
+/* SLIMbus Rx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
+/* SLIMbus Tx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
+/* SLIMbus Rx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
+/* SLIMbus Tx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
+/* SLIMbus Rx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX 0x400a
+/* SLIMbus Tx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX 0x400b
+/* SLIMbus Rx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
+/* SLIMbus Tx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
+/* SLIMbus Rx port on channel 7. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_RX 0x400e
+/* SLIMbus Tx port on channel 7. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_7_TX 0x400f
+/* SLIMbus Rx port on channel 8. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_RX 0x4010
+/* SLIMbus Tx port on channel 8. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_8_TX 0x4011
+/* AFE Rx port for audio over Display port */
+#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020
+/*USB AFE port */
+#define AFE_PORT_ID_USB_RX 0x7000
+#define AFE_PORT_ID_USB_TX 0x7001
+
+/* Generic pseudoport 1. */
+#define AFE_PORT_ID_PSEUDOPORT_01 0x8001
+/* Generic pseudoport 2. */
+#define AFE_PORT_ID_PSEUDOPORT_02 0x8002
+
+/* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx}
+ * Primary Aux PCM Tx port ID.
+ */
+#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
+/* Pseudoport that corresponds to the voice Rx path.
+ * For recording, the voice Rx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_RX 0x8003
+
+/* Pseudoport that corresponds to the voice Tx path.
+ * For recording, the voice Tx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_TX 0x8004
+/* Pseudoport that corresponds to in-call voice delivery samples.
+ * During in-call audio delivery, the audio path delivers samples
+ * to this port from where the voice path delivers them on the
+ * Rx path.
+ */
+#define AFE_PORT_ID_VOICE2_PLAYBACK_TX 0x8002
+#define AFE_PORT_ID_VOICE_PLAYBACK_TX 0x8005
+
+#define AFE_PORT_ID_PRIMARY_TDM_RX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x00)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_1 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_2 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_3 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_4 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_5 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_6 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_PRIMARY_TDM_RX_7 \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_PRIMARY_TDM_TX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x01)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_1 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_2 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_3 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_4 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_5 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_6 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_PRIMARY_TDM_TX_7 \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_SECONDARY_TDM_RX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x10)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_1 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_2 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_3 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_4 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_5 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_6 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_SECONDARY_TDM_RX_7 \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_SECONDARY_TDM_TX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x11)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_1 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_2 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_3 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_4 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_5 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_6 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_SECONDARY_TDM_TX_7 \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_TERTIARY_TDM_RX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x20)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_1 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_2 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_3 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_4 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_5 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_6 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_TERTIARY_TDM_RX_7 \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_TERTIARY_TDM_TX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x21)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_1 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_2 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_3 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_4 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_5 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_6 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_TERTIARY_TDM_TX_7 \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_QUATERNARY_TDM_RX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x30)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_1 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x02)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_2 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x04)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_3 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x06)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_4 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x08)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_5 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0A)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_6 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0C)
+#define AFE_PORT_ID_QUATERNARY_TDM_RX_7 \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x0E)
+
+#define AFE_PORT_ID_QUATERNARY_TDM_TX \
+ (AFE_PORT_ID_TDM_PORT_RANGE_START + 0x31)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_1 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x02)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_2 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x04)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_3 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x06)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_4 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x08)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_5 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0A)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_6 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0C)
+#define AFE_PORT_ID_QUATERNARY_TDM_TX_7 \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x0E)
+
+#define AFE_PORT_ID_INVALID 0xFFFF
+
+#define AAC_ENC_MODE_AAC_LC 0x02
+#define AAC_ENC_MODE_AAC_P 0x05
+#define AAC_ENC_MODE_EAAC_P 0x1D
+
+#define AFE_PSEUDOPORT_CMD_START 0x000100cf
+struct afe_pseudoport_start_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 timing; /* FTRT = 0 , AVTimer = 1, */
+} __packed;
+
+#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
+struct afe_pseudoport_stop_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 reserved;
+} __packed;
+
+
+#define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202
+#define AFE_PARAM_ID_ENABLE 0x00010203
+
+/* Payload of the #AFE_PARAM_ID_ENABLE
+ * parameter, which enables or
+ * disables any module.
+ * The fixed size of this structure is four bytes.
+ */
+
+struct afe_mod_enable_param {
+ u16 enable;
+ /* Enables (1) or disables (0) the module. */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+/* ID of the configuration parameter used by the
+ * #AFE_MODULE_SIDETONE_IIR_FILTER module.
+ */
+#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204
+#define MAX_SIDETONE_IIR_DATA_SIZE 224
+#define MAX_NO_IIR_FILTER_STAGE 10
+
+struct afe_sidetone_iir_filter_config_params {
+ u16 num_biquad_stages;
+/* Number of stages.
+ * Supported values: Minimum of 5 and maximum of 10
+ */
+
+ u16 pregain;
+/* Pregain for the compensating filter response.
+ * Supported values: Any number in Q13 format
+ */
+ uint8_t iir_config[MAX_SIDETONE_IIR_DATA_SIZE];
+} __packed;
+
+#define AFE_MODULE_LOOPBACK 0x00010205
+#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206
+
+/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter,
+ * which gets/sets loopback gain of a port to an Rx port.
+ * The Tx port ID of the loopback is part of the set_param command.
+ */
+
+/* Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's
+ * configuration/calibration settings for the AFE port.
+ */
+struct afe_port_cmd_set_param_v2 {
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. */
+
+ u16 payload_size;
+/* Actual size of the payload in bytes.
+ * This is used for parsing the parameter payload.
+ * Supported values: > 0
+ */
+
+u32 payload_address_lsw;
+/* LSW of 64 bit Payload address.
+ * Address should be 32-byte,
+ * 4kbyte aligned and must be contiguous memory.
+ */
+
+u32 payload_address_msw;
+/* MSW of 64 bit Payload address.
+ * In case of 32-bit shared memory address,
+ * this field must be set to zero.
+ * In case of 36-bit shared memory address,
+ * bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned
+ * and must be contiguous memory.
+ */
+
+u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values:
+ * - NULL -- Message. The parameter data is in-band.
+ * - Non-NULL -- The parameter data is Out-band.Pointer to
+ * the physical address
+ * in shared memory of the payload data.
+ * An optional field is available if parameter
+ * data is in-band:
+ * afe_param_data_v2 param_data[...].
+ * For detailed payload content, see the
+ * afe_port_param_data_v2 structure.
+ */
+} __packed;
+
+#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+
+struct afe_port_param_data_v2 {
+ u32 module_id;
+/* ID of the module to be configured.
+ * Supported values: Valid module ID
+ */
+
+u32 param_id;
+/* ID of the parameter corresponding to the supported parameters
+ * for the module ID.
+ * Supported values: Valid parameter ID
+ */
+
+u16 param_size;
+/* Actual size of the data for the
+ * module_id/param_id pair. The size is a
+ * multiple of four bytes.
+ * Supported values: > 0
+ */
+
+u16 reserved;
+/* This field must be set to zero.
+ */
+} __packed;
+
+struct afe_loopback_gain_per_path_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ u16 rx_port_id;
+/* Rx port of the loopback. */
+
+u16 gain;
+/* Loopback gain per path of the port.
+ * Supported values: Any number in Q13 format
+ */
+} __packed;
+
+/* Parameter ID used to configure and enable/disable the
+ * loopback path. The difference with respect to the existing
+ * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be
+ * configured as source port in loopback path. Port-id in
+ * AFE_PORT_CMD_SET_PARAM cmd is the source port which can be
+ * Tx or Rx port. In addition, we can configure the type of
+ * routing mode to handle different use cases.
+ */
+#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
+#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
+
+enum afe_loopback_routing_mode {
+ LB_MODE_DEFAULT = 1,
+ /* Regular loopback from source to destination port */
+ LB_MODE_SIDETONE,
+ /* Sidetone feed from Tx source to Rx destination port */
+ LB_MODE_EC_REF_VOICE_AUDIO,
+ /* Echo canceller reference, voice + audio + DTMF */
+ LB_MODE_EC_REF_VOICE
+ /* Echo canceller reference, voice alone */
+} __packed;
+
+/* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG ,
+ * which enables/disables one AFE loopback.
+ */
+struct afe_loopback_cfg_v1 {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ u32 loopback_cfg_minor_version;
+/* Minor version used for tracking the version of the RMC module
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
+ */
+ u16 dst_port_id;
+ /* Destination Port Id. */
+ u16 routing_mode;
+/* Specifies data path type from src to dest port.
+ * Supported values:
+ * #LB_MODE_DEFAULT
+ * #LB_MODE_SIDETONE
+ * #LB_MODE_EC_REF_VOICE_AUDIO
+ * #LB_MODE_EC_REF_VOICE_A
+ * #LB_MODE_EC_REF_VOICE
+ */
+
+ u16 enable;
+/* Specifies whether to enable (1) or
+ * disable (0) an AFE loopback.
+ */
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.
+ */
+
+} __packed;
+
+struct afe_loopback_sidetone_gain {
+ u16 rx_port_id;
+ u16 gain;
+} __packed;
+
+struct loopback_cfg_data {
+ u32 loopback_cfg_minor_version;
+/* Minor version used for tracking the version of the RMC module
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
+ */
+ u16 dst_port_id;
+ /* Destination Port Id. */
+ u16 routing_mode;
+/* Specifies data path type from src to dest port.
+ * Supported values:
+ * #LB_MODE_DEFAULT
+ * #LB_MODE_SIDETONE
+ * #LB_MODE_EC_REF_VOICE_AUDIO
+ * #LB_MODE_EC_REF_VOICE_A
+ * #LB_MODE_EC_REF_VOICE
+ */
+
+ u16 enable;
+/* Specifies whether to enable (1) or
+ * disable (0) an AFE loopback.
+ */
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.
+ */
+} __packed;
+
+struct afe_st_loopback_cfg_v1 {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 gain_pdata;
+ struct afe_loopback_sidetone_gain gain_data;
+ struct afe_port_param_data_v2 cfg_pdata;
+ struct loopback_cfg_data cfg_data;
+} __packed;
+
+struct afe_loopback_iir_cfg_v2 {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 st_iir_enable_pdata;
+ struct afe_mod_enable_param st_iir_mode_enable_data;
+ struct afe_port_param_data_v2 st_iir_filter_config_pdata;
+ struct afe_sidetone_iir_filter_config_params st_iir_filter_config_data;
+} __packed;
+#define AFE_MODULE_SPEAKER_PROTECTION 0x00010209
+#define AFE_PARAM_ID_SPKR_PROT_CONFIG 0x0001020a
+#define AFE_API_VERSION_SPKR_PROT_CONFIG 0x1
+#define AFE_SPKR_PROT_EXCURSIONF_LEN 512
+struct afe_spkr_prot_cfg_param_v1 {
+ u32 spkr_prot_minor_version;
+/*
+ * Minor version used for tracking the version of the
+ * speaker protection module configuration interface.
+ * Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG
+ */
+
+int16_t win_size;
+/* Analysis and synthesis window size (nWinSize).
+ * Supported values: 1024, 512, 256 samples
+ */
+
+int16_t margin;
+/* Allowable margin for excursion prediction,
+ * in L16Q15 format. This is a
+ * control parameter to allow
+ * for overestimation of peak excursion.
+ */
+
+int16_t spkr_exc_limit;
+/* Speaker excursion limit, in L16Q15 format.*/
+
+int16_t spkr_resonance_freq;
+/* Resonance frequency of the speaker; used
+ * to define a frequency range
+ * for signal modification.
+ *
+ * Supported values: 0 to 2000 Hz
+ */
+
+int16_t limhresh;
+/* Threshold of the hard limiter; used to
+ * prevent overshooting beyond a
+ * signal level that was set by the limiter
+ * prior to speaker protection.
+ * Supported values: 0 to 32767
+ */
+
+int16_t hpf_cut_off_freq;
+/* High pass filter cutoff frequency.
+ * Supported values: 100, 200, 300 Hz
+ */
+
+int16_t hpf_enable;
+/* Specifies whether the high pass filter
+ * is enabled (0) or disabled (1).
+ */
+
+int16_t reserved;
+/* This field must be set to zero. */
+
+int32_t amp_gain;
+/* Amplifier gain in L32Q15 format.
+ * This is the RMS voltage at the
+ * loudspeaker when a 0dBFS tone
+ * is played in the digital domain.
+ */
+
+int16_t excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN];
+/* Array of the excursion transfer function.
+ * The peak excursion of the
+ * loudspeaker diaphragm is
+ * measured in millimeters for 1 Vrms Sine
+ * tone at all FFT bin frequencies.
+ * Supported values: Q15 format
+ */
+} __packed;
+
+
+#define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER 0x000100E0
+
+/* Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER
+ * command, which registers a real-time port driver
+ * with the AFE service.
+ */
+struct afe_service_cmd_register_rt_port_driver {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port ID with which the real-time driver exchanges data
+ * (registers for events).
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+#define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER 0x000100E1
+
+/* Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER
+ * command, which unregisters a real-time port driver from
+ * the AFE service.
+ */
+struct afe_service_cmd_unregister_rt_port_driver {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port ID from which the real-time
+ * driver unregisters for events.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105
+#define AFE_EVENTYPE_RT_PROXY_PORT_START 0
+#define AFE_EVENTYPE_RT_PROXY_PORT_STOP 1
+#define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK 2
+#define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK 3
+#define AFE_EVENTYPE_RT_PROXY_PORT_INVALID 0xFFFF
+
+/* Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * message, which sends an event from the AFE service
+ * to a registered client.
+ */
+struct afe_event_rt_proxy_port_status {
+ u16 port_id;
+/* Port ID to which the event is sent.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 eventype;
+/* Type of event.
+ * Supported values:
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_START
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_STOP
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED
+
+struct afe_port_data_cmd_rt_proxy_port_write_v2 {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Tx (mic) proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 buffer_address_lsw;
+/* LSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+ u32 buffer_address_msw;
+/* MSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory
+ * attributes is returned if
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS
+ * command is successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+ u32 available_bytes;
+/* Number of valid bytes available
+ * in the buffer (including all
+ * channels: number of bytes per
+ * channel = availableBytesumChannels).
+ * Supported values: > 0
+ *
+ * This field must be equal to the frame
+ * size specified in the #AFE_PORT_AUDIO_IF_CONFIG
+ * command that was sent to configure this
+ * port.
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 0x000100EE
+
+/* Payload of the
+ * #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which
+ * delivers an empty buffer to the AFE service. On
+ * acknowledgment, data is filled in the buffer.
+ */
+struct afe_port_data_cmd_rt_proxy_port_read_v2 {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Rx proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ * (This must be an Rx (speaker) port.)
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 buffer_address_lsw;
+/* LSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+
+ u32 buffer_address_msw;
+/* MSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+ u32 available_bytes;
+/* Number of valid bytes available in the buffer (including all
+ * channels).
+ * Supported values: > 0
+ * This field must be equal to the frame size specified in the
+ * #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure
+ * this port.
+ */
+} __packed;
+
+/* This module ID is related to device configuring like I2S,PCM,
+ * HDMI, SLIMBus etc. This module supports following parameter ids.
+ * - #AFE_PARAM_ID_I2S_CONFIG
+ * - #AFE_PARAM_ID_PCM_CONFIG
+ * - #AFE_PARAM_ID_DIGI_MIC_CONFIG
+ * - #AFE_PARAM_ID_HDMI_CONFIG
+ * - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * - #AFE_PARAM_ID_SLIMBUS_CONFIG
+ * - #AFE_PARAM_ID_RT_PROXY_CONFIG
+ */
+
+#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+#define AFE_PORT_SAMPLE_RATE_8K 8000
+#define AFE_PORT_SAMPLE_RATE_16K 16000
+#define AFE_PORT_SAMPLE_RATE_48K 48000
+#define AFE_PORT_SAMPLE_RATE_96K 96000
+#define AFE_PORT_SAMPLE_RATE_176P4K 176400
+#define AFE_PORT_SAMPLE_RATE_192K 192000
+#define AFE_PORT_SAMPLE_RATE_352P8K 352800
+#define AFE_LINEAR_PCM_DATA 0x0
+#define AFE_NON_LINEAR_DATA 0x1
+#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2
+#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3
+#define AFE_GENERIC_COMPRESSED 0x8
+
+/* This param id is used to configure I2S interface */
+#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
+#define AFE_API_VERSION_I2S_CONFIG 0x1
+/* Enumeration for setting the I2S configuration
+ * channel_mode parameter to
+ * serial data wire number 1-3 (SD3).
+ */
+#define AFE_PORT_I2S_SD0 0x1
+#define AFE_PORT_I2S_SD1 0x2
+#define AFE_PORT_I2S_SD2 0x3
+#define AFE_PORT_I2S_SD3 0x4
+#define AFE_PORT_I2S_QUAD01 0x5
+#define AFE_PORT_I2S_QUAD23 0x6
+#define AFE_PORT_I2S_6CHS 0x7
+#define AFE_PORT_I2S_8CHS 0x8
+#define AFE_PORT_I2S_MONO 0x0
+#define AFE_PORT_I2S_STEREO 0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
+
+/* Payload of the #AFE_PARAM_ID_I2S_CONFIG
+ * command's (I2S configuration
+ * parameter).
+ */
+struct afe_param_id_i2s_cfg {
+ u32 i2s_cfg_minor_version;
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+ u16 channel_mode;
+/* I2S lines and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_I2S_SD0
+ * - #AFE_PORT_I2S_SD1
+ * - #AFE_PORT_I2S_SD2
+ * - #AFE_PORT_I2S_SD3
+ * - #AFE_PORT_I2S_QUAD01
+ * - #AFE_PORT_I2S_QUAD23
+ * - #AFE_PORT_I2S_6CHS
+ * - #AFE_PORT_I2S_8CHS
+ */
+
+ u16 mono_stereo;
+/* Specifies mono or stereo. This applies only when
+ * a single I2S line is used.
+ * Supported values:
+ * - #AFE_PORT_I2S_MONO
+ * - #AFE_PORT_I2S_STEREO
+ */
+
+ u16 ws_src;
+/* Word select source: internal or external.
+ * Supported values:
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+
+ u16 data_format;
+/* data format
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+/*
+ * This param id is used to configure PCM interface
+ */
+
+#define AFE_API_VERSION_SPDIF_CONFIG 0x1
+#define AFE_API_VERSION_SPDIF_CH_STATUS_CONFIG 0x1
+#define AFE_API_VERSION_SPDIF_CLK_CONFIG 0x1
+#define AFE_CH_STATUS_A 1
+#define AFE_CH_STATUS_B 2
+
+#define AFE_PARAM_ID_SPDIF_CONFIG 0x00010244
+#define AFE_PARAM_ID_CH_STATUS_CONFIG 0x00010245
+#define AFE_PARAM_ID_SPDIF_CLK_CONFIG 0x00010246
+
+#define AFE_PORT_CLK_ROOT_LPAPLL 0x3
+#define AFE_PORT_CLK_ROOT_LPAQ6PLL 0x4
+
+struct afe_param_id_spdif_cfg {
+/* Minor version used for tracking the version of the SPDIF
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SPDIF_CONFIG
+ */
+ u32 spdif_cfg_minor_version;
+
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_22_05K
+ * - #AFE_PORT_SAMPLE_RATE_32K
+ * - #AFE_PORT_SAMPLE_RATE_44_1K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_176_4K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+ u32 sample_rate;
+
+/* data format
+ * Supported values:
+ * - #AFE_LINEAR_PCM_DATA
+ * - #AFE_NON_LINEAR_DATA
+ */
+ u16 data_format;
+/* Number of channels supported by the port
+ * - PCM - 1, Compressed Case - 2
+ */
+ u16 num_channels;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 bit_width;
+/* This field must be set to zero. */
+ u16 reserved;
+} __packed;
+
+struct afe_param_id_spdif_ch_status_cfg {
+ u32 ch_status_cfg_minor_version;
+/* Minor version used for tracking the version of channel
+ * status configuration. Current supported version is 1
+ */
+
+ u32 status_type;
+/* Indicate if the channel status is for channel A or B
+ * Supported values:
+ * - #AFE_CH_STATUS_A
+ * - #AFE_CH_STATUS_B
+ */
+
+ u8 status_bits[24];
+/* Channel status - 192 bits for channel
+ * Byte ordering as defined by IEC60958-3
+ */
+
+ u8 status_mask[24];
+/* Channel status with mask bits 1 will be applied.
+ * Byte ordering as defined by IEC60958-3
+ */
+} __packed;
+
+struct afe_param_id_spdif_clk_cfg {
+ u32 clk_cfg_minor_version;
+/* Minor version used for tracking the version of SPDIF
+ * interface clock configuration. Current supported version
+ * is 1
+ */
+
+ u32 clk_value;
+/* Specifies the clock frequency in Hz to set
+ * Supported values:
+ * 0 - Disable the clock
+ * 2 (byphase) * 32 (60958 subframe size) * sampling rate * 2
+ * (channels A and B)
+ */
+
+ u32 clk_root;
+/* Specifies SPDIF root clk source
+ * Supported Values:
+ * - #AFE_PORT_CLK_ROOT_LPAPLL
+ * - #AFE_PORT_CLK_ROOT_LPAQ6PLL
+ */
+} __packed;
+
+struct afe_spdif_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_spdif_clk_cfg clk_cfg;
+} __packed;
+
+struct afe_spdif_chstatus_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_spdif_ch_status_cfg ch_status;
+} __packed;
+
+struct afe_spdif_port_config {
+ struct afe_param_id_spdif_cfg cfg;
+ struct afe_param_id_spdif_ch_status_cfg ch_status;
+} __packed;
+
+#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E
+#define AFE_API_VERSION_PCM_CONFIG 0x1
+/* Enumeration for the auxiliary PCM synchronization signal
+ * provided by an external source.
+ */
+
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0
+/* Enumeration for the auxiliary PCM synchronization signal
+ * provided by an internal source.
+ */
+#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1
+/* Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use
+ * short synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_PCM 0x0
+/*
+ * Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use long
+ * synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_AUX 0x1
+/*
+ * Enumeration for setting the PCM configuration frame to 8.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0
+/*
+ * Enumeration for setting the PCM configuration frame to 16.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1
+
+/* Enumeration for setting the PCM configuration frame to 32.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2
+
+/* Enumeration for setting the PCM configuration frame to 64.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3
+
+/* Enumeration for setting the PCM configuration frame to 128.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4
+
+/* Enumeration for setting the PCM configuration frame to 256.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5
+
+/* Enumeration for setting the PCM configuration
+ * quantype parameter to A-law with no padding.
+ */
+#define AFE_PORT_PCM_ALAW_NOPADDING 0x0
+
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with no padding.
+ */
+#define AFE_PORT_PCM_MULAW_NOPADDING 0x1
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to linear with no padding.
+ */
+#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to A-law with padding.
+ */
+#define AFE_PORT_PCM_ALAW_PADDING 0x3
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with padding.
+ */
+#define AFE_PORT_PCM_MULAW_PADDING 0x4
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to linear with padding.
+ */
+#define AFE_PORT_PCM_LINEAR_PADDING 0x5
+/* Enumeration for disabling the PCM configuration
+ * ctrl_data_out_enable parameter.
+ * The PCM block is the only master.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0
+/*
+ * Enumeration for enabling the PCM configuration
+ * ctrl_data_out_enable parameter. The PCM block shares
+ * the signal with other masters.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1
+
+/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
+ * (PCM configuration parameter).
+ */
+
+struct afe_param_id_pcm_cfg {
+ u32 pcm_cfg_minor_version;
+/* Minor version used for tracking the version of the AUX PCM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_PCM_CONFIG
+ */
+
+ u16 aux_mode;
+/* PCM synchronization setting.
+ * Supported values:
+ * - #AFE_PORT_PCM_AUX_MODE_PCM
+ * - #AFE_PORT_PCM_AUX_MODE_AUX
+ */
+
+ u16 sync_src;
+/* Synchronization source.
+ * Supported values:
+ * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL
+ * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL
+ */
+
+ u16 frame_setting;
+/* Number of bits per frame.
+ * Supported values:
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_8
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_16
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_32
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_64
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_128
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_256
+ */
+
+ u16 quantype;
+/* PCM quantization type.
+ * Supported values:
+ * - #AFE_PORT_PCM_ALAW_NOPADDING
+ * - #AFE_PORT_PCM_MULAW_NOPADDING
+ * - #AFE_PORT_PCM_LINEAR_NOPADDING
+ * - #AFE_PORT_PCM_ALAW_PADDING
+ * - #AFE_PORT_PCM_MULAW_PADDING
+ * - #AFE_PORT_PCM_LINEAR_PADDING
+ */
+
+ u16 ctrl_data_out_enable;
+/* Specifies whether the PCM block shares the data-out
+ * signal to the drive with other masters.
+ * Supported values:
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to 4
+ */
+
+ u16 slot_number_mapping[4];
+/* Specifies the slot number for the each channel in
+ * multi channel scenario.
+ * Supported values: 1 to 32
+ */
+} __packed;
+
+/*
+ * This param id is used to configure DIGI MIC interface
+ */
+#define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 0.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1
+
+/*Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 0.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 0.
+ */
+#define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 1.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to quad.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7
+
+/* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's
+ * (DIGI MIC configuration
+ * parameter).
+ */
+struct afe_param_id_digi_mic_cfg {
+ u32 digi_mic_cfg_minor_version;
+/* Minor version used for tracking the version of the DIGI Mic
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 channel_mode;
+/* Digital mic and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT0
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT0
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT1
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT1
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO0
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO1
+ * - #AFE_PORT_DIGI_MIC_MODE_QUAD
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+} __packed;
+
+/* This param id is used to configure HDMI interface */
+#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
+
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command,
+ * which configures a multichannel HDMI audio interface.
+ */
+struct afe_param_id_hdmi_multi_chan_audio_cfg {
+ u32 hdmi_cfg_minor_version;
+/* Minor version used for tracking the version of the HDMI
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_HDMI_CONFIG
+ */
+
+u16 datatype;
+/* data type
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+
+u16 channel_allocation;
+/* HDMI channel allocation information for programming an HDMI
+ * frame. The default is 0 (Stereo).
+ *
+ * This information is defined in the HDMI standard, CEA 861-D
+ * (refer to @xhyperref{S1,[S1]}). The number of channels is also
+ * inferred from this parameter.
+ */
+
+
+u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - 22050, 44100, 176400 for compressed streams
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+/* This param id is used to configure BT or FM(RIVA) interface */
+#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211
+
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * command's BT voice/BT audio/FM configuration parameter.
+ */
+struct afe_param_id_internal_bt_fm_cfg {
+ u32 bt_fm_cfg_minor_version;
+/* Minor version used for tracking the version of the BT and FM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to 2
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP)
+ */
+} __packed;
+
+/* This param id is used to configure SLIMBUS interface using
+ * shared channel approach.
+ */
+
+
+#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
+
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+
+/* Enumeration for setting SLIMbus device ID 1. */
+#define AFE_SLIMBUS_DEVICE_1 0x0
+
+/* Enumeration for setting SLIMbus device ID 2. */
+#define AFE_SLIMBUS_DEVICE_2 0x1
+
+/* Enumeration for setting the SLIMbus data formats. */
+#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0
+
+/* Enumeration for setting the maximum number of streams per
+ * device.
+ */
+
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+
+/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus
+ * port configuration parameter.
+ */
+
+struct afe_param_id_slimbus_cfg {
+ u32 sb_cfg_minor_version;
+/* Minor version used for tracking the version of the SLIMBUS
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
+ */
+
+ u16 slimbus_dev_id;
+/* SLIMbus hardware device ID, which is required to handle
+ * multiple SLIMbus hardware blocks.
+ * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
+ */
+
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+ u16 data_format;
+/* Data format supported by the SLIMbus hardware. The default is
+ * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
+ * hardware does not perform any format conversions before the data
+ * transfer.
+ */
+
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+ u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+/* Mapping of shared channel IDs (128 to 255) to which the
+ * master port is to be connected.
+ * Shared_channel_mapping[i] represents the shared channel assigned
+ * for audio channel i in multichannel audio data.
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+} __packed;
+
+
+/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS to configure
+ * USB audio device parameter. It should be used with
+ * AFE_MODULE_AUDIO_DEV_INTERFACE
+ */
+#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5
+
+
+/* ID of the parameter used to set the endianness value for the
+ * USB audio device. It should be used with
+ * AFE_MODULE_AUDIO_DEV_INTERFACE
+ */
+#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA
+
+/* Minor version used for tracking USB audio configuration */
+#define AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG 0x1
+
+/* Payload of the AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+ */
+struct afe_param_id_usb_audio_dev_params {
+/* Minor version used for tracking USB audio device parameter.
+ * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
+ */
+ u32 cfg_minor_version;
+/* Token of actual end USB aduio device */
+ u32 dev_token;
+} __packed;
+
+struct afe_param_id_usb_audio_dev_lpcm_fmt {
+/* Minor version used for tracking USB audio device parameter.
+ * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
+ */
+ u32 cfg_minor_version;
+/* Endianness of actual end USB audio device */
+ u32 endian;
+} __packed;
+
+/* ID of the parameter used by AFE_PARAM_ID_USB_AUDIO_CONFIG to configure
+ * USB audio interface. It should be used with AFE_MODULE_AUDIO_DEV_INTERFACE
+ */
+#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4
+
+/* Payload of the AFE_PARAM_ID_USB_AUDIO_CONFIG parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+ */
+struct afe_param_id_usb_audio_cfg {
+/* Minor version used for tracking USB audio device configuration.
+ * Supported values: AFE_API_MINIOR_VERSION_USB_AUDIO_CONFIG
+ */
+ u32 cfg_minor_version;
+/* Sampling rate of the port.
+ * Supported values:
+ * - AFE_PORT_SAMPLE_RATE_8K
+ * - AFE_PORT_SAMPLE_RATE_11025
+ * - AFE_PORT_SAMPLE_RATE_12K
+ * - AFE_PORT_SAMPLE_RATE_16K
+ * - AFE_PORT_SAMPLE_RATE_22050
+ * - AFE_PORT_SAMPLE_RATE_24K
+ * - AFE_PORT_SAMPLE_RATE_32K
+ * - AFE_PORT_SAMPLE_RATE_44P1K
+ * - AFE_PORT_SAMPLE_RATE_48K
+ * - AFE_PORT_SAMPLE_RATE_96K
+ * - AFE_PORT_SAMPLE_RATE_192K
+ */
+ u32 sample_rate;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 bit_width;
+/* Number of channels.
+ * Supported values: 1 and 2
+ */
+ u16 num_channels;
+/* Data format supported by the USB. The supported value is
+ * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM).
+ */
+ u16 data_format;
+/* this field must be 0 */
+ u16 reserved;
+/* device token of actual end USB aduio device */
+ u32 dev_token;
+/* endianness of this interface */
+ u32 endian;
+} __packed;
+
+struct afe_usb_audio_dev_param_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union {
+ struct afe_param_id_usb_audio_dev_params usb_dev;
+ struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt;
+ };
+} __packed;
+
+/* This param id is used to configure Real Time Proxy interface. */
+#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213
+
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG
+ * command (real-time proxy port configuration parameter).
+ */
+struct afe_param_id_rt_proxy_port_cfg {
+ u32 rt_proxy_cfg_minor_version;
+/* Minor version used for tracking the version of rt-proxy
+ * config interface.
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 interleaved;
+/* Specifies whether the data exchanged between the AFE
+ * interface and real-time port is interleaved.
+ * Supported values: - 0 -- Non-interleaved (samples from each
+ * channel are contiguous in the buffer) - 1 -- Interleaved
+ * (corresponding samples from each input channel are interleaved
+ * within the buffer)
+ */
+
+
+ u16 frame_size;
+/* Size of the frames that are used for PCM exchanges with this
+ * port.
+ * Supported values: > 0, in bytes
+ * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples
+ * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320
+ * bytes.
+ */
+ u16 jitter_allowance;
+/* Configures the amount of jitter that the port will allow.
+ * Supported values: > 0
+ * For example, if +/-10 ms of jitter is anticipated in the timing
+ * of sending frames to the port, and the configuration is 16 kHz
+ * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2
+ * bytes/sample = 320.
+ */
+
+ u16 low_water_mark;
+/* Low watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any low watermark events
+ * - > 0 -- Low watermark for triggering an event
+ * If the number of bytes in an internal circular buffer is lower
+ * than this low_water_mark parameter, a LOW_WATER_MARK event is
+ * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * Use of watermark events is optional for debugging purposes.
+ */
+
+ u16 high_water_mark;
+/* High watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any high watermark events
+ * - > 0 -- High watermark for triggering an event
+ * If the number of bytes in an internal circular buffer exceeds
+ * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event
+ * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * The use of watermark events is optional and for debugging
+ * purposes.
+ */
+
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+ u16 reserved;
+ /* For 32 bit alignment. */
+} __packed;
+
+
+/* This param id is used to configure the Pseudoport interface */
+
+#define AFE_PARAM_ID_PSEUDO_PORT_CONFIG 0x00010219
+
+/* Version information used to handle future additions to the configuration
+ * interface (for backward compatibility).
+ */
+#define AFE_API_VERSION_PSEUDO_PORT_CONFIG 0x1
+
+/* Enumeration for setting the timing_mode parameter to faster than real
+ * time.
+ */
+#define AFE_PSEUDOPORT_TIMING_MODE_FTRT 0x0
+
+/* Enumeration for setting the timing_mode parameter to real time using
+ * timers.
+ */
+#define AFE_PSEUDOPORT_TIMING_MODE_TIMER 0x1
+
+/* Payload of the AFE_PARAM_ID_PSEUDO_PORT_CONFIG parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+ */
+struct afe_param_id_pseudo_port_cfg {
+ u32 pseud_port_cfg_minor_version;
+ /*
+ * Minor version used for tracking the version of the pseudoport
+ * configuration interface.
+ */
+
+ u16 bit_width;
+ /* Bit width of the sample at values 16, 24 */
+
+ u16 num_channels;
+ /* Number of channels at values 1 to 8 */
+
+ u16 data_format;
+ /* Non-linear data format supported by the pseudoport (for future use).
+ * At values #AFE_LINEAR_PCM_DATA
+ */
+
+ u16 timing_mode;
+ /* Indicates whether the pseudoport synchronizes to the clock or
+ * operates faster than real time.
+ * at values
+ * - #AFE_PSEUDOPORT_TIMING_MODE_FTRT
+ * - #AFE_PSEUDOPORT_TIMING_MODE_TIMER @tablebulletend
+ */
+
+ u32 sample_rate;
+ /* Sample rate at which the pseudoport will run.
+ * at values
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_32K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K @tablebulletend
+ */
+} __packed;
+
+#define AFE_PARAM_ID_TDM_CONFIG 0x0001029D
+
+#define AFE_API_VERSION_TDM_CONFIG 1
+
+#define AFE_PORT_TDM_SHORT_SYNC_BIT_MODE 0
+#define AFE_PORT_TDM_LONG_SYNC_MODE 1
+#define AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE 2
+
+#define AFE_PORT_TDM_SYNC_SRC_EXTERNAL 0
+#define AFE_PORT_TDM_SYNC_SRC_INTERNAL 1
+
+#define AFE_PORT_TDM_CTRL_DATA_OE_DISABLE 0
+#define AFE_PORT_TDM_CTRL_DATA_OE_ENABLE 1
+
+#define AFE_PORT_TDM_SYNC_NORMAL 0
+#define AFE_PORT_TDM_SYNC_INVERT 1
+
+#define AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE 0
+#define AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE 1
+#define AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE 2
+
+/* Payload of the AFE_PARAM_ID_TDM_CONFIG parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+ */
+struct afe_param_id_tdm_cfg {
+ u32 tdm_cfg_minor_version;
+ /* < Minor version used to track TDM configuration.
+ * @values #AFE_API_VERSION_TDM_CONFIG
+ */
+
+ u32 num_channels;
+ /* < Number of enabled slots for TDM frame.
+ * @values 1 to 8
+ */
+
+ u32 sample_rate;
+ /* < Sampling rate of the port.
+ * @values
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_24K
+ * - #AFE_PORT_SAMPLE_RATE_32K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_176P4K
+ * - #AFE_PORT_SAMPLE_RATE_352P8K @tablebulletend
+ */
+
+ u32 bit_width;
+ /* < Bit width of the sample.
+ * @values 16, 24
+ */
+
+ u16 data_format;
+ /* < Data format: linear ,compressed, generic compresssed
+ * @values
+ * - #AFE_LINEAR_PCM_DATA
+ * - #AFE_NON_LINEAR_DATA
+ * - #AFE_GENERIC_COMPRESSED
+ */
+
+ u16 sync_mode;
+ /* < TDM synchronization setting.
+ * @values (short, long, slot) sync mode
+ * - #AFE_PORT_TDM_SHORT_SYNC_BIT_MODE
+ * - #AFE_PORT_TDM_LONG_SYNC_MODE
+ * - #AFE_PORT_TDM_SHORT_SYNC_SLOT_MODE @tablebulletend
+ */
+
+ u16 sync_src;
+ /* < Synchronization source.
+ * @values
+ * - #AFE_PORT_TDM_SYNC_SRC_EXTERNAL
+ * - #AFE_PORT_TDM_SYNC_SRC_INTERNAL @tablebulletend
+ */
+
+ u16 nslots_per_frame;
+ /* < Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
+ * @values 1 - 32
+ */
+
+ u16 ctrl_data_out_enable;
+ /* < Specifies whether the TDM block shares the data-out signal to the
+ * drive with other masters.
+ * @values
+ * - #AFE_PORT_TDM_CTRL_DATA_OE_DISABLE
+ * - #AFE_PORT_TDM_CTRL_DATA_OE_ENABLE @tablebulletend
+ */
+
+ u16 ctrl_invert_sync_pulse;
+ /* < Specifies whether to invert the sync or not.
+ * @values
+ * - #AFE_PORT_TDM_SYNC_NORMAL
+ * - #AFE_PORT_TDM_SYNC_INVERT @tablebulletend
+ */
+
+ u16 ctrl_sync_data_delay;
+ /* < Specifies the number of bit clock to delay data with respect to
+ * sync edge.
+ * @values
+ * - #AFE_PORT_TDM_DATA_DELAY_0_BCLK_CYCLE
+ * - #AFE_PORT_TDM_DATA_DELAY_1_BCLK_CYCLE
+ * - #AFE_PORT_TDM_DATA_DELAY_2_BCLK_CYCLE @tablebulletend
+ */
+
+ u16 slot_width;
+ /* < Slot width of the slot in a TDM frame. (slot_width >= bit_width)
+ * have to be satisfied.
+ * @values 16, 24, 32
+ */
+
+ u32 slot_mask;
+ /* < Position of active slots. When that bit is set,
+ * that paricular slot is active.
+ * Number of active slots can be inferred by number of
+ * bits set in the mask. Only 8 individual bits can be enabled.
+ * Bits 0..31 corresponding to slot 0..31
+ * @values 1 to 2^32 - 1
+ */
+} __packed;
+
+/* ID of Time Divsion Multiplexing (TDM) module,
+ * which is used for configuring the AFE TDM.
+ *
+ * This module supports following parameter IDs:
+ * - #AFE_PORT_TDM_SLOT_CONFIG
+ *
+ * To configure the TDM interface, the client must use the
+ * #AFE_PORT_CMD_SET_PARAM command, and fill the module ID with the
+ * respective parameter IDs as listed above.
+ */
+
+#define AFE_MODULE_TDM 0x0001028A
+
+/* ID of the parameter used by #AFE_MODULE_TDM to configure
+ * the TDM slot mapping. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
+ */
+#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297
+
+/* Version information used to handle future additions to slot mapping
+ * configuration (for backward compatibility).
+ */
+#define AFE_API_VERSION_SLOT_MAPPING_CONFIG 0x1
+
+/* Data align type */
+#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB 0
+#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB 1
+
+#define AFE_SLOT_MAPPING_OFFSET_INVALID 0xFFFF
+
+/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG
+ * command's TDM configuration parameter.
+ */
+struct afe_param_id_slot_mapping_cfg {
+ u32 minor_version;
+ /* < Minor version used for tracking TDM slot configuration.
+ * @values #AFE_API_VERSION_TDM_SLOT_CONFIG
+ */
+
+ u16 num_channel;
+ /* < number of channel of the audio sample.
+ * @values 1, 2, 4, 6, 8 @tablebulletend
+ */
+
+ u16 bitwidth;
+ /* < Slot bit width for each channel
+ * @values 16, 24, 32
+ */
+
+ u32 data_align_type;
+ /* < indicate how data packed from slot_offset for 32 slot bit width
+ * in case of sample bit width is 24.
+ * @values
+ * #AFE_SLOT_MAPPING_DATA_ALIGN_MSB
+ * #AFE_SLOT_MAPPING_DATA_ALIGN_LSB
+ */
+
+ u16 offset[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+ /* < Array of the slot mapping start offset in bytes for this frame.
+ * The bytes is counted from 0. The 0 is mapped to the 1st byte
+ * in or out of the digital serial data line this sub-frame belong to.
+ * slot_offset[] setting is per-channel based.
+ * The max num of channel supported is 8.
+ * The valid offset value must always be continuly placed in from
+ * index 0.
+ * Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays.
+ * If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24,
+ * "data_align_type" is used to indicate how 24 bit sample data in
+ * aligning with 32 bit slot width per-channel.
+ * @values, in byte
+ */
+} __packed;
+
+/* ID of the parameter used by #AFE_MODULE_TDM to configure
+ * the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.
+ */
+#define AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG 0x00010298
+
+/* Version information used to handle future additions to custom TDM header
+ * configuration (for backward compatibility).
+ */
+#define AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG 0x1
+
+#define AFE_CUSTOM_TDM_HEADER_TYPE_INVALID 0x0
+#define AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT 0x1
+#define AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST 0x2
+
+#define AFE_CUSTOM_TDM_HEADER_MAX_CNT 0x8
+
+/* Payload of the AFE_PARAM_ID_CUSTOM_TDM_HEADER_CONFIG parameter ID */
+struct afe_param_id_custom_tdm_header_cfg {
+ u32 minor_version;
+ /* < Minor version used for tracking custom TDM header configuration.
+ * @values #AFE_API_VERSION_CUSTOM_TDM_HEADER_CONFIG
+ */
+
+ u16 start_offset;
+ /* < the slot mapping start offset in bytes from this sub-frame
+ * The bytes is counted from 0. The 0 is mapped to the 1st byte in or
+ * out of the digital serial data line this sub-frame belong to.
+ * @values, in byte,
+ * supported values are 0, 4, 8
+ */
+
+ u16 header_width;
+ /* < the header width per-frame followed.
+ * 2 bytes for MOST/TDM case
+ * @values, in byte
+ * supported value is 2
+ */
+
+ u16 header_type;
+ /* < Indicate what kind of custom TDM header it is.
+ * @values #AFE_CUSTOM_TDM_HEADER_TYPE_INVALID = 0
+ * #AFE_CUSTOM_TDM_HEADER_TYPE_DEFAULT = 1 (for AAN channel per MOST)
+ * #AFE_CUSTOM_TDM_HEADER_TYPE_ENTERTAINMENT_MOST = 2
+ * (for entertainment channel, which will overwrite
+ * AFE_API_VERSION_TDM_SAD_HEADER_TYPE_DEFAULT per MOST)
+ */
+
+ u16 num_frame_repeat;
+ /* < num of header followed.
+ * @values, supported value is 8
+ */
+ u16 header[AFE_CUSTOM_TDM_HEADER_MAX_CNT];
+ /* < SAD header for MOST/TDM case is followed as payload as below.
+ * The size of followed SAD header in bytes is num_of_frame_repeat *
+ * header_width_per_frame, which is 2 * 8 = 16 bytes here.
+ * the supported payload format is in uint16_t as below
+ * uint16_t header0; SyncHi 0x3C Info[4] - CodecType -> 0x3C00
+ * uint16_t header1; SyncLo 0xB2 Info[5] - SampleWidth -> 0xB218
+ * uint16_t header2; DTCP Info Info[6] - unused -> 0x0
+ * uint16_t header3; Extension Info[7] - ASAD-Value -> 0xC0
+ * uint16_t header4; Reserved Info[0] - Num of bytes following -> 0x7
+ * uint16_t header5; Reserved Info[1] - Media Type -> 0x0
+ * uint16_t header6; Reserved Info[2] - Bitrate[kbps] - High Byte -> 0x0
+ * uint16_t header7; Reserved Info[3] - Bitrate[kbps] - Low Byte -> 0x0
+ */
+} __packed;
+
+struct afe_slot_mapping_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_slot_mapping_cfg slot_mapping;
+} __packed;
+
+struct afe_custom_tdm_header_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_custom_tdm_header_cfg custom_tdm_header;
+} __packed;
+
+struct afe_tdm_port_config {
+ struct afe_param_id_tdm_cfg tdm;
+ struct afe_param_id_slot_mapping_cfg slot_mapping;
+ struct afe_param_id_custom_tdm_header_cfg custom_tdm_header;
+} __packed;
+
+#define AFE_PARAM_ID_DEVICE_HW_DELAY 0x00010243
+#define AFE_API_VERSION_DEVICE_HW_DELAY 0x1
+
+struct afe_param_id_device_hw_delay_cfg {
+ uint32_t device_hw_delay_minor_version;
+ uint32_t delay_in_us;
+} __packed;
+
+#define AFE_PARAM_ID_SET_TOPOLOGY 0x0001025A
+#define AFE_API_VERSION_TOPOLOGY_V1 0x1
+
+struct afe_param_id_set_topology_cfg {
+ /*
+ * Minor version used for tracking afe topology id configuration.
+ * @values #AFE_API_VERSION_TOPOLOGY_V1
+ */
+ u32 minor_version;
+ /*
+ * Id of the topology for the afe session.
+ * @values Any valid AFE topology ID
+ */
+ u32 topology_id;
+} __packed;
+
+
+/*
+ * Generic encoder module ID.
+ * This module supports the following parameter IDs:
+ * #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time)
+ * Opcode - AVS_MODULE_ID_ENCODER
+ * AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID.
+ */
+#define AFE_MODULE_ID_ENCODER 0x00013229
+
+/* Macro for defining the packetizer ID: COP. */
+#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A
+
+/*
+ * Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E
+
+/*
+ * Encoder config block parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter may be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C
+
+/*
+ * Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID 0x0001322B
+
+/*
+ * Data format to send compressed data
+ * is transmitted/received over Slimbus lines.
+ */
+#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED 0x3
+
+/*
+ * ID for AFE port module. This will be used to define port properties.
+ * This module supports following parameter IDs:
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ * To configure the port property, the client must use the
+ * #AFE_PORT_CMD_SET_PARAM_V2 command,
+ * and fill the module ID with the respective parameter IDs as listed above.
+ * @apr_hdr_fields
+ * Opcode -- AFE_MODULE_PORT
+ */
+#define AFE_MODULE_PORT 0x000102a6
+
+/*
+ * ID of the parameter used by #AFE_MODULE_PORT to set the port media type.
+ * parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command.
+ */
+#define AFE_PARAM_ID_PORT_MEDIA_TYPE 0x000102a7
+
+/*
+ * Macros for defining the "data_format" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_PORT_DATA_FORMAT_PCM 0x0
+#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED 0x1
+
+/*
+ * Macro for defining the "minor_version" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_API_VERSION_PORT_MEDIA_TYPE 0x1
+
+#define ASM_MEDIA_FMT_NONE 0x0
+
+/*
+ * Media format ID for SBC encode configuration.
+ * @par SBC encode configuration (asm_sbc_enc_cfg_t)
+ * @table{weak__asm__sbc__enc__cfg__t}
+ */
+#define ASM_MEDIA_FMT_SBC 0x00010BF2
+
+/* SBC channel Mono mode.*/
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO 1
+
+/* SBC channel Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO 2
+
+/* SBC channel Dual Mono mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO 8
+
+/* SBC channel Joint Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9
+
+/* SBC bit allocation method = loudness. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0
+
+/* SBC bit allocation method = SNR. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1
+
+
+/*
+ * Payload of the SBC encoder configuration parameters in the
+ * #ASM_MEDIA_FMT_SBC media format.
+ */
+struct asm_sbc_enc_cfg_t {
+ /*
+ * Number of subbands.
+ * @values 4, 8
+ */
+ uint32_t num_subbands;
+
+ /*
+ * Size of the encoded block in samples.
+ * @values 4, 8, 12, 16
+ */
+ uint32_t blk_len;
+
+ /*
+ * Mode used to allocate bits between channels.
+ * @values
+ * 0 (Native mode)
+ * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO
+ * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO
+ * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO
+ * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO
+ * Native mode indicates that encoding must be performed with the number
+ * of channels at the input.
+ * If postprocessing outputs one-channel data, Mono mode is used. If
+ * postprocessing outputs two-channel data, Stereo mode is used.
+ * The number of channels must not change during encoding.
+ */
+ uint32_t channel_mode;
+
+ /*
+ * Encoder bit allocation method.
+ * @values
+ * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS
+ * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend
+ */
+ uint32_t alloc_method;
+
+ /*
+ * Number of encoded bits per second.
+ * @values
+ * Mono channel -- Maximum of 320 kbps
+ * Stereo channel -- Maximum of 512 kbps @tablebulletend
+ */
+ uint32_t bit_rate;
+
+ /*
+ * Number of samples per second.
+ * @values 0 (Native mode), 16000, 32000, 44100, 48000 Hz
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+ uint32_t sample_rate;
+};
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC 2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
+#define ASM_MEDIA_FMT_AAC_AOT_PS 29
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
+
+struct asm_aac_enc_cfg_v2_t {
+
+ /* Encoding rate in bits per second.*/
+ uint32_t bit_rate;
+
+ /*
+ * Encoding mode.
+ * Supported values:
+ * #ASM_MEDIA_FMT_AAC_AOT_LC
+ * #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * #ASM_MEDIA_FMT_AAC_AOT_PS
+ */
+ uint32_t enc_mode;
+
+ /*
+ * AAC format flag.
+ * Supported values:
+ * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+ uint16_t aac_fmt_flag;
+
+ /*
+ * Number of channels to encode.
+ * Supported values:
+ * 0 - Native mode
+ * 1 - Mono
+ * 2 - Stereo
+ * Other values are not supported.
+ * @note1hang The eAAC+ encoder mode supports only stereo.
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+ uint16_t channel_cfg;
+
+ /*
+ * Number of samples per second.
+ * Supported values: - 0 -- Native mode - For other values,
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+ uint32_t sample_rate;
+} __packed;
+
+/* FMT ID for apt-X Classic */
+#define ASM_MEDIA_FMT_APTX 0x000131ff
+
+/* FMT ID for apt-X HD */
+#define ASM_MEDIA_FMT_APTX_HD 0x00013200
+
+#define PCM_CHANNEL_L 1
+#define PCM_CHANNEL_R 2
+#define PCM_CHANNEL_C 3
+
+struct asm_custom_enc_cfg_aptx_t {
+ uint32_t sample_rate;
+ /* Mono or stereo */
+ uint16_t num_channels;
+ uint16_t reserved;
+ /* num_ch == 1, then PCM_CHANNEL_C,
+ * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
+ */
+ uint8_t channel_mapping[8];
+ uint32_t custom_size;
+} __packed;
+
+struct afe_enc_fmt_id_param_t {
+ /*
+ * Supported values:
+ * #ASM_MEDIA_FMT_SBC
+ * #ASM_MEDIA_FMT_AAC_V2
+ * Any OpenDSP supported values
+ */
+ uint32_t fmt_id;
+} __packed;
+
+struct afe_port_media_type_t {
+ /*
+ * Minor version
+ * @values #AFE_API_VERSION_PORT_MEDIA_TYPE.
+ */
+ uint32_t minor_version;
+
+ /*
+ * Sampling rate of the port.
+ * @values
+ * #AFE_PORT_SAMPLE_RATE_8K
+ * #AFE_PORT_SAMPLE_RATE_11_025K
+ * #AFE_PORT_SAMPLE_RATE_12K
+ * #AFE_PORT_SAMPLE_RATE_16K
+ * #AFE_PORT_SAMPLE_RATE_22_05K
+ * #AFE_PORT_SAMPLE_RATE_24K
+ * #AFE_PORT_SAMPLE_RATE_32K
+ * #AFE_PORT_SAMPLE_RATE_44_1K
+ * #AFE_PORT_SAMPLE_RATE_48K
+ * #AFE_PORT_SAMPLE_RATE_88_2K
+ * #AFE_PORT_SAMPLE_RATE_96K
+ * #AFE_PORT_SAMPLE_RATE_176_4K
+ * #AFE_PORT_SAMPLE_RATE_192K
+ * #AFE_PORT_SAMPLE_RATE_352_8K
+ * #AFE_PORT_SAMPLE_RATE_384K
+ */
+ uint32_t sample_rate;
+
+ /*
+ * Bit width of the sample.
+ * @values 16, 24
+ */
+ uint16_t bit_width;
+
+ /*
+ * Number of channels.
+ * @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+ uint16_t num_channels;
+
+ /*
+ * Data format supported by this port.
+ * If the port media type and device media type are different,
+ * it signifies a encoding/decoding use case
+ * @values
+ * #AFE_PORT_DATA_FORMAT_PCM
+ * #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED
+ */
+ uint16_t data_format;
+
+ /*This field must be set to zero.*/
+ uint16_t reserved;
+} __packed;
+
+union afe_enc_config_data {
+ struct asm_sbc_enc_cfg_t sbc_config;
+ struct asm_aac_enc_cfg_v2_t aac_config;
+ struct asm_custom_enc_cfg_aptx_t aptx_config;
+};
+
+struct afe_enc_config {
+ u32 format;
+ union afe_enc_config_data data;
+};
+
+struct afe_enc_cfg_blk_param_t {
+ uint32_t enc_cfg_blk_size;
+ /*
+ *Size of the encoder configuration block that follows this member
+ */
+ union afe_enc_config_data enc_blk_config;
+};
+
+/*
+ * Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
+ */
+struct avs_enc_packetizer_id_param_t {
+ /*
+ * Supported values:
+ * #AVS_MODULE_ID_PACKETIZER_COP
+ * Any OpenDSP supported values
+ */
+ uint32_t enc_packetizer_id;
+};
+
+union afe_port_config {
+ struct afe_param_id_pcm_cfg pcm;
+ struct afe_param_id_i2s_cfg i2s;
+ struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
+ struct afe_param_id_slimbus_cfg slim_sch;
+ struct afe_param_id_rt_proxy_port_cfg rtproxy;
+ struct afe_param_id_internal_bt_fm_cfg int_bt_fm;
+ struct afe_param_id_pseudo_port_cfg pseudo_port;
+ struct afe_param_id_device_hw_delay_cfg hw_delay;
+ struct afe_param_id_spdif_cfg spdif;
+ struct afe_param_id_set_topology_cfg topology;
+ struct afe_param_id_tdm_cfg tdm;
+ struct afe_param_id_usb_audio_cfg usb_audio;
+ struct afe_enc_fmt_id_param_t enc_fmt;
+ struct afe_port_media_type_t media_type;
+ struct afe_enc_cfg_blk_param_t enc_blk_param;
+ struct avs_enc_packetizer_id_param_t enc_pkt_id_param;
+} __packed;
+
+struct afe_audioif_config_command_no_payload {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+} __packed;
+
+struct afe_audioif_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union afe_port_config port;
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+
+/* Payload of the #AFE_PORT_CMD_DEVICE_START.*/
+struct afe_port_cmd_device_start {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
+
+/* Payload of the #AFE_PORT_CMD_DEVICE_STOP. */
+struct afe_port_cmd_device_stop {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+} __packed;
+
+#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA
+
+/* Memory map regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS .
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of afe_service_shared_map_region_payload.
+ */
+struct afe_service_cmd_shared_mem_map_regions {
+ struct apr_hdr hdr;
+u16 mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ * Supported values:
+ * - #ADSP_MEMORY_MAP_EBI_POOL
+ * - #ADSP_MEMORY_MAP_SMI_POOL
+ * - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL
+ * - Other values are reserved
+ *
+ * The memory pool ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory
+ * ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory
+ * ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte
+ * addressable, and 4 KB aligned.
+ */
+
+
+ u16 num_regions;
+/* Number of regions to map.
+ * Supported values:
+ * - Any value greater than zero
+ */
+
+ u32 property_flag;
+/* Configures one common property for all the regions in the
+ * payload.
+ *
+ * Supported values: - 0x00000000 to 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 Shared memory
+ * address provided in afe_service_shared_map_region_payloadis a
+ * physical address. The shared memory needs to be mapped( hardware
+ * TLB entry) and a software entry needs to be added for internal
+ * book keeping.
+ *
+ * 1 Shared memory address provided in
+ * afe_service_shared_map_region_payloadis a virtual address. The
+ * shared memory must not be mapped (since hardware TLB entry is
+ * already available) but a software entry needs to be added for
+ * internal book keeping. This can be useful if two services with in
+ * ADSP is communicating via APR. They can now directly communicate
+ * via the Virtual address instead of Physical address. The virtual
+ * regions must be contiguous. num_regions must be 1 in this case.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+
+} __packed;
+/* Map region payload used by the
+ * afe_service_shared_map_region_payloadstructure.
+ */
+struct afe_service_shared_map_region_payload {
+ u32 shm_addr_lsw;
+/* least significant word of starting address in the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ * Supported values: - Any 32 bit value
+ */
+
+
+ u32 shm_addr_msw;
+/* most significant word of startng address in the memory region
+ * to map. For 32 bit shared memory address, this field must be set
+ * to zero. For 36 bit shared memory address, bit31 to bit 4 must be
+ * set to zero
+ *
+ * Supported values: - For 32 bit shared memory address, this field
+ * must be set to zero. - For 36 bit shared memory address, bit31 to
+ * bit 4 must be set to zero - For 64 bit shared memory address, any
+ * 32 bit value
+ */
+
+
+ u32 mem_size_bytes;
+/* Number of bytes in the region. The aDSP will always map the
+ * regions as virtual contiguous memory, but the memory size must be
+ * in multiples of 4 KB to avoid gaps in the virtually contiguous
+ * mapped memory.
+ *
+ * Supported values: - multiples of 4KB
+ */
+
+} __packed;
+
+#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB
+struct afe_service_cmdrsp_shared_mem_map_regions {
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful. In the case of failure , a generic APR error response
+ * is returned to the client.
+ *
+ * Supported Values: - Any 32 bit value
+ */
+
+} __packed;
+#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC
+/* Memory unmap regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS
+ *
+ * This structure allows clients to unmap multiple shared memory
+ * regions in a single command.
+ */
+
+
+struct afe_service_cmd_shared_mem_unmap_regions {
+ struct apr_hdr hdr;
+u32 mem_map_handle;
+/* memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands
+ *
+ * Supported Values:
+ * - Any 32 bit value
+ */
+} __packed;
+
+#define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0
+
+/* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command,
+ * which queries for one post/preprocessing parameter of a
+ * stream.
+ */
+struct afe_port_cmd_get_param_v2 {
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. */
+
+ u16 payload_size;
+/* Maximum data size of the parameter ID/module ID combination.
+ * This is a multiple of four bytes
+ * Supported values: > 0
+ */
+
+ u32 payload_address_lsw;
+/* LSW of 64 bit Payload address. Address should be 32-byte,
+ * 4kbyte aligned and must be contig memory.
+ */
+
+
+ u32 payload_address_msw;
+/* MSW of 64 bit Payload address. In case of 32-bit shared
+ * memory address, this field must be set to zero. In case of 36-bit
+ * shared memory address, bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned and must be contiguous
+ * memory.
+ */
+
+ u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values: - NULL -- Message. The parameter data is
+ * in-band. - Non-NULL -- The parameter data is Out-band.Pointer to
+ * - the physical address in shared memory of the payload data.
+ * For detailed payload content, see the afe_port_param_data_v2
+ * structure
+ */
+
+
+ u32 module_id;
+/* ID of the module to be queried.
+ * Supported values: Valid module ID
+ */
+
+ u32 param_id;
+/* ID of the parameter to be queried.
+ * Supported values: Valid parameter ID
+ */
+} __packed;
+
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+
+/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which
+ * responds to an #AFE_PORT_CMD_GET_PARAM_V2 command.
+ *
+ * Immediately following this structure is the parameters structure
+ * (afe_port_param_data) containing the response(acknowledgment)
+ * parameter payload. This payload is included for an in-band
+ * scenario. For an address/shared memory-based set parameter, this
+ * payload is not needed.
+ */
+
+
+struct afe_port_cmdrsp_get_param_v2 {
+ u32 status;
+} __packed;
+
+#define AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG 0x0001028C
+#define AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG 0x1
+
+/* Payload of the AFE_PARAM_ID_LPASS_CORE_SHARED_CLOCK_CONFIG parameter used by
+ * AFE_MODULE_AUDIO_DEV_INTERFACE.
+ */
+struct afe_param_id_lpass_core_shared_clk_cfg {
+ u32 lpass_core_shared_clk_cfg_minor_version;
+/*
+ * Minor version used for lpass core shared clock configuration
+ * Supported value: AFE_API_VERSION_LPASS_CORE_SHARED_CLK_CONFIG
+ */
+ u32 enable;
+/*
+ * Specifies whether the lpass core shared clock is
+ * enabled (1) or disabled (0).
+ */
+} __packed;
+
+struct afe_lpass_core_shared_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_lpass_core_shared_clk_cfg clk_cfg;
+} __packed;
+
+/* adsp_afe_service_commands.h */
+
+#define ADSP_MEMORY_MAP_EBI_POOL 0
+
+#define ADSP_MEMORY_MAP_SMI_POOL 1
+#define ADSP_MEMORY_MAP_IMEM_POOL 2
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
+
+/* Definition of virtual memory flag */
+#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1
+
+/* Definition of physical memory flag */
+#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0
+
+#define NULL_POPP_TOPOLOGY 0x00010C68
+#define NULL_COPP_TOPOLOGY 0x00010312
+#define DEFAULT_COPP_TOPOLOGY 0x00010314
+#define DEFAULT_POPP_TOPOLOGY 0x00010BE4
+#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY 0x0001076B
+#define COMPRESSED_PASSTHROUGH_NONE_TOPOLOGY 0x00010774
+#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
+#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
+#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75
+#define VPM_TX_DM_RFECNS_COPP_TOPOLOGY 0x00010F86
+#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_DTS_HPX 0x10015002
+#define ADM_CMD_COPP_OPEN_TOPOLOGY_ID_AUDIOSPHERE 0x10028000
+
+/* Memory map regions command payload used by the
+ * #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * commands.
+ *
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of avs_shared_map_region_payload.
+ */
+
+
+struct avs_cmd_shared_mem_map_regions {
+ struct apr_hdr hdr;
+ u16 mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ *
+ * Supported values: - #ADSP_MEMORY_MAP_EBI_POOL -
+ * #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL
+ * (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values
+ * are reserved
+ *
+ * The memory ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned.
+ */
+
+
+ u16 num_regions;
+ /* Number of regions to map.*/
+
+ u32 property_flag;
+/* Configures one common property for all the regions in the
+ * payload. No two regions in the same memory map regions cmd can
+ * have differnt property. Supported values: - 0x00000000 to
+ * 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 shared memory
+ * address provided in avs_shared_map_regions_payload is physical
+ * address. The shared memory needs to be mapped( hardware TLB
+ * entry)
+ *
+ * and a software entry needs to be added for internal book keeping.
+ *
+ * 1 Shared memory address provided in MayPayload[usRegions] is
+ * virtual address. The shared memory must not be mapped (since
+ * hardware TLB entry is already available) but a software entry
+ * needs to be added for internal book keeping. This can be useful
+ * if two services with in ADSP is communicating via APR. They can
+ * now directly communicate via the Virtual address instead of
+ * Physical address. The virtual regions must be contiguous.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+} __packed;
+
+struct avs_shared_map_region_payload {
+ u32 shm_addr_lsw;
+/* least significant word of shared memory address of the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ */
+
+ u32 shm_addr_msw;
+/* most significant word of shared memory address of the memory
+ * region to map. For 32 bit shared memory address, this field must
+ * tbe set to zero. For 36 bit shared memory address, bit31 to bit 4
+ * must be set to zero
+ */
+
+ u32 mem_size_bytes;
+/* Number of bytes in the region.
+ *
+ * The aDSP will always map the regions as virtual contiguous
+ * memory, but the memory size must be in multiples of 4 KB to avoid
+ * gaps in the virtually contiguous mapped memory.
+ */
+
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+ struct apr_hdr hdr;
+ u32 mem_map_handle;
+/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS
+ * , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands
+ */
+
+} __packed;
+
+/* Memory map command response payload used by the
+ * #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ * ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ */
+
+
+struct avs_cmdrsp_shared_mem_map_regions {
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned
+ */
+
+} __packed;
+
+/*adsp_audio_memmap_api.h*/
+
+/* ASM related data structures */
+struct asm_wma_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+} __packed;
+
+struct asm_wmapro_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+} __packed;
+
+struct asm_aac_cfg {
+ u16 format;
+ u16 aot;
+ u16 ep_config;
+ u16 section_data_resilience;
+ u16 scalefactor_data_resilience;
+ u16 spectral_data_resilience;
+ u16 ch_cfg;
+ u16 reserved;
+ u32 sample_rate;
+} __packed;
+
+struct asm_amrwbplus_cfg {
+ u32 size_bytes;
+ u32 version;
+ u32 num_channels;
+ u32 amr_band_mode;
+ u32 amr_dtx_mode;
+ u32 amr_frame_fmt;
+ u32 amr_lsf_idx;
+} __packed;
+
+struct asm_flac_cfg {
+ u32 sample_rate;
+ u32 ext_sample_rate;
+ u32 min_frame_size;
+ u32 max_frame_size;
+ u16 stream_info_present;
+ u16 min_blk_size;
+ u16 max_blk_size;
+ u16 ch_cfg;
+ u16 sample_size;
+ u16 md5_sum;
+};
+
+struct asm_alac_cfg {
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+};
+
+struct asm_g711_dec_cfg {
+ u32 sample_rate;
+};
+
+struct asm_vorbis_cfg {
+ u32 bit_stream_fmt;
+};
+
+struct asm_ape_cfg {
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+};
+
+struct asm_dsd_cfg {
+ u16 num_version;
+ u16 is_bitwise_big_endian;
+ u16 dsd_channel_block_size;
+ u16 num_channels;
+ u8 channel_mapping[8];
+ u32 dsd_data_rate;
+};
+
+struct asm_softpause_params {
+ u32 enable;
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+struct asm_softvolume_params {
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+#define ASM_END_POINT_DEVICE_MATRIX 0
+
+#define PCM_CHANNEL_NULL 0
+
+/* Front left channel. */
+#define PCM_CHANNEL_FL 1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR 2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC 3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS 4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS 5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE 6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS 7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB 8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB 9
+
+/* Top surround channel. */
+#define PCM_CHANNELS 10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH 11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS 12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC 13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC 14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC 15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC 16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC
+
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C
+
+#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
+
+#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
+
+#define ASM_MEDIA_FMT_GENERIC_COMPRESSED 0x00013212
+
+#define ASM_MAX_EQ_BANDS 12
+
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+
+struct asm_data_cmd_media_fmt_update_v2 {
+u32 fmt_blk_size;
+ /* Media format block size in bytes.*/
+} __packed;
+
+struct asm_generic_compressed_fmt_blk_t {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+ /*
+ * Channel mapping array of bitstream output.
+ * Channel[i] mapping describes channel i inside the buffer, where
+ * i < num_channels. All valid used channels must be
+ * present at the beginning of the array.
+ */
+ uint8_t channel_mapping[8];
+
+ /*
+ * Number of channels of the incoming bitstream.
+ * Supported values: 1,2,3,4,5,6,7,8
+ */
+ uint16_t num_channels;
+
+ /*
+ * Nominal bits per sample value of the incoming bitstream.
+ * Supported values: 16, 32
+ */
+ uint16_t bits_per_sample;
+
+ /*
+ * Nominal sampling rate of the incoming bitstream.
+ * Supported values: 8000, 11025, 16000, 22050, 24000, 32000,
+ * 44100, 48000, 88200, 96000, 176400, 192000,
+ * 352800, 384000
+ */
+ uint32_t sampling_rate;
+
+} __packed;
+
+
+/* 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;
+
+/*
+ * Payload of the multichannel PCM configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
+ */
+struct asm_multi_channel_pcm_fmt_blk_param_v3 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ struct asm_multi_channel_pcm_fmt_blk_v3 param;
+} __packed;
+
+/*
+ * Payload of the multichannel PCM configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
+ */
+struct asm_multi_channel_pcm_fmt_blk_param_v4 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ struct asm_multi_channel_pcm_fmt_blk_v4 param;
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+ u32 param_id;
+ /* ID of the parameter. */
+
+ u32 param_size;
+/* Data size of this parameter, in bytes. The size is a multiple
+ * of 4 bytes.
+ */
+
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+ u32 frames_per_buf;
+/* Number of encoded frames to pack into each buffer.
+ *
+ * @note1hang This is only guidance information for the aDSP. The
+ * number of encoded frames put into each buffer (specified by the
+ * client) is less than or equal to this number.
+ */
+
+ u32 enc_cfg_blk_size;
+/* Size in bytes of the encoder configuration block that follows
+ * this member.
+ */
+
+} __packed;
+
+/* @brief Dolby Digital Plus end point configuration structure
+ */
+struct asm_dec_ddp_endp_param_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ int endp_param_value;
+} __packed;
+
+/*
+ * Payload of the multichannel PCM encoder configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v4 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+ /*
+ * Number of PCM channels.
+ * @values
+ * - 0 -- Native mode
+ * - 1 -- 8 channels
+ * Native mode indicates that encoding must be performed with the number
+ * of channels at the input.
+ */
+ uint16_t bits_per_sample;
+ /*
+ * Number of bits per sample per channel.
+ * @values 16, 24
+ */
+ uint32_t sample_rate;
+ /*
+ * Number of samples per second.
+ * @values 0, 8000 to 48000 Hz
+ * A value of 0 indicates the native sampling rate. Encoding is
+ * performed at the input sampling rate.
+ */
+ uint16_t is_signed;
+ /*
+ * Flag that indicates the PCM samples are signed (1). Currently, only
+ * signed PCM samples are supported.
+ */
+ uint16_t sample_word_size;
+ /*
+ * The size in bits of the word that holds a sample of a channel.
+ * @values 16, 24, 32
+ * 16-bit samples are always placed in 16-bit words:
+ * sample_word_size = 1.
+ * 24-bit samples can be placed in 32-bit words or in consecutive
+ * 24-bit words.
+ * - If sample_word_size = 32, 24-bit samples are placed in the
+ * most significant 24 bits of a 32-bit word.
+ * - If sample_word_size = 24, 24-bit samples are placed in
+ * 24-bit words. @tablebulletend
+ */
+ uint8_t channel_mapping[8];
+ /*
+ * Channel mapping array expected at the encoder output.
+ * Channel[i] mapping describes channel i inside the buffer, where
+ * 0 @le i < num_channels. All valid used channels must be present at
+ * the beginning of the array.
+ * If Native mode is set for the channels, this field is ignored.
+ * @values See Section @xref{dox:PcmChannelDefs}
+ */
+ uint16_t endianness;
+ /*
+ * Flag to indicate the endianness of the pcm sample
+ * Supported values: 0 - Little endian (all other formats)
+ * 1 - Big endian (AIFF)
+ */
+ uint16_t mode;
+ /*
+ * Mode to provide additional info about the pcm input data.
+ * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
+ * Q31 for unpacked 24b or 32b)
+ * 15 - for 16 bit
+ * 23 - for 24b packed or 8.24 format
+ * 31 - for 24b unpacked or 32bit
+ */
+} __packed;
+
+/*
+ * Payload of the multichannel PCM encoder configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v3 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+ /*
+ * Number of PCM channels.
+ * @values
+ * - 0 -- Native mode
+ * - 1 -- 8 channels
+ * Native mode indicates that encoding must be performed with the number
+ * of channels at the input.
+ */
+ uint16_t bits_per_sample;
+ /*
+ * Number of bits per sample per channel.
+ * @values 16, 24
+ */
+ uint32_t sample_rate;
+ /*
+ * Number of samples per second.
+ * @values 0, 8000 to 48000 Hz
+ * A value of 0 indicates the native sampling rate. Encoding is
+ * performed at the input sampling rate.
+ */
+ uint16_t is_signed;
+ /*
+ * Flag that indicates the PCM samples are signed (1). Currently, only
+ * signed PCM samples are supported.
+ */
+ uint16_t sample_word_size;
+ /*
+ * The size in bits of the word that holds a sample of a channel.
+ * @values 16, 24, 32
+ * 16-bit samples are always placed in 16-bit words:
+ * sample_word_size = 1.
+ * 24-bit samples can be placed in 32-bit words or in consecutive
+ * 24-bit words.
+ * - If sample_word_size = 32, 24-bit samples are placed in the
+ * most significant 24 bits of a 32-bit word.
+ * - If sample_word_size = 24, 24-bit samples are placed in
+ * 24-bit words. @tablebulletend
+ */
+ uint8_t channel_mapping[8];
+ /*
+ * Channel mapping array expected at the encoder output.
+ * Channel[i] mapping describes channel i inside the buffer, where
+ * 0 @le i < num_channels. All valid used channels must be present at
+ * the beginning of the array.
+ * If Native mode is set for the channels, this field is ignored.
+ * @values See Section @xref{dox:PcmChannelDefs}
+ */
+};
+
+/* @brief Multichannel PCM encoder configuration structure used
+ * in the #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+/*< Number of PCM channels.
+ *
+ * Supported values: - 0 -- Native mode - 1 -- 8 Native mode
+ * indicates that encoding must be performed with the number of
+ * channels at the input.
+ */
+
+ uint16_t bits_per_sample;
+/*< Number of bits per sample per channel.
+ * Supported values: 16, 24
+ */
+
+ uint32_t sample_rate;
+/*< Number of samples per second (in Hertz).
+ *
+ * Supported values: 0, 8000 to 48000 A value of 0 indicates the
+ * native sampling rate. Encoding is performed at the input sampling
+ * rate.
+ */
+
+ uint16_t is_signed;
+/*< Specifies whether the samples are signed (1). Currently,
+ * only signed samples are supported.
+ */
+
+ uint16_t reserved;
+/*< reserved field for 32 bit alignment. must be set to zero.*/
+
+
+ uint8_t channel_mapping[8];
+} __packed;
+
+#define ASM_MEDIA_FMT_MP3 0x00010BE9
+#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6
+
+/* @xreflabel
+ * {hdr:AsmMediaFmtDolbyAac} Media format ID for the
+ * Dolby AAC decoder. This format ID is be used if the client wants
+ * to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC
+ * contents.
+ */
+
+#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86
+
+/* Enumeration for the audio data transport stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
+
+/* Enumeration for low overhead audio stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS 1
+
+/* Enumeration for the audio data interchange format
+ * AAC format.
+ */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF 2
+
+/* Enumeration for the raw AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
+
+/* Enumeration for the AAC LATM format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM 4
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC 2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
+#define ASM_MEDIA_FMT_AAC_AOT_PS 29
+#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22
+
+struct asm_aac_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+ u16 aac_fmt_flag;
+/* Bitstream format option.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+
+ u16 audio_objype;
+/* Audio Object Type (AOT) present in the AAC stream.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_BSAC
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ * - Otherwise -- Not supported
+ */
+
+ u16 channel_config;
+/* Number of channels present in the AAC stream.
+ * Supported values:
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - 6 -- 5.1 content
+ */
+
+ u16 total_size_of_PCE_bits;
+/* greater or equal to zero. * -In case of RAW formats and
+ * channel config = 0 (PCE), client can send * the bit stream
+ * containing PCE immediately following this structure * (in-band).
+ * -This number does not include bits included for 32 bit alignment.
+ * -If zero, then the PCE info is assumed to be available in the
+ * audio -bit stream & not in-band.
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ *
+ * Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000,
+ * 44100, 48000
+ *
+ * This field must be equal to the sample rate of the AAC-LC
+ * decoder's output. - For MP4 or 3GP containers, this is indicated
+ * by the samplingFrequencyIndex field in the AudioSpecificConfig
+ * element. - For ADTS format, this is indicated by the
+ * samplingFrequencyIndex in the ADTS fixed header. - For ADIF
+ * format, this is indicated by the samplingFrequencyIndex in the
+ * program_config_element present in the ADIF header.
+ */
+
+} __packed;
+
+struct asm_aac_enc_cfg_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u32 bit_rate;
+ /* Encoding rate in bits per second. */
+ u32 enc_mode;
+/* Encoding mode.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ */
+ u16 aac_fmt_flag;
+/* AAC format flag.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+ u16 channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ * @note1hang The eAAC+ encoder mode supports only stereo.
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+ u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: - 0 -- Native mode - For other values,
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_G711_ALAW_FS 0x00010BF7
+#define ASM_MEDIA_FMT_G711_MLAW_FS 0x00010C2E
+
+struct asm_g711_enc_cfg_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u32 sample_rate;
+/*
+ * Number of samples per second.
+ * Supported values: 8000, 16000 Hz
+ */
+
+} __packed;
+
+struct asm_vorbis_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u32 bit_stream_fmt;
+/* Bit stream format.
+ * Supported values:
+ * - 0 -- Raw bitstream
+ * - 1 -- Transcoded bitstream
+ *
+ * Transcoded bitstream containing the size of the frame as the first
+ * word in each frame.
+ */
+
+} __packed;
+
+struct asm_flac_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u16 is_stream_info_present;
+/* Specifies whether stream information is present in the FLAC format
+ * block.
+ *
+ * Supported values:
+ * - 0 -- Stream information is not present in this message
+ * - 1 -- Stream information is present in this message
+ *
+ * When set to 1, the FLAC bitstream was successfully parsed by the
+ * client, and other fields in the FLAC format block can be read by the
+ * decoder to get metadata stream information.
+ */
+
+ u16 num_channels;
+/* Number of channels for decoding.
+ * Supported values: 1 to 2
+ */
+
+ u16 min_blk_size;
+/* Minimum block size (in samples) used in the stream. It must be less
+ * than or equal to max_blk_size.
+ */
+
+ u16 max_blk_size;
+/* Maximum block size (in samples) used in the stream. If the
+ * minimum block size equals the maximum block size, a fixed block
+ * size stream is implied.
+ */
+
+ u16 md5_sum[8];
+/* MD5 signature array of the unencoded audio data. This allows the
+ * decoder to determine if an error exists in the audio data, even when
+ * the error does not result in an invalid bitstream.
+ */
+
+ u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: 8000 to 48000 Hz
+ */
+
+ u32 min_frame_size;
+/* Minimum frame size used in the stream.
+ * Supported values:
+ * - > 0 bytes
+ * - 0 -- The value is unknown
+ */
+
+ u32 max_frame_size;
+/* Maximum frame size used in the stream.
+ * Supported values:
+ * -- > 0 bytes
+ * -- 0 . The value is unknown
+ */
+
+ u16 sample_size;
+/* Bits per sample.Supported values: 8, 16 */
+
+ u16 reserved;
+/* Clients must set this field to zero
+ */
+
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+
+} __packed;
+
+struct asm_g711_dec_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u32 sample_rate;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+
+} __packed;
+
+struct asm_dsd_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u16 num_version;
+ u16 is_bitwise_big_endian;
+ u16 dsd_channel_block_size;
+ u16 num_channels;
+ u8 channel_mapping[8];
+ u32 dsd_data_rate;
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRNB_FS 0x00010BEB
+
+/* Enumeration for 4.75 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR475 0
+
+/* Enumeration for 5.15 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR515 1
+
+/* Enumeration for 5.90 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR59 2
+
+/* Enumeration for 6.70 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR67 3
+
+/* Enumeration for 7.40 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR74 4
+
+/* Enumeration for 7.95 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR795 5
+
+/* Enumeration for 10.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR102 6
+
+/* Enumeration for 12.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR122 7
+
+/* Enumeration for AMR-NB Discontinuous Transmission mode off. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF 0
+
+/* Enumeration for AMR-NB DTX mode VAD1. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 1
+
+/* Enumeration for AMR-NB DTX mode VAD2. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD2 2
+
+/* Enumeration for AMR-NB DTX mode auto. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_AUTO 3
+
+struct asm_amrnb_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u16 enc_mode;
+/* AMR-NB encoding rate.
+ * Supported values:
+ * Use the ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+ u16 dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRWB_FS 0x00010BEC
+
+/* Enumeration for 6.6 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR66 0
+
+/* Enumeration for 8.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR885 1
+
+/* Enumeration for 12.65 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1265 2
+
+/* Enumeration for 14.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1425 3
+
+/* Enumeration for 15.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1585 4
+
+/* Enumeration for 18.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1825 5
+
+/* Enumeration for 19.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1985 6
+
+/* Enumeration for 23.05 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2305 7
+
+/* Enumeration for 23.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2385 8
+
+struct asm_amrwb_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u16 enc_mode;
+/* AMR-WB encoding rate.
+ * Suupported values:
+ * Use the ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+ u16 dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_V13K_FS 0x00010BED
+
+/* Enumeration for 14.4 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 0
+
+/* Enumeration for 12.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220 1
+
+/* Enumeration for 11.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120 2
+
+/* Enumeration for 9.0 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90 3
+
+/* Enumeration for 7.2 kbps V13K eEncoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720 4
+
+/* Enumeration for 1/8 vocoder rate.*/
+#define ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE 1
+
+/* Enumeration for 1/4 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE 2
+
+/* Enumeration for 1/2 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_HALF_RATE 3
+
+/* Enumeration for full vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_FULL_RATE 4
+
+struct asm_v13k_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u16 max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 reduced_rate_cmd;
+/* Reduced rate command, used to change
+ * the average bitrate of the V13K
+ * vocoder.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 (Default)
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720
+ */
+
+ u16 rate_mod_cmd;
+/* Rate modulation command. Default = 0.
+ *- If bit 0=1, rate control is enabled.
+ *- If bit 1=1, the maximum number of consecutive full rate
+ * frames is limited with numbers supplied in
+ * bits 2 to 10.
+ *- If bit 1=0, the minimum number of non-full rate frames
+ * in between two full rate frames is forced to
+ * the number supplied in bits 2 to 10. In both cases, if necessary,
+ * half rate is used to substitute full rate. - Bits 15 to 10 are
+ * reserved and must all be set to zero.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_EVRC_FS 0x00010BEE
+
+/* EVRC encoder configuration structure used in the
+ * #ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 command.
+ */
+struct asm_evrc_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u16 max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 rate_mod_cmd;
+/* Rate modulation command. Default: 0.
+ * - If bit 0=1, rate control is enabled.
+ * - If bit 1=1, the maximum number of consecutive full rate frames
+ * is limited with numbers supplied in bits 2 to 10.
+ *
+ * - If bit 1=0, the minimum number of non-full rate frames in
+ * between two full rate frames is forced to the number supplied in
+ * bits 2 to 10. In both cases, if necessary, half rate is used to
+ * substitute full rate.
+ *
+ * - Bits 15 to 10 are reserved and must all be set to zero.
+ */
+
+ u16 reserved;
+ /* Reserved. Clients must set this field to zero. */
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V10PRO_V2 0x00010DA7
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u16 fmtag;
+/* WMA format type.
+ * Supported values:
+ * - 0x162 -- WMA 9 Pro
+ * - 0x163 -- WMA 9 Pro Lossless
+ * - 0x166 -- WMA 10 Pro
+ * - 0x167 -- WMA 10 Pro Lossless
+ */
+
+ u16 num_channels;
+/* Number of channels encoded in the input stream.
+ * Supported values: 1 to 8
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 11025, 16000, 22050, 32000, 44100, 48000,
+ * 88200, 96000
+ */
+
+ u32 avg_bytes_per_sec;
+/* Bitrate expressed as the average bytes per second.
+ * Supported values: 2000 to 96000
+ */
+
+ u16 blk_align;
+/* Size of the bitstream packet size in bytes. WMA Pro files
+ * have a payload of one block per bitstream packet.
+ * Supported values: @le 13376
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample in the encoded WMA stream.
+ * Supported values: 16, 24
+ */
+
+ u32 channel_mask;
+/* Bit-packed double word (32-bits) that indicates the
+ * recommended speaker positions for each source channel.
+ */
+
+ u16 enc_options;
+/* Bit-packed word with values that indicate whether certain
+ * features of the bitstream are used.
+ * Supported values: - 0x0001 -- ENCOPT3_PURE_LOSSLESS - 0x0006 --
+ * ENCOPT3_FRM_SIZE_MOD - 0x0038 -- ENCOPT3_SUBFRM_DIV - 0x0040 --
+ * ENCOPT3_WRITE_FRAMESIZE_IN_HDR - 0x0080 --
+ * ENCOPT3_GENERATE_DRC_PARAMS - 0x0100 -- ENCOPT3_RTMBITS
+ */
+
+
+ u16 usAdvancedEncodeOpt;
+ /* Advanced encoding option. */
+
+ u32 advanced_enc_options2;
+ /* Advanced encoding option 2. */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V9_V2 0x00010DA8
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u16 fmtag;
+/* WMA format tag.
+ * Supported values: 0x161 (WMA 9 standard)
+ */
+
+ u16 num_channels;
+/* Number of channels in the stream.
+ * Supported values: 1, 2
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 48000
+ */
+
+ u32 avg_bytes_per_sec;
+ /* Bitrate expressed as the average bytes per second. */
+
+ u16 blk_align;
+/* Block align. All WMA files with a maximum packet size of
+ * 13376 are supported.
+ */
+
+
+ u16 bits_per_sample;
+/* Number of bits per sample in the output.
+ * Supported values: 16
+ */
+
+ u32 channel_mask;
+/* Channel mask.
+ * Supported values:
+ * - 3 -- Stereo (front left/front right)
+ * - 4 -- Mono (center)
+ */
+
+ u16 enc_options;
+ /* Options used during encoding. */
+
+ u16 reserved;
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V8 0x00010D91
+
+struct asm_wmastdv8_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u32 bit_rate;
+ /* Encoding rate in bits per second. */
+
+ u32 sample_rate;
+/* Number of samples per second.
+ *
+ * Supported values:
+ * - 0 -- Native mode
+ * - Other Supported values are 22050, 32000, 44100, and 48000.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+ u16 channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+ u16 reserved;
+ /* Reserved. Clients must set this field to zero.*/
+ } __packed;
+
+#define ASM_MEDIA_FMT_AMR_WB_PLUS_V2 0x00010DA9
+
+struct asm_amrwbplus_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u32 amr_frame_fmt;
+/* AMR frame format.
+ * Supported values:
+ * - 6 -- Transport Interface Format (TIF)
+ * - Any other value -- File storage format (FSF)
+ *
+ * TIF stream contains 2-byte header for each frame within the
+ * superframe. FSF stream contains one 2-byte header per superframe.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AC3 0x00010DEE
+#define ASM_MEDIA_FMT_EAC3 0x00010DEF
+#define ASM_MEDIA_FMT_DTS 0x00010D88
+#define ASM_MEDIA_FMT_MP2 0x00010DE9
+#define ASM_MEDIA_FMT_FLAC 0x00010C16
+#define ASM_MEDIA_FMT_ALAC 0x00012F31
+#define ASM_MEDIA_FMT_VORBIS 0x00010C15
+#define ASM_MEDIA_FMT_APE 0x00012F32
+#define ASM_MEDIA_FMT_DSD 0x00012F3E
+#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 doesn't reset unless the system is reset.
+ */
+
+
+u32 timestamp_msw;
+/* Upper 32 bits of the 64-bit session time in microseconds of
+ * the first sample in the buffer.
+ */
+
+
+u32 flags;
+/* Bitfield of flags. Bit 30 indicates whether frame metadata is
+ * present. If frame metadata is present, num_frames consecutive
+ * instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start
+ * at the buffer address.
+ * Supported values for bit 31:
+ * - 1 -- Timestamp is valid.
+ * - 0 -- Timestamp is invalid.
+ * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and
+ * #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit.
+ *
+ * Supported values for bit 30:
+ * - 1 -- Frame metadata is present.
+ * - 0 -- Frame metadata is absent.
+ * - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and
+ * #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit.
+ *
+ * All other bits are reserved; the aDSP sets them to 0.
+ */
+
+u32 num_frames;
+/* Number of encoded frames in the buffer. */
+
+u32 seq_id;
+/* Optional buffer sequence ID. */
+} __packed;
+
+struct asm_data_read_buf_metadata_v2 {
+ u32 offset;
+/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to
+ * the frame associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32 frm_size;
+/* Size of the encoded frame in bytes.
+ * Supported values: > 0
+ */
+
+u32 num_encoded_pcm_samples;
+/* Number of encoded PCM samples (per channel) in the frame
+ * associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32 timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ * If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1
+ * then the 64 bit timestamp is absolute capture time otherwise it
+ * is relative session time. The absolute timestamp doesn't reset
+ * unless the system is reset.
+ */
+
+
+u32 timestamp_msw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ */
+
+u32 flags;
+/* Frame flags.
+ * Supported values for bit 31:
+ * - 1 -- Time stamp is valid
+ * - 0 -- Time stamp is not valid
+ * - All other bits are reserved; the aDSP sets them to 0.
+ */
+} __packed;
+
+/* Notifies the client of a change in the data sampling rate or
+ * Channel mode. This event is raised by the decoder service. The
+ * event is enabled through the mode flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in the output sampling frequency or the number/positioning of
+ * output channels, or if it is the first frame decoded.The new
+ * sampling frequency or the new channel configuration is
+ * communicated back to the client asynchronously.
+ */
+
+#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
+
+/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event.
+ * This event is raised when the following conditions are both true:
+ * - The event is enabled through the mode_flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in either the output sampling frequency or the number/positioning
+ * of output channels, or if it is the first frame decoded.
+ * This event is not raised (even if enabled) if the decoder is
+ * MIDI, because
+ */
+
+
+struct asm_data_event_sr_cm_change_notify {
+ u32 sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * bitstream.
+ * Supported values: 2000 to 48000
+ */
+
+ u16 num_channels;
+/* New number of channels after detecting a change in the
+ * bitstream.
+ * Supported values: 1 to 8
+ */
+
+
+ u16 reserved;
+ /* Reserved for future use. This field must be set to 0.*/
+
+ u8 channel_mapping[8];
+
+} __packed;
+
+/* Notifies the client of a data sampling rate or channel mode
+ * change. This event is raised by the encoder service.
+ * This event is raised when :
+ * - Native mode encoding was requested in the encoder
+ * configuration (i.e., the channel number was 0), the sample rate
+ * was 0, or both were 0.
+ *
+ * - The input data frame at the encoder is the first one, or the
+ * sampling rate/channel mode is different from the previous input
+ * data frame.
+ *
+ */
+#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE
+
+struct asm_data_event_enc_sr_cm_change_notify {
+ u32 sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * input data.
+ * Supported values: 2000 to 48000
+ */
+
+
+ u16 num_channels;
+/* New number of channels after detecting a change in the input
+ * data. Supported values: 1 to 8
+ */
+
+
+ u16 bits_per_sample;
+/* New bits per sample after detecting a change in the input
+ * data.
+ * Supported values: 16, 24
+ */
+
+
+ u8 channel_mapping[8];
+
+} __packed;
+#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87
+
+
+/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command,
+ * which is used to indicate the IEC 60958 frame rate of a given
+ * packetized audio stream.
+ */
+
+struct asm_data_cmd_iec_60958_frame_rate {
+ u32 frame_rate;
+/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream.
+ * Supported values: Any valid frame rate
+ */
+} __packed;
+
+/* adsp_asm_data_commands.h*/
+/* Definition of the stream ID bitmask.*/
+#define ASM_BIT_MASK_STREAM_ID (0x000000FFUL)
+
+/* Definition of the stream ID shift value.*/
+#define ASM_SHIFT_STREAM_ID 0
+
+/* Definition of the session ID bitmask.*/
+#define ASM_BIT_MASK_SESSION_ID (0x0000FF00UL)
+
+/* Definition of the session ID shift value.*/
+#define ASM_SHIFT_SESSION_ID 8
+
+/* Definition of the service ID bitmask.*/
+#define ASM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
+
+/* Definition of the service ID shift value.*/
+#define ASM_SHIFT_SERVICE_ID 16
+
+/* Definition of the domain ID bitmask.*/
+#define ASM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
+
+/* Definition of the domain ID shift value.*/
+#define ASM_SHIFT_DOMAIN_ID 24
+
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
+
+/* adsp_asm_service_commands.h */
+
+#define ASM_MAX_SESSION_ID (15)
+
+/* Maximum number of sessions.*/
+#define ASM_MAX_NUM_SESSIONS ASM_MAX_SESSION_ID
+
+/* Maximum number of streams per session.*/
+#define ASM_MAX_STREAMS_PER_SESSION (8)
+#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE 0
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY 3
+
+#define ASM_BIT_MASK_RUN_STARTIME (0x00000003UL)
+
+/* Bit shift value used to specify the start time for the
+ * ASM_SESSION_CMD_RUN_V2 command.
+ */
+#define ASM_SHIFT_RUN_STARTIME 0
+struct asm_session_cmd_run_v2 {
+ struct apr_hdr hdr;
+ u32 flags;
+/* Specifies whether to run immediately or at a specific
+ * rendering time or with a specified delay. Run with delay is
+ * useful for delaying in case of ASM loopback opened through
+ * ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME
+ * and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag.
+ *
+ *
+ *Bits 0 and 1 can take one of four possible values:
+ *
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY
+ *
+ *All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 time_lsw;
+/* Lower 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+ u32 time_msw;
+/* Upper 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D
+#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
+
+struct asm_session_cmd_rgstr_rx_underflow {
+ struct apr_hdr hdr;
+ u16 enable_flag;
+/* Specifies whether a client is to receive events when an Rx
+ * session underflows.
+ * Supported values:
+ * - 0 -- Do not send underflow events
+ * - 1 -- Send underflow events
+ */
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6
+
+struct asm_session_cmd_regx_overflow {
+ struct apr_hdr hdr;
+ u16 enable_flag;
+/* Specifies whether a client is to receive events when a Tx
+ * session overflows.
+ * Supported values:
+ * - 0 -- Do not send overflow events
+ * - 1 -- Send overflow events
+ */
+
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17
+#define ASM_SESSION_EVENTX_OVERFLOW 0x00010C18
+#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E
+
+struct asm_session_cmdrsp_get_sessiontime_v3 {
+ u32 status;
+ /* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+
+ u32 sessiontime_lsw;
+ /* Lower 32 bits of the current session time in microseconds.*/
+
+ u32 sessiontime_msw;
+ /* Upper 32 bits of the current session time in microseconds.*/
+
+ u32 absolutetime_lsw;
+/* Lower 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered * to hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+
+ u32 absolutetime_msw;
+/* Upper 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered to * hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2 0x00010D9F
+
+struct asm_session_cmd_adjust_session_clock_v2 {
+ struct apr_hdr hdr;
+u32 adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies the
+ * adjustment time in microseconds to the session clock.
+ *
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+
+ u32 adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the adjustment time in microseconds to the session clock.
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 0x00010DA0
+
+struct asm_session_cmdrsp_adjust_session_clock_v2 {
+ u32 status;
+/* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ * An error means the session clock is not adjusted. In this case,
+ * the next two fields are irrelevant.
+ */
+
+
+ u32 actual_adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+ u32 actual_adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+ u32 cmd_latency_lsw;
+/* Lower 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+
+ u32 cmd_latency_msw;
+/* Upper 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_GET_PATH_DELAY_V2 0x00010DAF
+#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0
+
+struct asm_session_cmdrsp_get_path_delay_v2 {
+ u32 status;
+/* Status message (error code). Whether this get delay operation
+ * is successful or not. Delay value is valid only if status is
+ * success.
+ * Supported values: Refer to @xhyperref{Q5,[Q5]}
+ */
+
+ u32 audio_delay_lsw;
+ /* Upper 32 bits of the aDSP delay in microseconds. */
+
+ u32 audio_delay_msw;
+ /* Lower 32 bits of the aDSP delay in microseconds. */
+
+} __packed;
+
+/* adsp_asm_session_command.h*/
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
+
+#define ASM_LOW_LATENCY_STREAM_SESSION 0x10000000
+
+#define ASM_ULTRA_LOW_LATENCY_STREAM_SESSION 0x20000000
+
+#define ASM_ULL_POST_PROCESSING_STREAM_SESSION 0x40000000
+
+#define ASM_LEGACY_STREAM_SESSION 0
+
+
+struct asm_stream_cmd_open_write_v3 {
+ struct apr_hdr hdr;
+ uint32_t mode_flags;
+/* Mode flags that configure the stream to notify the client
+ * whenever it detects an SR/CM change at the input to its POPP.
+ * Supported values for bits 0 to 1:
+ * - Reserved; clients must set them to zero.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled.
+ * - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit.
+ *
+ * Supported values for bit 31:
+ * - 0 -- Stream to be opened in on-Gapless mode.
+ * - 1 -- Stream to be opened in Gapless mode. In Gapless mode,
+ * successive streams must be opened with same session ID but
+ * different stream IDs.
+ *
+ * - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and
+ * #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit.
+ *
+ *
+ * @note1hang MIDI and DTMF streams cannot be opened in Gapless mode.
+ */
+
+ uint16_t sink_endpointype;
+/*< Sink point type.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - Other values are reserved.
+ *
+ * The device matrix is the gateway to the hardware ports.
+ */
+
+ uint16_t bits_per_sample;
+/*< Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ uint32_t postprocopo_id;
+/*< Specifies the topology (order of processing) of
+ * postprocessing algorithms. <i>None</i> means no postprocessing.
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+ uint32_t dec_fmt_id;
+/*< Configuration ID of the decoder media format.
+ *
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_FR_FS
+ * - #ASM_MEDIA_FMT_VORBIS
+ * - #ASM_MEDIA_FMT_FLAC
+ * - #ASM_MEDIA_FMT_ALAC
+ * - #ASM_MEDIA_FMT_APE
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_PULL_MODE_WRITE 0x00010DD9
+
+/* Bitmask for the stream_perf_mode subfield. */
+#define ASM_BIT_MASK_STREAM_PERF_FLAG_PULL_MODE_WRITE 0xE0000000UL
+
+/* Bitmask for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE 29
+
+#define ASM_STREAM_CMD_OPEN_PUSH_MODE_READ 0x00010DDA
+
+#define ASM_BIT_MASK_STREAM_PERF_FLAG_PUSH_MODE_READ 0xE0000000UL
+
+#define ASM_SHIFT_STREAM_PERF_FLAG_PUSH_MODE_READ 29
+
+#define ASM_DATA_EVENT_WATERMARK 0x00010DDB
+
+struct asm_shared_position_buffer {
+ volatile uint32_t frame_counter;
+/* Counter used to handle interprocessor synchronization issues.
+ * When frame_counter is 0: read_index, wall_clock_us_lsw, and
+ * wall_clock_us_msw are invalid.
+ * Supported values: >= 0.
+ */
+
+ volatile uint32_t index;
+/* Index in bytes from where the aDSP is reading/writing.
+ * Supported values: 0 to circular buffer size - 1
+ */
+
+ volatile uint32_t wall_clock_us_lsw;
+/* Lower 32 bits of the 64-bit wall clock time in microseconds when the
+ * read index was updated.
+ * Supported values: >= 0
+ */
+
+ volatile uint32_t wall_clock_us_msw;
+/* Upper 32 bits of the 64 bit wall clock time in microseconds when the
+ * read index was updated
+ * Supported values: >= 0
+ */
+} __packed;
+
+struct asm_shared_watermark_level {
+ uint32_t watermark_level_bytes;
+} __packed;
+
+struct asm_stream_cmd_open_shared_io {
+ struct apr_hdr hdr;
+ uint32_t mode_flags;
+ uint16_t endpoint_type;
+ uint16_t topo_bits_per_sample;
+ uint32_t topo_id;
+ uint32_t fmt_id;
+ uint32_t shared_pos_buf_phy_addr_lsw;
+ uint32_t shared_pos_buf_phy_addr_msw;
+ uint16_t shared_pos_buf_mem_pool_id;
+ uint16_t shared_pos_buf_num_regions;
+ uint32_t shared_pos_buf_property_flag;
+ uint32_t shared_circ_buf_start_phy_addr_lsw;
+ uint32_t shared_circ_buf_start_phy_addr_msw;
+ uint32_t shared_circ_buf_size;
+ uint16_t shared_circ_buf_mem_pool_id;
+ uint16_t shared_circ_buf_num_regions;
+ uint32_t shared_circ_buf_property_flag;
+ uint32_t num_watermark_levels;
+ struct asm_multi_channel_pcm_fmt_blk_v3 fmt;
+ struct avs_shared_map_region_payload map_region_pos_buf;
+ struct avs_shared_map_region_payload map_region_circ_buf;
+ struct asm_shared_watermark_level watermark[0];
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
+
+/* Definition of the timestamp type flag bitmask */
+#define ASM_BIT_MASKIMESTAMPYPE_FLAG (0x00000020UL)
+
+/* Definition of the timestamp type flag shift value. */
+#define ASM_SHIFTIMESTAMPYPE_FLAG 5
+
+/* Relative timestamp is identified by this value.*/
+#define ASM_RELATIVEIMESTAMP 0
+
+/* Absolute timestamp is identified by this value.*/
+#define ASM_ABSOLUTEIMESTAMP 1
+
+/* Bit value for Low Latency Tx stream subfield */
+#define ASM_LOW_LATENCY_TX_STREAM_SESSION 1
+
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
+
+struct asm_stream_cmd_open_read_v3 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ *
+ * - 0 -- Return data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ *
+ * - 1 -- Return data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG as the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG as the shift value for this bit.
+ *
+ *
+ * Supported values for bit 5:
+ *
+ * - ASM_RELATIVEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will have
+ * - relative time-stamp.
+ * - ASM_ABSOLUTEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will
+ * - have absolute time-stamp.
+ *
+ * - Use #ASM_BIT_MASKIMESTAMPYPE_FLAG as the bitmask and
+ * #ASM_SHIFTIMESTAMPYPE_FLAG as the shift value for this bit.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 src_endpointype;
+/* Specifies the endpoint providing the input samples.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - All other values are reserved; clients must set them to zero.
+ * Otherwise, an error is returned.
+ * The device matrix is the gateway from the tunneled Tx ports.
+ */
+
+ u32 preprocopo_id;
+/* Specifies the topology (order of processing) of preprocessing
+ * algorithms. <i>None</i> means no preprocessing.
+ * Supported values:
+ * - #ASM_STREAM_PREPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_PREPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+ u32 enc_cfg_id;
+/* Media configuration ID for encoded output.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero.*/
+} __packed;
+
+#define ASM_POPP_OUTPUT_SR_NATIVE_RATE 0
+
+/* Enumeration for the maximum sampling rate at the POPP output.*/
+#define ASM_POPP_OUTPUT_SR_MAX_RATE 48000
+
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+
+struct asm_stream_cmd_open_readwrite_v2 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled. Use
+ * #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or
+ * getting this flag.
+ *
+ * Supported values for bit 4:
+ * - 0 -- Return read data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ * - 1 -- Return read data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 postprocopo_id;
+/* Specifies the topology (order of processing) of postprocessing
+ * algorithms. <i>None</i> means no postprocessing.
+ *
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ */
+
+ u32 dec_fmt_id;
+/* Specifies the media type of the input data. PCM indicates that
+ * no decoding must be performed, e.g., this is an NT encoder
+ * session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+
+ u32 enc_cfg_id;
+/* Specifies the media type for the output of the stream. PCM
+ * indicates that no encoding must be performed, e.g., this is an NT
+ * decoder session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero.*/
+
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E
+struct asm_stream_cmd_open_loopback_v2 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags.
+ * Bit 0-31: reserved; client should set these bits to 0
+ */
+ u16 src_endpointype;
+ /* Endpoint type. 0 = Tx Matrix */
+ u16 sink_endpointype;
+ /* Endpoint type. 0 = Rx Matrix */
+ u32 postprocopo_id;
+/* Postprocessor topology ID. Specifies the topology of
+ * postprocessing algorithms.
+ */
+
+ u16 bits_per_sample;
+/* The number of bits per sample processed by ASM modules
+ * Supported values: 16 and 24 bits per sample
+ */
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero. */
+} __packed;
+
+
+#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA
+
+/* Bitmask for the stream's Performance mode. */
+#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \
+ (0x70000000UL)
+
+/* Bit shift for the stream's Performance mode. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28
+
+/* Bitmask for the decoder converter enable flag. */
+#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL)
+
+/* Shift value for the decoder converter enable flag. */
+#define ASM_SHIFT_DECODER_CONVERTER_FLAG 3
+
+/* Converter mode is None (Default). */
+#define ASM_CONVERTER_MODE_NONE 0
+
+/* Converter mode is DDP-to-DD. */
+#define ASM_DDP_DD_CONVERTER_MODE 1
+
+/* Identifies a special converter mode where source and sink formats
+ * are the same but postprocessing must applied. Therefore, Decode
+ * @rarrow Re-encode is necessary.
+ */
+#define ASM_POST_PROCESS_CONVERTER_MODE 2
+
+
+struct asm_stream_cmd_open_transcode_loopback_t {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode Flags specifies the performance mode in which this stream
+ * is to be opened.
+ * Supported values{for bits 30 to 28}(stream_perf_mode flag)
+ *
+ * #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward
+ * compatibility to the original behavior
+ * of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK
+ *
+ * #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using
+ * shortened buffers in low latency POPP
+ * - Recommendation: Do not enable high latency algorithms. They might
+ * negate the benefits of opening a low latency stream, and they
+ * might also suffer quality degradation from unexpected jitter.
+ * - This Low Latency mode is supported only for PCM In and PCM Out
+ * loopbacks. An error is returned if Low Latency mode is opened for
+ * other transcode loopback modes.
+ * - To configure this subfield, use
+ * ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and
+ * ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK.
+ *
+ * Supported values{for bits 6 to 3} (decoder-converter compatibility)
+ * #ASM_CONVERTER_MODE_NONE (0x0) -- Default
+ * #ASM_DDP_DD_CONVERTER_MODE (0x1)
+ * #ASM_POST_PROCESS_CONVERTER_MODE (0x2)
+ * 0x3-0xF -- Reserved for future use
+ * - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and
+ * ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit
+ * All other bits are reserved; clients must set them to 0.
+ */
+
+ u32 src_format_id;
+/* Specifies the media format of the input audio stream.
+ *
+ * Supported values
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
+ * - #ASM_MEDIA_FMT_DTS
+ * - #ASM_MEDIA_FMT_EAC3_DEC
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_AC3_DEC
+ * - #ASM_MEDIA_FMT_AC3
+ */
+ u32 sink_format_id;
+/* Specifies the media format of the output stream.
+ *
+ * Supported values
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
+ * - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode)
+ */
+
+ u32 audproc_topo_id;
+/* Postprocessing topology ID, which specifies the topology (order of
+ * processing) of postprocessing algorithms.
+ *
+ * Supported values
+ * - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER
+ * - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROC_TOPO_ID_NONE
+ * Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES.
+ * This field is ignored for the Converter mode, in which no
+ * postprocessing is performed.
+ */
+
+ u16 src_endpoint_type;
+/* Specifies the source endpoint that provides the input samples.
+ *
+ * Supported values
+ * - 0 -- Tx device matrix or stream router (gateway to the hardware
+ * ports)
+ * - All other values are reserved
+ * Clients must set this field to 0. Otherwise, an error is returned.
+ */
+
+ u16 sink_endpoint_type;
+/* Specifies the sink endpoint type.
+ *
+ * Supported values
+ * - 0 -- Rx device matrix or stream router (gateway to the hardware
+ * ports)
+ * - All other values are reserved
+ * Clients must set this field to 0. Otherwise, an error is returned.
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample processed by the ASM modules.
+ * Supported values 16, 24
+ */
+
+ u16 reserved;
+/* This field must be set to 0.
+ */
+} __packed;
+
+
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+
+
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1
+
+struct asm_stream_cmd_set_pp_params_v2 {
+ u32 data_payload_addr_lsw;
+/* LSW of parameter data payload address. Supported values: any. */
+ u32 data_payload_addr_msw;
+/* MSW of Parameter data payload address. Supported values: any.
+ * - Must be set to zero for in-band data.
+ * - In the case of 32 bit Shared memory address, msw field must be
+ * - set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw
+ *
+ * - must be set to zero.
+ */
+ u32 mem_map_handle;
+/* Supported Values: Any.
+ * memory map handle returned by DSP through
+ * ASM_CMD_SHARED_MEM_MAP_REGIONS
+ * command.
+ * if mmhandle is NULL, the ParamData payloads are within the
+ * message payload (in-band).
+ * If mmhandle is non-NULL, the ParamData payloads begin at the
+ * address specified in the address msw and lsw (out-of-band).
+ */
+
+ u32 data_payload_size;
+/* Size in bytes of the variable payload accompanying the
+ * message, or in shared memory. This field is used for parsing the
+ * parameter payload.
+ */
+} __packed;
+
+
+struct asm_stream_param_data_v2 {
+ u32 module_id;
+ /* Unique module ID. */
+
+ u32 param_id;
+ /* Unique parameter ID. */
+
+ u16 param_size;
+/* Data size of the param_id/module_id combination. This is
+ * a multiple of 4 bytes.
+ */
+
+ u16 reserved;
+/* Reserved for future enhancements. This field must be set to
+ * zero.
+ */
+
+} __packed;
+
+#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2
+
+struct asm_stream_cmd_get_pp_params_v2 {
+ u32 data_payload_addr_lsw;
+ /* LSW of the parameter data payload address. */
+ u32 data_payload_addr_msw;
+/* MSW of the parameter data payload address.
+ * - Size of the shared memory, if specified, shall be large enough
+ * to contain the whole ParamData payload, including Module ID,
+ * Param ID, Param Size, and Param Values
+ * - Must be set to zero for in-band data
+ * - In the case of 32 bit Shared memory address, msw field must be
+ * set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw must be set to zero.
+ */
+
+ u32 mem_map_handle;
+/* Supported Values: Any.
+ * memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS
+ * command.
+ * if mmhandle is NULL, the ParamData payloads in the ACK are within the
+ * message payload (in-band).
+ * If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the
+ * address specified in the address msw and lsw.
+ * (out-of-band).
+ */
+
+ u32 module_id;
+/* Unique module ID. */
+
+ u32 param_id;
+/* Unique parameter ID. */
+
+ u16 param_max_size;
+/* Maximum data size of the module_id/param_id combination. This
+ * is a multiple of 4 bytes.
+ */
+
+
+ u16 reserved;
+/* Reserved for backward compatibility. Clients must set this
+ * field to zero.
+ */
+} __packed;
+
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+
+#define ASM_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
+
+struct asm_stream_cmdrsp_get_pp_params_v2 {
+ u32 status;
+} __packed;
+
+#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73
+
+/* Enumeration for both vocals in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_NO_VOCAL (0)
+
+/* Enumeration for only the left vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_LEFT_VOCAL (1)
+
+/* Enumeration for only the right vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2)
+
+/* Enumeration for both vocal channels in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_BOTH_VOCAL (3)
+#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74
+/* Enumeration for the Custom Analog mode.*/
+#define AC3_DRC_MODE_CUSTOM_ANALOG (0)
+
+/* Enumeration for the Custom Digital mode.*/
+#define AC3_DRC_MODE_CUSTOM_DIGITAL (1)
+/* Enumeration for the Line Out mode (light compression).*/
+#define AC3_DRC_MODE_LINE_OUT (2)
+
+/* Enumeration for the RF remodulation mode (heavy compression).*/
+#define AC3_DRC_MODE_RF_REMOD (3)
+#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75
+
+/* Enumeration for playing dual mono in stereo mode.*/
+#define AC3_DUAL_MONO_MODE_STEREO (0)
+
+/* Enumeration for playing left mono.*/
+#define AC3_DUAL_MONO_MODE_LEFT_MONO (1)
+
+/* Enumeration for playing right mono.*/
+#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2)
+
+/* Enumeration for mixing both dual mono channels and playing them.*/
+#define AC3_DUAL_MONO_MODE_MIXED_MONO (3)
+#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76
+
+/* Enumeration for using the Downmix mode indicated in the bitstream. */
+
+#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0)
+
+/* Enumeration for Surround Compatible mode (preserves the
+ * surround information).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1)
+/* Enumeration for Mono Compatible mode (if the output is to be
+ * further downmixed to mono).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2)
+
+/* ID of the AC3 PCM scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78
+
+/* ID of the AC3 DRC boost scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79
+
+/* ID of the AC3 DRC cut scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A
+
+/* Structure for AC3 Generic Parameter. */
+
+/* Payload of the AC3 parameters in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_ac3_generic_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u32 generic_parameter;
+/* AC3 generic parameter. Select from one of the following
+ * possible values.
+ *
+ * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are:
+ * - AC3_KARAOKE_MODE_NO_VOCAL
+ * - AC3_KARAOKE_MODE_LEFT_VOCAL
+ * - AC3_KARAOKE_MODE_RIGHT_VOCAL
+ * - AC3_KARAOKE_MODE_BOTH_VOCAL
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are:
+ * - AC3_DRC_MODE_CUSTOM_ANALOG
+ * - AC3_DRC_MODE_CUSTOM_DIGITAL
+ * - AC3_DRC_MODE_LINE_OUT
+ * - AC3_DRC_MODE_RF_REMOD
+ *
+ * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are:
+ * - AC3_DUAL_MONO_MODE_STEREO
+ * - AC3_DUAL_MONO_MODE_LEFT_MONO
+ * - AC3_DUAL_MONO_MODE_RIGHT_MONO
+ * - AC3_DUAL_MONO_MODE_MIXED_MONO
+ *
+ * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are:
+ * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT
+ * - AC3_STEREO_DOWNMIX_MODE_LT_RT
+ * - AC3_STEREO_DOWNMIX_MODE_LO_RO
+ *
+ * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ */
+} __packed;
+
+/* Enumeration for Raw mode (no downmixing), which specifies
+ * that all channels in the bitstream are to be played out as is
+ * without any downmixing. (Default)
+ */
+
+#define WMAPRO_CHANNEL_MASK_RAW (-1)
+
+/* Enumeration for setting the channel mask to 0. The 7.1 mode
+ * (Home Theater) is assigned.
+ */
+
+
+#define WMAPRO_CHANNEL_MASK_ZERO 0x0000
+
+/* Speaker layout mask for one channel (Home Theater, mono).
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_1_C 0x0004
+
+/* Speaker layout mask for two channels (Home Theater, stereo).
+ * - Speaker front left
+ * - Speaker front right
+ */
+#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003
+
+/* Speaker layout mask for three channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007
+
+/* Speaker layout mask for two channels (stereo).
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030
+
+/* Speaker layout mask for four channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033
+
+/* Speaker layout mask for four channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107
+/* Speaker layout mask for five channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037
+
+/* Speaker layout mask for five channels (5 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607
+/* Speaker layout mask for six channels (5.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F
+/* Speaker layout mask for six channels (5.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F
+/* Speaker layout mask for six channels (5.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137
+/* Speaker layout mask for six channels (5.1 mode, Home Theater,
+ * no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707
+
+/* Speaker layout mask for seven channels (6.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+ * Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F
+
+/* Speaker layout mask for seven channels (6.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker front left of center
+ * - Speaker front right of center
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+ * Theater, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker front left of center
+ * - Speaker front right of center
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637
+
+/* Speaker layout mask for eight channels (7.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ */
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \
+ 0x00FF
+
+/* Speaker layout mask for eight channels (7.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ *
+ */
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \
+ 0x063F
+
+#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82
+
+/* Maximum number of decoder output channels. */
+#define MAX_CHAN_MAP_CHANNELS 16
+
+/* Structure for decoder output channel mapping. */
+
+/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_dec_out_chan_map_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ u32 num_channels;
+/* Number of decoder output channels.
+ * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS
+ *
+ * A value of 0 indicates native channel mapping, which is valid
+ * only for NT mode. This means the output of the decoder is to be
+ * preserved as is.
+ */
+ u8 channel_mapping[MAX_CHAN_MAP_CHANNELS];
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84
+
+/* Bitmask for the IEC 61937 enable flag.*/
+#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL)
+
+/* Shift value for the IEC 61937 enable flag.*/
+#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0
+
+/* Bitmask for the IEC 60958 enable flag.*/
+#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL)
+
+/* Shift value for the IEC 60958 enable flag.*/
+#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1
+
+/* Payload format for open write compressed command */
+
+/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
+ * command, which opens a stream for a given session ID and stream ID
+ * to be rendered in the compressed format.
+ */
+
+struct asm_stream_cmd_open_write_compressed {
+ struct apr_hdr hdr;
+ u32 flags;
+/* Mode flags that configure the stream for a specific format.
+ * Supported values:
+ * - Bit 0 -- IEC 61937 compatibility
+ * - 0 -- Stream is not in IEC 61937 format
+ * - 1 -- Stream is in IEC 61937 format
+ * - Bit 1 -- IEC 60958 compatibility
+ * - 0 -- Stream is not in IEC 60958 format
+ * - 1 -- Stream is in IEC 60958 format
+ * - Bits 2 to 31 -- 0 (Reserved)
+ *
+ * For the same stream, bit 0 cannot be set to 0 and bit 1 cannot
+ * be set to 1. A compressed stream connot have IEC 60958
+ * packetization applied without IEC 61937 packetization.
+ * @note1hang Currently, IEC 60958 packetized input streams are not
+ * supported.
+ */
+
+
+ u32 fmt_id;
+/* Specifies the media type of the HDMI stream to be opened.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AC3
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_DTS
+ * - #ASM_MEDIA_FMT_ATRAC
+ * - #ASM_MEDIA_FMT_MAT
+ *
+ * @note1hang This field must be set to a valid media type even if
+ * IEC 61937 packetization is not performed by the aDSP.
+ */
+
+} __packed;
+
+
+/* Indicates the number of samples per channel to be removed from the
+ * beginning of the stream.
+ */
+#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67
+
+/* Indicates the number of samples per channel to be removed from
+ * the end of the stream.
+ */
+#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
+
+struct asm_data_cmd_remove_silence {
+ struct apr_hdr hdr;
+ u32 num_samples_to_remove;
+ /* < Number of samples per channel to be removed.
+ * @values 0 to (2@sscr{32}-1)
+ */
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95
+
+struct asm_stream_cmd_open_read_compressed {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ * - 0 -- Return data buffer contains all encoded frames only; it does
+ * not contain frame metadata.
+ * - 1 -- Return data buffer contains an array of metadata and encoded
+ * frames.
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit.
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 frames_per_buf;
+/* Indicates the number of frames that need to be returned per
+ * read buffer
+ * Supported values: should be greater than 0
+ */
+
+} __packed;
+
+/* adsp_asm_stream_commands.h*/
+
+
+/* adsp_asm_api.h (no changes)*/
+#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \
+ 0x00010BE4
+#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \
+ 0x00010D83
+#define ASM_STREAM_POSTPROCOPO_ID_NONE \
+ 0x00010C68
+#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \
+ 0x00010D8B
+#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \
+ ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+#define ASM_STREAM_PREPROCOPO_ID_NONE \
+ ASM_STREAM_POSTPROCOPO_ID_NONE
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \
+ 0x00010312
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \
+ 0x00010313
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \
+ 0x00010314
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\
+ 0x00010704
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\
+ 0x0001070D
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\
+ 0x0001070E
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\
+ 0x0001070F
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRC_V3 \
+ 0x11000000
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \
+ 0x0001031B
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP 0x00010315
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316
+#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO 0x00010715
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP 0x00010BE3
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317
+#define AUDPROC_MODULE_ID_AIG 0x00010716
+#define AUDPROC_PARAM_ID_AIG_ENABLE 0x00010717
+#define AUDPROC_PARAM_ID_AIG_CONFIG 0x00010718
+
+struct Audio_AigParam {
+ uint16_t mode;
+/*< Mode word for enabling AIG/SIG mode .
+ * Byte offset: 0
+ */
+ int16_t staticGainL16Q12;
+/*< Static input gain when aigMode is set to 1.
+ * Byte offset: 2
+ */
+ int16_t initialGainDBL16Q7;
+/*<Initial value that the adaptive gain update starts from dB
+ * Q7 Byte offset: 4
+ */
+ int16_t idealRMSDBL16Q7;
+/*<Average RMS level that AIG attempts to achieve Q8.7
+ * Byte offset: 6
+ */
+ int32_t noiseGateL32;
+/*Threshold below which signal is considered as noise and AIG
+ * Byte offset: 8
+ */
+ int32_t minGainL32Q15;
+/*Minimum gain that can be provided by AIG Q16.15
+ * Byte offset: 12
+ */
+ int32_t maxGainL32Q15;
+/*Maximum gain that can be provided by AIG Q16.15
+ * Byte offset: 16
+ */
+ uint32_t gainAtRtUL32Q31;
+/*Attack/release time for AIG update Q1.31
+ * Byte offset: 20
+ */
+ uint32_t longGainAtRtUL32Q31;
+/*Long attack/release time while updating gain for
+ * noise/silence Q1.31 Byte offset: 24
+ */
+
+ uint32_t rmsTavUL32Q32;
+/* RMS smoothing time constant used for long-term RMS estimate
+ * Q0.32 Byte offset: 28
+ */
+
+ uint32_t gainUpdateStartTimMsUL32Q0;
+/* The waiting time before which AIG starts to apply adaptive
+ * gain update Q32.0 Byte offset: 32
+ */
+
+} __packed;
+
+
+#define ADM_MODULE_ID_EANS 0x00010C4A
+#define ADM_PARAM_ID_EANS_ENABLE 0x00010C4B
+#define ADM_PARAM_ID_EANS_PARAMS 0x00010C4C
+
+struct adm_eans_enable {
+
+ uint32_t enable_flag;
+/*< Specifies whether EANS is disabled (0) or enabled
+ * (nonzero).
+ * This is supported only for sampling rates of 8, 12, 16, 24, 32,
+ * and 48 kHz. It is not supported for sampling rates of 11.025,
+ * 22.05, or 44.1 kHz.
+ */
+
+} __packed;
+
+
+struct adm_eans_params {
+ int16_t eans_mode;
+/*< Mode word for enabling/disabling submodules.
+ * Byte offset: 0
+ */
+
+ int16_t eans_input_gain;
+/*< Q2.13 input gain to the EANS module.
+ * Byte offset: 2
+ */
+
+ int16_t eans_output_gain;
+/*< Q2.13 output gain to the EANS module.
+ * Byte offset: 4
+ */
+
+ int16_t eansarget_ns;
+/*< Target noise suppression level in dB.
+ * Byte offset: 6
+ */
+
+ int16_t eans_s_alpha;
+/*< Q3.12 over-subtraction factor for stationary noise
+ * suppression.
+ * Byte offset: 8
+ */
+
+ int16_t eans_n_alpha;
+/* < Q3.12 over-subtraction factor for nonstationary noise
+ * suppression.
+ * Byte offset: 10
+ */
+
+ int16_t eans_n_alphamax;
+/*< Q3.12 maximum over-subtraction factor for nonstationary
+ * noise suppression.
+ * Byte offset: 12
+ */
+ int16_t eans_e_alpha;
+/*< Q15 scaling factor for excess noise suppression.
+ * Byte offset: 14
+ */
+
+ int16_t eans_ns_snrmax;
+/*< Upper boundary in dB for SNR estimation.
+ * Byte offset: 16
+ */
+
+ int16_t eans_sns_block;
+/*< Quarter block size for stationary noise suppression.
+ * Byte offset: 18
+ */
+
+ int16_t eans_ns_i;
+/*< Initialization block size for noise suppression.
+ * Byte offset: 20
+ */
+ int16_t eans_np_scale;
+/*< Power scale factor for nonstationary noise update.
+ * Byte offset: 22
+ */
+
+ int16_t eans_n_lambda;
+/*< Smoothing factor for higher level nonstationary noise
+ * update.
+ * Byte offset: 24
+ */
+
+ int16_t eans_n_lambdaf;
+/*< Medium averaging factor for noise update.
+ * Byte offset: 26
+ */
+
+ int16_t eans_gs_bias;
+/*< Bias factor in dB for gain calculation.
+ * Byte offset: 28
+ */
+
+ int16_t eans_gs_max;
+/*< SNR lower boundary in dB for aggressive gain calculation.
+ * Byte offset: 30
+ */
+
+ int16_t eans_s_alpha_hb;
+/*< Q3.12 over-subtraction factor for high-band stationary
+ * noise suppression.
+ * Byte offset: 32
+ */
+
+ int16_t eans_n_alphamax_hb;
+/*< Q3.12 maximum over-subtraction factor for high-band
+ * nonstationary noise suppression.
+ * Byte offset: 34
+ */
+
+ int16_t eans_e_alpha_hb;
+/*< Q15 scaling factor for high-band excess noise suppression.
+ * Byte offset: 36
+ */
+
+ int16_t eans_n_lambda0;
+/*< Smoothing factor for nonstationary noise update during
+ * speech activity.
+ * Byte offset: 38
+ */
+
+ int16_t thresh;
+/*< Threshold for generating a binary VAD decision.
+ * Byte offset: 40
+ */
+
+ int16_t pwr_scale;
+/*< Indirect lower boundary of the noise level estimate.
+ * Byte offset: 42
+ */
+
+ int16_t hangover_max;
+/*< Avoids mid-speech clipping and reliably detects weak speech
+ * bursts at the end of speech activity.
+ * Byte offset: 44
+ */
+
+ int16_t alpha_snr;
+/*< Controls responsiveness of the VAD.
+ * Byte offset: 46
+ */
+
+ int16_t snr_diff_max;
+/*< Maximum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 48
+ */
+
+ int16_t snr_diff_min;
+/*< Minimum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 50
+ */
+
+ int16_t init_length;
+/*< Defines the number of frames for which a noise level
+ * estimate is set to a fixed value.
+ * Byte offset: 52
+ */
+
+ int16_t max_val;
+/*< Defines the upper limit of the noise level.
+ * Byte offset: 54
+ */
+
+ int16_t init_bound;
+/*< Defines the initial bounding value for the noise level
+ * estimate. This is used during the initial segment defined by the
+ * init_length parameter.
+ * Byte offset: 56
+ */
+
+ int16_t reset_bound;
+/*< Reset boundary for noise tracking.
+ * Byte offset: 58
+ */
+
+ int16_t avar_scale;
+/*< Defines the bias factor in noise estimation.
+ * Byte offset: 60
+ */
+
+ int16_t sub_nc;
+/*< Defines the window length for noise estimation.
+ * Byte offset: 62
+ */
+
+ int16_t spow_min;
+/*< Defines the minimum signal power required to update the
+ * boundaries for the noise floor estimate.
+ * Byte offset: 64
+ */
+
+ int16_t eans_gs_fast;
+/*< Fast smoothing factor for postprocessor gain.
+ * Byte offset: 66
+ */
+
+ int16_t eans_gs_med;
+/*< Medium smoothing factor for postprocessor gain.
+ * Byte offset: 68
+ */
+
+ int16_t eans_gs_slow;
+/*< Slow smoothing factor for postprocessor gain.
+ * Byte offset: 70
+ */
+
+ int16_t eans_swb_salpha;
+/*< Q3.12 super wideband aggressiveness factor for stationary
+ * noise suppression.
+ * Byte offset: 72
+ */
+
+ int16_t eans_swb_nalpha;
+/*< Q3.12 super wideband aggressiveness factor for
+ * nonstationary noise suppression.
+ * Byte offset: 74
+ */
+} __packed;
+#define ADM_MODULE_IDX_MIC_GAIN_CTRL 0x00010C35
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Tx mic gain control parameter used by the
+ * #ADM_MODULE_IDX_MIC_GAIN_CTRL module.
+ * @messagepayload
+ * @structure{admx_mic_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_MIC_GAIN 0x00010C36
+
+/* Structure for a Tx mic gain parameter for the mic gain
+ * control module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the
+ * Tx Mic Gain Control module.
+ */
+struct admx_mic_gain {
+ uint16_t tx_mic_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero. */
+} __packed;
+
+struct adm_set_mic_gain_params {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+ struct admx_mic_gain mic_gain_data;
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Rx Codec Gain Control module.
+ *
+ * This module supports the following parameter ID:
+ * - #ADM_PARAM_ID_RX_CODEC_GAIN
+ */
+#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL 0x00010C37
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Rx codec gain control parameter used by the
+ * #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module.
+ *
+ * @messagepayload
+ * @structure{adm_rx_codec_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex}
+ */
+#define ADM_PARAM_ID_RX_CODEC_GAIN 0x00010C38
+
+/* Structure for the Rx common codec gain control module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter
+ * in the Rx Codec Gain Control module.
+ */
+
+
+struct adm_rx_codec_gain {
+ uint16_t rx_codec_gain;
+ /* Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /* Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the HPF Tuning Filter module on the Tx path.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ */
+#define ADM_MODULE_ID_HPF_IIRX_FILTER 0x00010C3D
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the Tx HPF IIR filter enable parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG 0x00010C3E
+
+/* ID of the Tx HPF IIR filter pregain parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN 0x00010C3F
+
+/* ID of the Tx HPF IIR filter configuration parameters used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA
+ * RAMS.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS 0x00010C40
+
+/* Structure for enabling a configuration parameter for
+ * the HPF IIR tuning filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * parameter in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_enable_cfg {
+ uint32_t enable_flag;
+/* Specifies whether the HPF tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the pregain parameter for the HPF
+ * IIR tuning filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter
+ * in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_pre_gain {
+ uint16_t pre_gain;
+ /* Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /* Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+ * HPF IIR tuning filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ * parameters in the Tx path HPF Tuning Filter module. \n
+ * \n
+ * This structure is followed by tuning filter coefficients as follows: \n
+ * - Sequence of int32_t FilterCoeffs.
+ * Each band has five coefficients, each in int32_t format in the order of
+ * b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor.
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting.
+ * One uint16_t for each band to indicate application of the filter to
+ * left (0), right (1), or both (2) channels.
+ */
+struct adm_hpfx_iir_filter_cfg_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_module_ids */
+/* ID of the Tx path IIR Tuning Filter module.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ */
+#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41
+
+/* ID of the Rx path IIR Tuning Filter module for the left channel.
+ * The parameter IDs of the IIR tuning filter module
+ * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning
+ * filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter; the pan
+ * parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER 0x00010705
+
+/* ID of the the Rx path IIR Tuning Filter module for the right
+ * channel.
+ * The parameter IDs of the IIR tuning filter module
+ * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx
+ * tuning filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter;
+ * the pan parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER 0x00010706
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_param_ids */
+
+/* ID of the Tx IIR filter enable parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG 0x00010C42
+
+/* ID of the Tx IIR filter pregain parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN 0x00010C43
+
+/* ID of the Tx IIR filter configuration parameters used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS 0x00010C44
+
+/* Structure for enabling the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_enable_cfg {
+ uint32_t enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+
+} __packed;
+
+
+/* Structure for the pregain parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_pre_gain {
+ uint16_t pre_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS
+ * parameter in the Tx Path IIR Tuning Filter module. \n
+ * \n
+ * This structure is followed by the HPF IIR filter coefficients on
+ * the Tx path as follows: \n
+ * - Sequence of int32_t ulFilterCoeffs. Each band has five
+ * coefficients, each in int32_t format in the order of b0, b1, b2,
+ * a1, a2.
+ * - Sequence of int16_t sNumShiftFactor. One int16_t per band. The
+ * numerator shift factor is related to the Q factor of the filter
+ * coefficients.
+ * - Sequence of uint16_t usPanSetting. One uint16_t for each band
+ * to indicate if the filter is applied to left (0), right (1), or
+ * both (2) channels.
+ */
+struct admx_iir_filter_cfg_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the QEnsemble module.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ * - #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ * - #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ */
+#define ADM_MODULE_ID_QENSEMBLE 0x00010C59
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the QEnsemble enable parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_ENABLE 0x00010C60
+
+/* ID of the QEnsemble back gain parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_backgain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN 0x00010C61
+
+/* ID of the QEnsemble new angle parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_set_new_angle}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE 0x00010C62
+
+/* Structure for enabling the configuration parameter for the
+ * QEnsemble module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ * parameter used by the QEnsemble module.
+ */
+struct adm_qensemble_enable {
+ uint32_t enable_flag;
+/*< Specifies whether the QEnsemble module is disabled (0) or enabled
+ * (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the background gain for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ * parameter used by
+ * the QEnsemble module.
+ */
+struct adm_qensemble_param_backgain {
+ int16_t back_gain;
+/*< Linear gain in Q15 format.
+ * Supported values: 0 to 32767
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+/* Structure for setting a new angle for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ * parameter used
+ * by the QEnsemble module.
+ */
+struct adm_qensemble_param_set_new_angle {
+ int16_t new_angle;
+/*< New angle in degrees.
+ * Supported values: 0 to 359
+ */
+
+ int16_t time_ms;
+/*< Transition time in milliseconds to set the new angle.
+ * Supported values: 0 to 32767
+ */
+} __packed;
+
+
+#define ADM_CMD_GET_PP_TOPO_MODULE_LIST 0x00010349
+#define ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST 0x00010350
+#define AUDPROC_PARAM_ID_ENABLE 0x00010904
+ /*
+ * Payload of the ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
+ */
+struct adm_cmd_get_pp_topo_module_list_t {
+ struct apr_hdr hdr;
+ /* Lower 32 bits of the 64-bit parameter data payload address. */
+ uint32_t data_payload_addr_lsw;
+ /*
+ * Upper 32 bits of the 64-bit parameter data payload address.
+ *
+ *
+ * The size of the shared memory, if specified, must be large enough to
+ * contain the entire parameter data payload, including the module ID,
+ * parameter ID, parameter size, and parameter values.
+ */
+ uint32_t data_payload_addr_msw;
+ /*
+ * Unique identifier for an address.
+ *
+ * This memory map handle is returned by the aDSP through the
+ * #ADM_CMD_SHARED_MEM_MAP_REGIONS command.
+ *
+ * @values
+ * - Non-NULL -- On acknowledgment, the parameter data payloads begin at
+ * the address specified (out-of-band)
+ * - NULL -- The acknowledgment's payload contains the parameter data
+ * (in-band) @tablebulletend
+ */
+ uint32_t mem_map_handle;
+ /*
+ * Maximum data size of the list of modules. This
+ * field is a multiple of 4 bytes.
+ */
+ uint16_t param_max_size;
+ /* This field must be set to zero. */
+ uint16_t reserved;
+} __packed;
+
+/*
+ * Payload of the ADM_CMDRSP_GET_PP_TOPO_MODULE_LIST message, which returns
+ * module ids in response to an ADM_CMD_GET_PP_TOPO_MODULE_LIST command.
+ * Immediately following this structure is the acknowledgment <b>module id
+ * data variable payload</b> containing the pre/postprocessing module id
+ * values. For an in-band scenario, the variable payload depends on the size
+ * of the parameter.
+ */
+struct adm_cmd_rsp_get_pp_topo_module_list_t {
+ /* Status message (error code). */
+ uint32_t status;
+} __packed;
+
+struct audproc_topology_module_id_info_t {
+ uint32_t num_modules;
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Volume Control module pre/postprocessing block.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * - #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * - #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * - #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ */
+#define ASM_MODULE_ID_VOL_CTRL 0x00010BFE
+#define ASM_MODULE_ID_VOL_CTRL2 0x00010910
+#define AUDPROC_MODULE_ID_VOL_CTRL ASM_MODULE_ID_VOL_CTRL
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_master_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN 0x00010BFF
+#define AUDPROC_PARAM_ID_VOL_CTRL_MASTER_GAIN ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+
+/* ID of the left/right channel gain parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_lr_chan_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MULTICHANNEL_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN 0x00010C00
+
+/* ID of the mute configuration parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_mute_config}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG 0x00010C01
+
+/* ID of the soft stepping volume parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_soft_step_volume_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET
+ * ERS.tex}
+ */
+#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS 0x00010C29
+#define AUDPROC_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS\
+ ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+
+/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ */
+#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A
+
+/* ID of the multiple-channel volume control parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713
+
+/* ID of the multiple-channel mute configuration parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+
+#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714
+
+/* Structure for the master gain parameter for a volume control
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * parameter used by the Volume Control module.
+ */
+
+
+
+struct asm_volume_ctrl_master_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint16_t master_gain;
+ /* Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /* Clients must set this field to zero. */
+} __packed;
+
+
+struct asm_volume_ctrl_lr_chan_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+
+ uint16_t l_chan_gain;
+ /*< Linear gain in Q13 format for the left channel. */
+
+ uint16_t r_chan_gain;
+ /*< Linear gain in Q13 format for the right channel.*/
+} __packed;
+
+
+/* Structure for the mute configuration parameter for a
+ * volume control module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * parameter used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_mute_config {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t mute_flag;
+/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/
+
+} __packed;
+
+/*
+ * Supported parameters for a soft stepping linear ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0
+
+/*
+ * Exponential ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1
+
+/*
+ * Logarithmic ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2
+
+/* Structure for holding soft stepping volume parameters. */
+
+
+/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+struct asm_soft_step_volume_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+ uint32_t step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+ uint32_t ramping_curve;
+/*< Ramping curve type.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Structure for holding soft pause parameters. */
+
+
+/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_soft_pause_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t enable_flag;
+/*< Specifies whether soft pause is disabled (0) or enabled
+ * (nonzero).
+ */
+
+
+
+ uint32_t period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+ uint32_t step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+ uint32_t ramping_curve;
+/*< Ramping curve.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Maximum number of channels.*/
+#define VOLUME_CONTROL_MAX_CHANNELS 8
+
+/* Structure for holding one channel type - gain pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel
+ * type/gain pairs used by the Volume Control module. \n \n This
+ * structure immediately follows the
+ * asm_volume_ctrl_multichannel_gain structure.
+ */
+
+
+struct asm_volume_ctrl_channeltype_gain_pair {
+ uint8_t channeltype;
+ /*
+ * Channel type for which the gain setting is to be applied.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ */
+
+ uint8_t reserved1;
+ /* Clients must set this field to zero. */
+
+ uint8_t reserved2;
+ /* Clients must set this field to zero. */
+
+ uint8_t reserved3;
+ /* Clients must set this field to zero. */
+
+ uint32_t gain;
+ /*
+ * Gain value for this channel in Q28 format.
+ * Supported values: Any
+ */
+} __packed;
+
+
+/* Structure for the multichannel gain command */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t num_channels;
+ /*
+ * Number of channels for which gain values are provided. Any
+ * channels present in the data for which gain is not provided are
+ * set to unity gain.
+ * Supported values: 1 to 8
+ */
+
+ struct asm_volume_ctrl_channeltype_gain_pair
+ gain_data[VOLUME_CONTROL_MAX_CHANNELS];
+ /* Array of channel type/gain pairs.*/
+} __packed;
+
+
+/* Structure for holding one channel type - mute pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel
+ * type/mute setting pairs used by the Volume Control module. \n \n
+ * This structure immediately follows the
+ * asm_volume_ctrl_multichannel_mute structure.
+ */
+
+
+struct asm_volume_ctrl_channelype_mute_pair {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint8_t channelype;
+/*< Channel type for which the mute setting is to be applied.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ */
+
+ uint8_t reserved1;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved2;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved3;
+ /*< Clients must set this field to zero. */
+
+ uint32_t mute;
+/*< Mute setting for this channel.
+ * Supported values:
+ * - 0 = Unmute
+ * - Nonzero = Mute
+ */
+} __packed;
+
+
+/* Structure for the multichannel mute command */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_mute {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t num_channels;
+/*< Number of channels for which mute configuration is
+ * provided. Any channels present in the data for which mute
+ * configuration is not provided are set to unmute.
+ * Supported values: 1 to 8
+ */
+
+struct asm_volume_ctrl_channelype_mute_pair
+ mute_data[VOLUME_CONTROL_MAX_CHANNELS];
+ /*< Array of channel type/mute setting pairs.*/
+} __packed;
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the IIR Tuning Filter module.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the IIR tuning filter enable parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ * @messagepayload
+ * @structure{asm_iiruning_filter_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO
+ * NFIG.tex}
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03
+
+/* ID of the IIR tuning filter pregain parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04
+
+/* ID of the IIR tuning filter configuration parameters used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05
+
+/* Structure for an enable configuration parameter for an
+ * IIR tuning filter module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * parameter used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_enable {
+ uint32_t enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (1).
+ */
+} __packed;
+
+/* Structure for the pregain parameter for an IIR tuning filter module. */
+
+
+/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * parameters used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_pregain {
+ uint16_t pregain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* Structure for the configuration parameter for an IIR tuning filter
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ * parameters used by the IIR Tuning Filter module. \n
+ * \n
+ * This structure is followed by the IIR filter coefficients: \n
+ * - Sequence of int32_t FilterCoeffs \n
+ * Five coefficients for each band. Each coefficient is in int32_t format, in
+ * the order of b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor \n
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting \n
+ * One uint16_t per band, indicating if the filter is applied to left (0),
+ * right (1), or both (2) channels.
+ */
+struct asm_iir_filter_config_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* audio_pp_module_ids
+ * ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx
+ * paths.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_MBDRC_ENABLE
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_MBDRC 0x00010C06
+
+/* audio_pp_param_ids */
+/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex}
+ */
+#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07
+
+/* ID of the MBDRC configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex}
+ *
+ * @parspace Sub-band DRC configuration parameters
+ * @structure{asm_subband_drc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex}
+ *
+ * @keep{6}
+ * To obtain legacy ADRC from MBDRC, use the calibration tool to:
+ *
+ * - Enable MBDRC (EnableFlag = TRUE)
+ * - Set number of bands to 1 (uiNumBands = 1)
+ * - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1)
+ * - Clear the first band mute flag (MuteFlag[0] = 0)
+ * - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000)
+ * - Use the legacy ADRC parameters to calibrate the rest of the MBDRC
+ * parameters.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the MMBDRC module version 2 pre/postprocessing block.
+ * This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in
+ * the length of the filters used in each sub-band.
+ * This module supports the following parameter ID:
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2
+ */
+#define ASM_MODULE_ID_MBDRCV2 0x0001070B
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure
+ * of the MBDRC v2 pre/postprocessing block.
+ * The update to this configuration structure from the original
+ * MBDRC is the number of filter coefficients in the filter
+ * structure. The sequence for is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t
+ * padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags +
+ * uint16_t padding
+ * This block uses the same parameter structure as
+ * #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \
+ 0x0001070C
+
+#define ASM_MODULE_ID_MBDRCV3 0x0001090B
+/*
+ * ID of the MMBDRC module version 3 pre/postprocessing block.
+ * This module differs from MBDRCv2 (#ASM_MODULE_ID_MBDRCV2) in
+ * that it supports both 16- and 24-bit data.
+ * This module supports the following parameter ID:
+ * - #ASM_PARAM_ID_MBDRC_ENABLE
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_V3
+ * - #ASM_PARAM_ID_MBDRC_FILTER_XOVER_FREQS
+ */
+
+/* Structure for the enable parameter for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the
+ * MBDRC module.
+ */
+struct asm_mbdrc_enable {
+ uint32_t enable_flag;
+/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/
+} __packed;
+
+/* Structure for the configuration parameters for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ * parameters used by the MBDRC module. \n \n Following this
+ * structure is the payload for sub-band DRC configuration
+ * parameters (asm_subband_drc_config_params). This sub-band
+ * structure must be repeated for each band.
+ */
+
+
+struct asm_mbdrc_config_params {
+ uint16_t num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 5
+ */
+
+ int16_t limiterhreshold;
+/*< Threshold in decibels for the limiter output.
+ * Supported values: -72 to 18 \n
+ * Recommended value: 3994 (-0.22 db in Q3.12 format)
+ */
+
+ int16_t limiter_makeup_gain;
+/*< Makeup gain in decibels for the limiter output.
+ * Supported values: -42 to 42 \n
+ * Recommended value: 256 (0 dB in Q7.8 format)
+ */
+
+ int16_t limiter_gc;
+/*< Limiter gain recovery coefficient.
+ * Supported values: 0.5 to 0.99 \n
+ * Recommended value: 32440 (0.99 in Q15 format)
+ */
+
+ int16_t limiter_delay;
+/*< Limiter delay in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+
+ int16_t limiter_max_wait;
+/*< Maximum limiter waiting time in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+} __packed;
+
+/* DRC configuration structure for each sub-band of an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC
+ * configuration parameters for each sub-band in the MBDRC module.
+ * After this DRC structure is configured for valid bands, the next
+ * MBDRC setparams expects the sequence of sub-band MBDRC filter
+ * coefficients (the length depends on the number of bands) plus the
+ * mute flag for that band plus uint16_t padding.
+ *
+ * @keep{10}
+ * The filter coefficient and mute flag are of type int16_t:
+ * - FIR coefficient = int16_t firFilter
+ * - Mute flag = int16_t fMuteFlag
+ *
+ * The sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding
+ *
+ * For improved filterbank, the sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding
+ */
+struct asm_subband_drc_config_params {
+ int16_t drc_stereo_linked_flag;
+/*< Specifies whether all stereo channels have the same applied
+ * dynamics (1) or if they process their dynamics independently (0).
+ * Supported values:
+ * - 0 -- Not linked
+ * - 1 -- Linked
+ */
+
+ int16_t drc_mode;
+/*< Specifies whether DRC mode is bypassed for sub-bands.
+ * Supported values:
+ * - 0 -- Disabled
+ * - 1 -- Enabled
+ */
+
+ int16_t drc_down_sample_level;
+/*< DRC down sample level.
+ * Supported values: @ge 1
+ */
+
+ int16_t drc_delay;
+/*< DRC delay in samples.
+ * Supported values: 0 to 1200
+ */
+
+ uint16_t drc_rmsime_avg_const;
+/*< RMS signal energy time-averaging constant.
+ * Supported values: 0 to 2^16-1
+ */
+
+ uint16_t drc_makeup_gain;
+/*< DRC makeup gain in decibels.
+ * Supported values: 258 to 64917
+ */
+ /* Down expander settings */
+ int16_t down_expdrhreshold;
+/*< Down expander threshold.
+ * Supported Q7 format values: 1320 to up_cmpsrhreshold
+ */
+
+ int16_t down_expdr_slope;
+/*< Down expander slope.
+ * Supported Q8 format values: -32768 to 0.
+ */
+
+ uint32_t down_expdr_attack;
+/*< Down expander attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t down_expdr_release;
+/*< Down expander release constant.
+ * Supported Q31 format values: 19685 to 2^31
+ */
+
+ uint16_t down_expdr_hysteresis;
+/*< Down expander hysteresis constant.
+ * Supported Q14 format values: 1 to 32690
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero. */
+
+ int32_t down_expdr_min_gain_db;
+/*< Down expander minimum gain.
+ * Supported Q23 format values: -805306368 to 0.
+ */
+
+ /* Up compressor settings */
+
+ int16_t up_cmpsrhreshold;
+/*< Up compressor threshold.
+ * Supported Q7 format values: down_expdrhreshold to
+ * down_cmpsrhreshold.
+ */
+
+ uint16_t up_cmpsr_slope;
+/*< Up compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+ uint32_t up_cmpsr_attack;
+/*< Up compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t up_cmpsr_release;
+/*< Up compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+ uint16_t up_cmpsr_hysteresis;
+/*< Up compressor hysteresis constant.
+ * Supported Q14 format values: 1 to 32690.
+ */
+
+ /* Down compressor settings */
+
+ int16_t down_cmpsrhreshold;
+/*< Down compressor threshold.
+ * Supported Q7 format values: up_cmpsrhreshold to 11560.
+ */
+
+ uint16_t down_cmpsr_slope;
+/*< Down compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+ uint16_t reserved1;
+/*< Clients must set this field to zero. */
+
+ uint32_t down_cmpsr_attack;
+/*< Down compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t down_cmpsr_release;
+/*< Down compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+ uint16_t down_cmpsr_hysteresis;
+/*< Down compressor hysteresis constant.
+ * Supported Q14 values: 1 to 32690.
+ */
+
+ uint16_t reserved2;
+/*< Clients must set this field to zero.*/
+} __packed;
+
+#define ASM_MODULE_ID_EQUALIZER 0x00010C27
+#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28
+
+#define ASM_MAX_EQ_BANDS 12
+
+struct asm_eq_per_band_params {
+ uint32_t band_idx;
+/*< Band index.
+ * Supported values: 0 to 11
+ */
+
+ uint32_t filterype;
+/*< Type of filter.
+ * Supported values:
+ * - #ASM_PARAM_EQYPE_NONE
+ * - #ASM_PARAM_EQ_BASS_BOOST
+ * - #ASM_PARAM_EQ_BASS_CUT
+ * - #ASM_PARAM_EQREBLE_BOOST
+ * - #ASM_PARAM_EQREBLE_CUT
+ * - #ASM_PARAM_EQ_BAND_BOOST
+ * - #ASM_PARAM_EQ_BAND_CUT
+ */
+
+ uint32_t center_freq_hz;
+ /*< Filter band center frequency in Hertz. */
+
+ int32_t filter_gain;
+/*< Filter band initial gain.
+ * Supported values: +12 to -12 dB in 1 dB increments
+ */
+
+ int32_t q_factor;
+/*< Filter band quality factor expressed as a Q8 number, i.e., a
+ * fixed-point number with q factor of 8. For example, 3000/(2^8).
+ */
+} __packed;
+
+struct asm_eq_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t enable_flag;
+/*< Specifies whether the equalizer module is disabled (0) or enabled
+ * (nonzero).
+ */
+
+ uint32_t num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 12
+ */
+ struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS];
+
+} __packed;
+
+/* No equalizer effect.*/
+#define ASM_PARAM_EQYPE_NONE 0
+
+/* Bass boost equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_BOOST 1
+
+/*Bass cut equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_CUT 2
+
+/* Treble boost equalizer effect */
+#define ASM_PARAM_EQREBLE_BOOST 3
+
+/* Treble cut equalizer effect.*/
+#define ASM_PARAM_EQREBLE_CUT 4
+
+/* Band boost equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_BOOST 5
+
+/* Band cut equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_CUT 6
+
+/* Get & set params */
+#define VSS_ICOMMON_CMD_SET_PARAM_V2 0x0001133D
+#define VSS_ICOMMON_CMD_GET_PARAM_V2 0x0001133E
+#define VSS_ICOMMON_RSP_GET_PARAM 0x00011008
+
+/* ID of the Bass Boost module.
+ * This module supports the following parameter IDs:
+ * - #AUDPROC_PARAM_ID_BASS_BOOST_ENABLE
+ * - #AUDPROC_PARAM_ID_BASS_BOOST_MODE
+ * - #AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH
+ */
+#define AUDPROC_MODULE_ID_BASS_BOOST 0x000108A1
+/* ID of the Bass Boost enable parameter used by
+ * AUDPROC_MODULE_ID_BASS_BOOST.
+ */
+#define AUDPROC_PARAM_ID_BASS_BOOST_ENABLE 0x000108A2
+/* ID of the Bass Boost mode parameter used by
+ * AUDPROC_MODULE_ID_BASS_BOOST.
+ */
+#define AUDPROC_PARAM_ID_BASS_BOOST_MODE 0x000108A3
+/* ID of the Bass Boost strength parameter used by
+ * AUDPROC_MODULE_ID_BASS_BOOST.
+ */
+#define AUDPROC_PARAM_ID_BASS_BOOST_STRENGTH 0x000108A4
+
+/* ID of the PBE module.
+ * This module supports the following parameter IDs:
+ * - #AUDPROC_PARAM_ID_PBE_ENABLE
+ * - #AUDPROC_PARAM_ID_PBE_PARAM_CONFIG
+ */
+#define AUDPROC_MODULE_ID_PBE 0x00010C2A
+/* ID of the Bass Boost enable parameter used by
+ * AUDPROC_MODULE_ID_BASS_BOOST.
+ */
+#define AUDPROC_PARAM_ID_PBE_ENABLE 0x00010C2B
+/* ID of the Bass Boost mode parameter used by
+ * AUDPROC_MODULE_ID_BASS_BOOST.
+ */
+#define AUDPROC_PARAM_ID_PBE_PARAM_CONFIG 0x00010C49
+
+/* ID of the Virtualizer module. This module supports the
+ * following parameter IDs:
+ * - #AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE
+ * - #AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH
+ * - #AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE
+ * - #AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST
+ */
+#define AUDPROC_MODULE_ID_VIRTUALIZER 0x000108A5
+/* ID of the Virtualizer enable parameter used by
+ * AUDPROC_MODULE_ID_VIRTUALIZER.
+ */
+#define AUDPROC_PARAM_ID_VIRTUALIZER_ENABLE 0x000108A6
+/* ID of the Virtualizer strength parameter used by
+ * AUDPROC_MODULE_ID_VIRTUALIZER.
+ */
+#define AUDPROC_PARAM_ID_VIRTUALIZER_STRENGTH 0x000108A7
+/* ID of the Virtualizer out type parameter used by
+ * AUDPROC_MODULE_ID_VIRTUALIZER.
+ */
+#define AUDPROC_PARAM_ID_VIRTUALIZER_OUT_TYPE 0x000108A8
+/* ID of the Virtualizer out type parameter used by
+ * AUDPROC_MODULE_ID_VIRTUALIZER.
+ */
+#define AUDPROC_PARAM_ID_VIRTUALIZER_GAIN_ADJUST 0x000108A9
+
+/* ID of the Reverb module. This module supports the following
+ * parameter IDs:
+ * - #AUDPROC_PARAM_ID_REVERB_ENABLE
+ * - #AUDPROC_PARAM_ID_REVERB_MODE
+ * - #AUDPROC_PARAM_ID_REVERB_PRESET
+ * - #AUDPROC_PARAM_ID_REVERB_WET_MIX
+ * - #AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST
+ * - #AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL
+ * - #AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL
+ * - #AUDPROC_PARAM_ID_REVERB_DECAY_TIME
+ * - #AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO
+ * - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL
+ * - #AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY
+ * - #AUDPROC_PARAM_ID_REVERB_LEVEL
+ * - #AUDPROC_PARAM_ID_REVERB_DELAY
+ * - #AUDPROC_PARAM_ID_REVERB_DIFFUSION
+ * - #AUDPROC_PARAM_ID_REVERB_DENSITY
+ */
+#define AUDPROC_MODULE_ID_REVERB 0x000108AA
+/* ID of the Reverb enable parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_ENABLE 0x000108AB
+/* ID of the Reverb mode parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_MODE 0x000108AC
+/* ID of the Reverb preset parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_PRESET 0x000108AD
+/* ID of the Reverb wet mix parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_WET_MIX 0x000108AE
+/* ID of the Reverb gain adjust parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_GAIN_ADJUST 0x000108AF
+/* ID of the Reverb room level parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_ROOM_LEVEL 0x000108B0
+/* ID of the Reverb room hf level parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_ROOM_HF_LEVEL 0x000108B1
+/* ID of the Reverb decay time parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_DECAY_TIME 0x000108B2
+/* ID of the Reverb decay hf ratio parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_DECAY_HF_RATIO 0x000108B3
+/* ID of the Reverb reflections level parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_LEVEL 0x000108B4
+/* ID of the Reverb reflections delay parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_REFLECTIONS_DELAY 0x000108B5
+/* ID of the Reverb level parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_LEVEL 0x000108B6
+/* ID of the Reverb delay parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_DELAY 0x000108B7
+/* ID of the Reverb diffusion parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_DIFFUSION 0x000108B8
+/* ID of the Reverb density parameter used by
+ * AUDPROC_MODULE_ID_REVERB.
+ */
+#define AUDPROC_PARAM_ID_REVERB_DENSITY 0x000108B9
+
+/* ID of the Popless Equalizer module. This module supports the
+ * following parameter IDs:
+ * - #AUDPROC_PARAM_ID_EQ_ENABLE
+ * - #AUDPROC_PARAM_ID_EQ_CONFIG
+ * - #AUDPROC_PARAM_ID_EQ_NUM_BANDS
+ * - #AUDPROC_PARAM_ID_EQ_BAND_LEVELS
+ * - #AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE
+ * - #AUDPROC_PARAM_ID_EQ_BAND_FREQS
+ * - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE
+ * - #AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ
+ * - #AUDPROC_PARAM_ID_EQ_BAND_INDEX
+ * - #AUDPROC_PARAM_ID_EQ_PRESET_ID
+ * - #AUDPROC_PARAM_ID_EQ_NUM_PRESETS
+ * - #AUDPROC_PARAM_ID_EQ_GET_PRESET_NAME
+ */
+#define AUDPROC_MODULE_ID_POPLESS_EQUALIZER 0x000108BA
+/* ID of the Popless Equalizer enable parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+ */
+#define AUDPROC_PARAM_ID_EQ_ENABLE 0x000108BB
+/* ID of the Popless Equalizer config parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+ */
+#define AUDPROC_PARAM_ID_EQ_CONFIG 0x000108BC
+/* ID of the Popless Equalizer number of bands parameter used
+ * by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+ * used for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_NUM_BANDS 0x000108BD
+/* ID of the Popless Equalizer band levels parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+ * used for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_BAND_LEVELS 0x000108BE
+/* ID of the Popless Equalizer band level range parameter used
+ * by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+ * used for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_BAND_LEVEL_RANGE 0x000108BF
+/* ID of the Popless Equalizer band frequencies parameter used
+ * by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is
+ * used for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_BAND_FREQS 0x000108C0
+/* ID of the Popless Equalizer single band frequency range
+ * parameter used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+ * This param ID is used for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ_RANGE 0x000108C1
+/* ID of the Popless Equalizer single band frequency parameter
+ * used by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID
+ * is used for set param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_SINGLE_BAND_FREQ 0x000108C2
+/* ID of the Popless Equalizer band index parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER.
+ */
+#define AUDPROC_PARAM_ID_EQ_BAND_INDEX 0x000108C3
+/* ID of the Popless Equalizer preset id parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+ * for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_PRESET_ID 0x000108C4
+/* ID of the Popless Equalizer number of presets parameter used
+ * by AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+ * for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_NUM_PRESETS 0x000108C5
+/* ID of the Popless Equalizer preset name parameter used by
+ * AUDPROC_MODULE_ID_POPLESS_EQUALIZER. This param ID is used
+ * for get param only.
+ */
+#define AUDPROC_PARAM_ID_EQ_PRESET_NAME 0x000108C6
+
+/* Set Q6 topologies */
+#define ASM_CMD_ADD_TOPOLOGIES 0x00010DBE
+#define ADM_CMD_ADD_TOPOLOGIES 0x00010335
+#define AFE_CMD_ADD_TOPOLOGIES 0x000100f8
+/* structure used for both ioctls */
+struct cmd_set_topologies {
+ struct apr_hdr hdr;
+ u32 payload_addr_lsw;
+ /* LSW of parameter data payload address.*/
+ u32 payload_addr_msw;
+ /* MSW of parameter data payload address.*/
+ u32 mem_map_handle;
+ /* Memory map handle returned by mem map command */
+ u32 payload_size;
+ /* Size in bytes of the variable payload in shared memory */
+} __packed;
+
+/* This module represents the Rx processing of Feedback speaker protection.
+ * It contains the excursion control, thermal protection,
+ * analog clip manager features in it.
+ * This module id will support following param ids.
+ * - AFE_PARAM_ID_FBSP_MODE_RX_CFG
+ */
+
+#define AFE_MODULE_FB_SPKR_PROT_RX 0x0001021C
+#define AFE_MODULE_FB_SPKR_PROT_V2_RX 0x0001025F
+
+#define AFE_PARAM_ID_FBSP_MODE_RX_CFG 0x0001021D
+#define AFE_PARAM_ID_FBSP_PTONE_RAMP_CFG 0x00010260
+
+struct asm_fbsp_mode_rx_cfg {
+ uint32_t minor_version;
+ uint32_t mode;
+} __packed;
+
+/* This module represents the VI processing of feedback speaker protection.
+ * It will receive Vsens and Isens from codec and generates necessary
+ * parameters needed by Rx processing.
+ * This module id will support following param ids.
+ * - AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG
+ * - AFE_PARAM_ID_CALIB_RES_CFG
+ * - AFE_PARAM_ID_FEEDBACK_PATH_CFG
+ */
+
+#define AFE_MODULE_FB_SPKR_PROT_VI_PROC 0x00010226
+#define AFE_MODULE_FB_SPKR_PROT_VI_PROC_V2 0x0001026A
+
+#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG 0x0001022A
+#define AFE_PARAM_ID_SPKR_CALIB_VI_PROC_CFG_V2 0x0001026B
+
+struct asm_spkr_calib_vi_proc_cfg {
+ uint32_t minor_version;
+ uint32_t operation_mode;
+ uint32_t r0_t0_selection_flag[SP_V2_NUM_MAX_SPKR];
+ int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
+ int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR];
+ uint32_t quick_calib_flag;
+} __packed;
+
+#define AFE_PARAM_ID_CALIB_RES_CFG 0x0001022B
+#define AFE_PARAM_ID_CALIB_RES_CFG_V2 0x0001026E
+
+struct asm_calib_res_cfg {
+ uint32_t minor_version;
+ int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
+ uint32_t th_vi_ca_state;
+} __packed;
+
+#define AFE_PARAM_ID_FEEDBACK_PATH_CFG 0x0001022C
+#define AFE_MODULE_FEEDBACK 0x00010257
+
+struct asm_feedback_path_cfg {
+ uint32_t minor_version;
+ int32_t dst_portid;
+ int32_t num_channels;
+ int32_t chan_info[4];
+} __packed;
+
+#define AFE_PARAM_ID_MODE_VI_PROC_CFG 0x00010227
+
+struct asm_mode_vi_proc_cfg {
+ uint32_t minor_version;
+ uint32_t cal_mode;
+} __packed;
+
+#define AFE_MODULE_SPEAKER_PROTECTION_V2_TH_VI 0x0001026A
+#define AFE_PARAM_ID_SP_V2_TH_VI_MODE_CFG 0x0001026B
+#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_CFG 0x0001029F
+#define AFE_PARAM_ID_SP_V2_TH_VI_FTM_PARAMS 0x000102A0
+
+struct afe_sp_th_vi_mode_cfg {
+ uint32_t minor_version;
+ uint32_t operation_mode;
+ /*
+ * Operation mode of thermal VI module.
+ * 0 -- Normal Running mode
+ * 1 -- Calibration mode
+ * 2 -- FTM mode
+ */
+ uint32_t r0t0_selection_flag[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Specifies which set of R0, T0 values the algorithm will use.
+ * This field is valid only in Normal mode (operation_mode = 0).
+ * 0 -- Use calibrated R0, T0 value
+ * 1 -- Use safe R0, T0 value
+ */
+ int32_t r0_cali_q24[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Calibration point resistance per device. This field is valid
+ * only in Normal mode (operation_mode = 0).
+ * values 33554432 to 1073741824 Ohms (in Q24 format)
+ */
+ int16_t t0_cali_q6[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Calibration point temperature per device. This field is valid
+ * in both Normal mode and Calibration mode.
+ * values -1920 to 5120 degrees C (in Q6 format)
+ */
+ uint32_t quick_calib_flag;
+ /*
+ * Indicates whether calibration is to be done in quick mode or not.
+ * This field is valid only in Calibration mode (operation_mode = 1).
+ * 0 -- Disabled
+ * 1 -- Enabled
+ */
+} __packed;
+
+struct afe_sp_th_vi_ftm_cfg {
+ uint32_t minor_version;
+ uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Wait time to heat up speaker before collecting statistics
+ * for ftm mode in ms.
+ * values 0 to 4294967295 ms
+ */
+ uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
+ /*
+ * duration for which FTM statistics are collected in ms.
+ * values 0 to 2000 ms
+ */
+} __packed;
+
+struct afe_sp_th_vi_ftm_params {
+ uint32_t minor_version;
+ int32_t dc_res_q24[SP_V2_NUM_MAX_SPKR];
+ /*
+ * DC resistance value in q24 format
+ * values 0 to 2147483647 Ohms (in Q24 format)
+ */
+ int32_t temp_q22[SP_V2_NUM_MAX_SPKR];
+ /*
+ * temperature value in q22 format
+ * values -125829120 to 2147483647 degC (in Q22 format)
+ */
+ uint32_t status[SP_V2_NUM_MAX_SPKR];
+ /*
+ * FTM packet status
+ * 0 - Incorrect operation mode.This status is returned
+ * when GET_PARAM is called in non FTM Mode
+ * 1 - Inactive mode -- Port is not yet started.
+ * 2 - Wait state. wait_time_ms has not yet elapsed
+ * 3 - In progress state. ftm_time_ms has not yet elapsed.
+ * 4 - Success.
+ * 5 - Failed.
+ */
+} __packed;
+
+struct afe_sp_th_vi_get_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_get_param_v2 get_param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_sp_th_vi_ftm_params param;
+} __packed;
+
+struct afe_sp_th_vi_get_param_resp {
+ uint32_t status;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_sp_th_vi_ftm_params param;
+} __packed;
+
+
+#define AFE_MODULE_SPEAKER_PROTECTION_V2_EX_VI 0x0001026F
+#define AFE_PARAM_ID_SP_V2_EX_VI_MODE_CFG 0x000102A1
+#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_CFG 0x000102A2
+#define AFE_PARAM_ID_SP_V2_EX_VI_FTM_PARAMS 0x000102A3
+
+struct afe_sp_ex_vi_mode_cfg {
+ uint32_t minor_version;
+ uint32_t operation_mode;
+ /*
+ * Operation mode of Excursion VI module.
+ * 0 - Normal Running mode
+ * 2 - FTM mode
+ */
+} __packed;
+
+struct afe_sp_ex_vi_ftm_cfg {
+ uint32_t minor_version;
+ uint32_t wait_time_ms[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Wait time to heat up speaker before collecting statistics
+ * for ftm mode in ms.
+ * values 0 to 4294967295 ms
+ */
+ uint32_t ftm_time_ms[SP_V2_NUM_MAX_SPKR];
+ /*
+ * duration for which FTM statistics are collected in ms.
+ * values 0 to 2000 ms
+ */
+} __packed;
+
+struct afe_sp_ex_vi_ftm_params {
+ uint32_t minor_version;
+ int32_t freq_q20[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Resonance frequency in q20 format
+ * values 0 to 2147483647 Hz (in Q20 format)
+ */
+ int32_t resis_q24[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Mechanical resistance in q24 format
+ * values 0 to 2147483647 Ohms (in Q24 format)
+ */
+ int32_t qmct_q24[SP_V2_NUM_MAX_SPKR];
+ /*
+ * Mechanical Qfactor in q24 format
+ * values 0 to 2147483647 (in Q24 format)
+ */
+ uint32_t status[SP_V2_NUM_MAX_SPKR];
+ /*
+ * FTM packet status
+ * 0 - Incorrect operation mode.This status is returned
+ * when GET_PARAM is called in non FTM Mode.
+ * 1 - Inactive mode -- Port is not yet started.
+ * 2 - Wait state. wait_time_ms has not yet elapsed
+ * 3 - In progress state. ftm_time_ms has not yet elapsed.
+ * 4 - Success.
+ * 5 - Failed.
+ */
+} __packed;
+
+struct afe_sp_ex_vi_get_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_get_param_v2 get_param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_sp_ex_vi_ftm_params param;
+} __packed;
+
+struct afe_sp_ex_vi_get_param_resp {
+ uint32_t status;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_sp_ex_vi_ftm_params param;
+} __packed;
+
+union afe_spkr_prot_config {
+ struct asm_fbsp_mode_rx_cfg mode_rx_cfg;
+ struct asm_spkr_calib_vi_proc_cfg vi_proc_cfg;
+ struct asm_feedback_path_cfg feedback_path_cfg;
+ struct asm_mode_vi_proc_cfg mode_vi_proc_cfg;
+ struct afe_sp_th_vi_mode_cfg th_vi_mode_cfg;
+ struct afe_sp_th_vi_ftm_cfg th_vi_ftm_cfg;
+ struct afe_sp_ex_vi_mode_cfg ex_vi_mode_cfg;
+ struct afe_sp_ex_vi_ftm_cfg ex_vi_ftm_cfg;
+} __packed;
+
+struct afe_spkr_prot_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union afe_spkr_prot_config prot_config;
+} __packed;
+
+struct afe_spkr_prot_get_vi_calib {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_get_param_v2 get_param;
+ struct afe_port_param_data_v2 pdata;
+ struct asm_calib_res_cfg res_cfg;
+} __packed;
+
+struct afe_spkr_prot_calib_get_resp {
+ uint32_t status;
+ struct afe_port_param_data_v2 pdata;
+ struct asm_calib_res_cfg res_cfg;
+} __packed;
+
+
+/* SRS TRUMEDIA start */
+/* topology */
+#define SRS_TRUMEDIA_TOPOLOGY_ID 0x00010D90
+/* module */
+#define SRS_TRUMEDIA_MODULE_ID 0x10005010
+/* parameters */
+#define SRS_TRUMEDIA_PARAMS 0x10005011
+#define SRS_TRUMEDIA_PARAMS_WOWHD 0x10005012
+#define SRS_TRUMEDIA_PARAMS_CSHP 0x10005013
+#define SRS_TRUMEDIA_PARAMS_HPF 0x10005014
+#define SRS_TRUMEDIA_PARAMS_AEQ 0x10005015
+#define SRS_TRUMEDIA_PARAMS_HL 0x10005016
+#define SRS_TRUMEDIA_PARAMS_GEQ 0x10005017
+
+#define SRS_ID_GLOBAL 0x00000001
+#define SRS_ID_WOWHD 0x00000002
+#define SRS_ID_CSHP 0x00000003
+#define SRS_ID_HPF 0x00000004
+#define SRS_ID_AEQ 0x00000005
+#define SRS_ID_HL 0x00000006
+#define SRS_ID_GEQ 0x00000007
+
+#define SRS_CMD_UPLOAD 0x7FFF0000
+#define SRS_PARAM_OFFSET_MASK 0x3FFF0000
+#define SRS_PARAM_VALUE_MASK 0x0000FFFF
+
+struct srs_trumedia_params_GLOBAL {
+ uint8_t v1;
+ uint8_t v2;
+ uint8_t v3;
+ uint8_t v4;
+ uint8_t v5;
+ uint8_t v6;
+ uint8_t v7;
+ uint8_t v8;
+ uint16_t v9;
+} __packed;
+
+struct srs_trumedia_params_WOWHD {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v5;
+ uint16_t v6;
+ uint16_t v7;
+ uint16_t v8;
+ uint16_t v____A1;
+ uint32_t v9;
+ uint16_t v10;
+ uint16_t v11;
+ uint32_t v12[16];
+ uint32_t v13[16];
+ uint32_t v14[16];
+ uint32_t v15[16];
+ uint32_t v16;
+ uint16_t v17;
+ uint16_t v18;
+} __packed;
+
+struct srs_trumedia_params_CSHP {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v5;
+ uint16_t v6;
+ uint16_t v____A1;
+ uint32_t v7;
+ uint16_t v8;
+ uint16_t v9;
+ uint32_t v10[16];
+} __packed;
+
+struct srs_trumedia_params_HPF {
+ uint32_t v1;
+ uint32_t v2[26];
+} __packed;
+
+struct srs_trumedia_params_AEQ {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v____A1;
+ uint32_t v5[74];
+ uint32_t v6[74];
+ uint16_t v7[2048];
+} __packed;
+
+struct srs_trumedia_params_HL {
+ uint16_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v____A1;
+ int32_t v4;
+ uint32_t v5;
+ uint16_t v6;
+ uint16_t v____A2;
+ uint32_t v7;
+} __packed;
+
+struct srs_trumedia_params_GEQ {
+ int16_t v1[10];
+} __packed;
+struct srs_trumedia_params {
+ struct srs_trumedia_params_GLOBAL global;
+ struct srs_trumedia_params_WOWHD wowhd;
+ struct srs_trumedia_params_CSHP cshp;
+ struct srs_trumedia_params_HPF hpf;
+ struct srs_trumedia_params_AEQ aeq;
+ struct srs_trumedia_params_HL hl;
+ struct srs_trumedia_params_GEQ geq;
+} __packed;
+/* SRS TruMedia end */
+
+#define AUDPROC_PARAM_ID_ENABLE 0x00010904
+#define ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF
+/* DTS Eagle */
+#define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C
+#define AUDPROC_MODULE_ID_DTS_HPX_POSTMIX 0x0001077B
+#define ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX 0x00010DED
+#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS 0x10015000
+#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER 0x10015001
+struct asm_dts_eagle_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+} __packed;
+
+struct asm_dts_eagle_param_get {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_get_pp_params_v2 param;
+} __packed;
+
+/* Opcode to set BT address and license for aptx decoder */
+#define APTX_DECODER_BT_ADDRESS 0x00013201
+#define APTX_CLASSIC_DEC_LICENSE_ID 0x00013202
+
+struct aptx_dec_bt_addr_cfg {
+ uint32_t lap;
+ uint32_t uap;
+ uint32_t nap;
+} __packed;
+
+struct aptx_dec_bt_dev_addr {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct aptx_dec_bt_addr_cfg bt_addr_cfg;
+} __packed;
+
+struct asm_aptx_dec_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: 44100 and 48000 Hz
+ */
+} __packed;
+
+/* LSM Specific */
+#define VW_FEAT_DIM (39)
+
+#define APRV2_IDS_SERVICE_ID_ADSP_LSM_V (0xD)
+#define APRV2_IDS_DOMAIN_ID_ADSP_V (0x4)
+#define APRV2_IDS_DOMAIN_ID_APPS_V (0x5)
+
+#define LSM_SESSION_CMD_SHARED_MEM_MAP_REGIONS (0x00012A7F)
+#define LSM_SESSION_CMDRSP_SHARED_MEM_MAP_REGIONS (0x00012A80)
+#define LSM_SESSION_CMD_SHARED_MEM_UNMAP_REGIONS (0x00012A81)
+#define LSM_SESSION_CMD_OPEN_TX (0x00012A82)
+#define LSM_SESSION_CMD_CLOSE_TX (0x00012A88)
+#define LSM_SESSION_CMD_SET_PARAMS (0x00012A83)
+#define LSM_SESSION_CMD_SET_PARAMS_V2 (0x00012A8F)
+#define LSM_SESSION_CMD_REGISTER_SOUND_MODEL (0x00012A84)
+#define LSM_SESSION_CMD_DEREGISTER_SOUND_MODEL (0x00012A85)
+#define LSM_SESSION_CMD_START (0x00012A86)
+#define LSM_SESSION_CMD_STOP (0x00012A87)
+#define LSM_SESSION_CMD_EOB (0x00012A89)
+#define LSM_SESSION_CMD_READ (0x00012A8A)
+#define LSM_SESSION_CMD_OPEN_TX_V2 (0x00012A8B)
+#define LSM_CMD_ADD_TOPOLOGIES (0x00012A8C)
+
+#define LSM_SESSION_EVENT_DETECTION_STATUS (0x00012B00)
+#define LSM_SESSION_EVENT_DETECTION_STATUS_V2 (0x00012B01)
+#define LSM_DATA_EVENT_READ_DONE (0x00012B02)
+#define LSM_DATA_EVENT_STATUS (0x00012B03)
+#define LSM_SESSION_EVENT_DETECTION_STATUS_V3 (0x00012B04)
+
+#define LSM_MODULE_ID_VOICE_WAKEUP (0x00012C00)
+#define LSM_PARAM_ID_ENDPOINT_DETECT_THRESHOLD (0x00012C01)
+#define LSM_PARAM_ID_OPERATION_MODE (0x00012C02)
+#define LSM_PARAM_ID_GAIN (0x00012C03)
+#define LSM_PARAM_ID_CONNECT_TO_PORT (0x00012C04)
+#define LSM_PARAM_ID_FEATURE_COMPENSATION_DATA (0x00012C07)
+#define LSM_PARAM_ID_MIN_CONFIDENCE_LEVELS (0x00012C07)
+#define LSM_MODULE_ID_LAB (0x00012C08)
+#define LSM_PARAM_ID_LAB_ENABLE (0x00012C09)
+#define LSM_PARAM_ID_LAB_CONFIG (0x00012C0A)
+#define LSM_MODULE_ID_FRAMEWORK (0x00012C0E)
+#define LSM_PARAM_ID_SWMAD_CFG (0x00012C18)
+#define LSM_PARAM_ID_SWMAD_MODEL (0x00012C19)
+#define LSM_PARAM_ID_SWMAD_ENABLE (0x00012C1A)
+#define LSM_PARAM_ID_POLLING_ENABLE (0x00012C1B)
+#define LSM_PARAM_ID_MEDIA_FMT (0x00012C1E)
+#define LSM_PARAM_ID_FWK_MODE_CONFIG (0x00012C27)
+
+/* HW MAD specific */
+#define AFE_MODULE_HW_MAD (0x00010230)
+#define AFE_PARAM_ID_HW_MAD_CFG (0x00010231)
+#define AFE_PARAM_ID_HW_MAD_CTRL (0x00010232)
+#define AFE_PARAM_ID_SLIMBUS_SLAVE_PORT_CFG (0x00010233)
+
+/* SW MAD specific */
+#define AFE_MODULE_SW_MAD (0x0001022D)
+#define AFE_PARAM_ID_SW_MAD_CFG (0x0001022E)
+#define AFE_PARAM_ID_SVM_MODEL (0x0001022F)
+
+/* Commands/Params to pass the codec/slimbus data to DSP */
+#define AFE_SVC_CMD_SET_PARAM (0x000100f3)
+#define AFE_MODULE_CDC_DEV_CFG (0x00010234)
+#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG (0x00010235)
+#define AFE_PARAM_ID_CDC_REG_CFG (0x00010236)
+#define AFE_PARAM_ID_CDC_REG_CFG_INIT (0x00010237)
+#define AFE_PARAM_ID_CDC_REG_PAGE_CFG (0x00010296)
+
+#define AFE_MAX_CDC_REGISTERS_TO_CONFIG (20)
+
+/* AANC Port Config Specific */
+#define AFE_PARAM_ID_AANC_PORT_CONFIG (0x00010215)
+#define AFE_API_VERSION_AANC_PORT_CONFIG (0x1)
+#define AANC_TX_MIC_UNUSED (0)
+#define AANC_TX_VOICE_MIC (1)
+#define AANC_TX_ERROR_MIC (2)
+#define AANC_TX_NOISE_MIC (3)
+#define AFE_PORT_MAX_CHANNEL_CNT (8)
+#define AFE_MODULE_AANC (0x00010214)
+#define AFE_PARAM_ID_CDC_AANC_VERSION (0x0001023A)
+#define AFE_API_VERSION_CDC_AANC_VERSION (0x1)
+#define AANC_HW_BLOCK_VERSION_1 (1)
+#define AANC_HW_BLOCK_VERSION_2 (2)
+
+/*Clip bank selection*/
+#define AFE_API_VERSION_CLIP_BANK_SEL_CFG 0x1
+#define AFE_CLIP_MAX_BANKS 4
+#define AFE_PARAM_ID_CLIP_BANK_SEL_CFG 0x00010242
+
+struct afe_param_aanc_port_cfg {
+ /* Minor version used for tracking the version of the module's
+ * source port configuration.
+ */
+ uint32_t aanc_port_cfg_minor_version;
+
+ /* Sampling rate of the source Tx port. 8k - 192k*/
+ uint32_t tx_port_sample_rate;
+
+ /* Channel mapping for the Tx port signal carrying Noise (X),
+ * Error (E), and Voice (V) signals.
+ */
+ uint8_t tx_port_channel_map[AFE_PORT_MAX_CHANNEL_CNT];
+
+ /* Number of channels on the source Tx port. */
+ uint16_t tx_port_num_channels;
+
+ /* Port ID of the Rx path reference signal. */
+ uint16_t rx_path_ref_port_id;
+
+ /* Sampling rate of the reference port. 8k - 192k*/
+ uint32_t ref_port_sample_rate;
+} __packed;
+
+struct afe_param_id_cdc_aanc_version {
+ /* Minor version used for tracking the version of the module's
+ * hw version
+ */
+ uint32_t cdc_aanc_minor_version;
+
+ /* HW version. */
+ uint32_t aanc_hw_version;
+} __packed;
+
+struct afe_param_id_clip_bank_sel {
+ /* Minor version used for tracking the version of the module's
+ * hw version
+ */
+ uint32_t minor_version;
+
+ /* Number of banks to be read */
+ uint32_t num_banks;
+
+ uint32_t bank_map[AFE_CLIP_MAX_BANKS];
+} __packed;
+
+/* ERROR CODES */
+/* Success. The operation completed with no errors. */
+#define ADSP_EOK 0x00000000
+/* General failure. */
+#define ADSP_EFAILED 0x00000001
+/* Bad operation parameter. */
+#define ADSP_EBADPARAM 0x00000002
+/* Unsupported routine or operation. */
+#define ADSP_EUNSUPPORTED 0x00000003
+/* Unsupported version. */
+#define ADSP_EVERSION 0x00000004
+/* Unexpected problem encountered. */
+#define ADSP_EUNEXPECTED 0x00000005
+/* Unhandled problem occurred. */
+#define ADSP_EPANIC 0x00000006
+/* Unable to allocate resource. */
+#define ADSP_ENORESOURCE 0x00000007
+/* Invalid handle. */
+#define ADSP_EHANDLE 0x00000008
+/* Operation is already processed. */
+#define ADSP_EALREADY 0x00000009
+/* Operation is not ready to be processed. */
+#define ADSP_ENOTREADY 0x0000000A
+/* Operation is pending completion. */
+#define ADSP_EPENDING 0x0000000B
+/* Operation could not be accepted or processed. */
+#define ADSP_EBUSY 0x0000000C
+/* Operation aborted due to an error. */
+#define ADSP_EABORTED 0x0000000D
+/* Operation preempted by a higher priority. */
+#define ADSP_EPREEMPTED 0x0000000E
+/* Operation requests intervention to complete. */
+#define ADSP_ECONTINUE 0x0000000F
+/* Operation requests immediate intervention to complete. */
+#define ADSP_EIMMEDIATE 0x00000010
+/* Operation is not implemented. */
+#define ADSP_ENOTIMPL 0x00000011
+/* Operation needs more data or resources. */
+#define ADSP_ENEEDMORE 0x00000012
+/* Operation does not have memory. */
+#define ADSP_ENOMEMORY 0x00000014
+/* Item does not exist. */
+#define ADSP_ENOTEXIST 0x00000015
+/* Max count for adsp error code sent to HLOS*/
+#define ADSP_ERR_MAX (ADSP_ENOTEXIST + 1)
+/* Operation is finished. */
+#define ADSP_ETERMINATED 0x00011174
+
+/*bharath, adsp_error_codes.h */
+
+/* LPASS clock for I2S Interface */
+
+/* Supported OSR clock values */
+#define Q6AFE_LPASS_OSR_CLK_12_P288_MHZ 0xBB8000
+#define Q6AFE_LPASS_OSR_CLK_11_P2896_MHZ 0xAC4400
+#define Q6AFE_LPASS_OSR_CLK_9_P600_MHZ 0x927C00
+#define Q6AFE_LPASS_OSR_CLK_8_P192_MHZ 0x7D0000
+#define Q6AFE_LPASS_OSR_CLK_6_P144_MHZ 0x5DC000
+#define Q6AFE_LPASS_OSR_CLK_4_P096_MHZ 0x3E8000
+#define Q6AFE_LPASS_OSR_CLK_3_P072_MHZ 0x2EE000
+#define Q6AFE_LPASS_OSR_CLK_2_P048_MHZ 0x1F4000
+#define Q6AFE_LPASS_OSR_CLK_1_P536_MHZ 0x177000
+#define Q6AFE_LPASS_OSR_CLK_1_P024_MHZ 0xFA000
+#define Q6AFE_LPASS_OSR_CLK_768_kHZ 0xBB800
+#define Q6AFE_LPASS_OSR_CLK_512_kHZ 0x7D000
+#define Q6AFE_LPASS_OSR_CLK_DISABLE 0x0
+
+/* Supported Bit clock values */
+#define Q6AFE_LPASS_IBIT_CLK_12_P288_MHZ 0xBB8000
+#define Q6AFE_LPASS_IBIT_CLK_11_P2896_MHZ 0xAC4400
+#define Q6AFE_LPASS_IBIT_CLK_8_P192_MHZ 0x7D0000
+#define Q6AFE_LPASS_IBIT_CLK_6_P144_MHZ 0x5DC000
+#define Q6AFE_LPASS_IBIT_CLK_4_P096_MHZ 0x3E8000
+#define Q6AFE_LPASS_IBIT_CLK_3_P072_MHZ 0x2EE000
+#define Q6AFE_LPASS_IBIT_CLK_2_P8224_MHZ 0x2b1100
+#define Q6AFE_LPASS_IBIT_CLK_2_P048_MHZ 0x1F4000
+#define Q6AFE_LPASS_IBIT_CLK_1_P536_MHZ 0x177000
+#define Q6AFE_LPASS_IBIT_CLK_1_P4112_MHZ 0x158880
+#define Q6AFE_LPASS_IBIT_CLK_1_P024_MHZ 0xFA000
+#define Q6AFE_LPASS_IBIT_CLK_768_KHZ 0xBB800
+#define Q6AFE_LPASS_IBIT_CLK_512_KHZ 0x7D000
+#define Q6AFE_LPASS_IBIT_CLK_256_KHZ 0x3E800
+#define Q6AFE_LPASS_IBIT_CLK_DISABLE 0x0
+
+/* Supported LPASS CLK sources */
+#define Q6AFE_LPASS_CLK_SRC_EXTERNAL 0
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+
+/* Supported LPASS CLK root*/
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
+
+enum afe_lpass_clk_mode {
+ Q6AFE_LPASS_MODE_BOTH_INVALID,
+ Q6AFE_LPASS_MODE_CLK1_VALID,
+ Q6AFE_LPASS_MODE_CLK2_VALID,
+ Q6AFE_LPASS_MODE_BOTH_VALID,
+} __packed;
+
+/* Clock ID Enumeration Define. */
+/* Clock ID for Primary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100
+/* Clock ID for Primary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101
+/* Clock ID for Secondary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102
+/* Clock ID for Secondary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103
+/* Clock ID for Tertiary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104
+/* Clock ID for Tertiary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105
+/* Clock ID for Quartnery I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106
+/* Clock ID for Quartnery I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107
+/* Clock ID for Speaker I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108
+/* Clock ID for Speaker I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109
+/* Clock ID for Speaker I2S OSR */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A
+
+/* Clock ID for QUINARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B
+/* Clock ID for QUINARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C
+/* Clock ID for SENARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D
+/* Clock ID for SENARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E
+/* Clock ID for INT0 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
+/* Clock ID for INT1 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
+/* Clock ID for INT2 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
+/* Clock ID for INT3 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
+/* Clock ID for INT4 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
+/* Clock ID for INT5 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
+/* Clock ID for INT6 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
+
+/* Clock ID for Primary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200
+/* Clock ID for Primary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201
+/* Clock ID for Secondary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202
+/* Clock ID for Secondary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203
+/* Clock ID for Tertiary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204
+/* Clock ID for Tertiary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205
+/* Clock ID for Quartery PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206
+/* Clock ID for Quartery PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207
+
+/** Clock ID for Primary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200
+/** Clock ID for Primary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201
+/** Clock ID for Secondary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202
+/** Clock ID for Secondary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203
+/** Clock ID for Tertiary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204
+/** Clock ID for Tertiary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205
+/** Clock ID for Quartery TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206
+/** Clock ID for Quartery TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207
+
+/* Clock ID for MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300
+/* Clock ID for MCLK2 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301
+/* Clock ID for MCLK3 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302
+/* Clock ID for MCLK4 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
+/* Clock ID for Internal Digital Codec Core */
+#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
+/* Clock ID for INT MCLK0 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
+/* Clock ID for INT MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
+/*
+ * Clock ID for soundwire NPL.
+ * This is the clock to be used to enable NPL clock for internal Soundwire.
+ */
+#define AFE_CLOCK_SET_CLOCK_ID_SWR_NPL_CLK 0x307
+
+/* Clock ID for AHB HDMI input */
+#define Q6AFE_LPASS_CLK_ID_AHB_HDMI_INPUT 0x400
+
+/* Clock ID for SPDIF core */
+#define Q6AFE_LPASS_CLK_ID_SPDIF_CORE 0x500
+
+
+/* Clock attribute for invalid use (reserved for internal usage) */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0
+/* Clock attribute for no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1
+/* Clock attribute for dividend couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2
+/* Clock attribute for divisor couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3
+/* Clock attribute for invert and no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4
+/* Clock set API version */
+#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1
+
+struct afe_clk_set {
+ /*
+ * Minor version used for tracking clock set.
+ * @values #AFE_API_VERSION_CLOCK_SET
+ */
+ uint32_t clk_set_minor_version;
+
+ /*
+ * Clock ID
+ * @values
+ * - 0x100 to 0x10A - MSM8996
+ * - 0x200 to 0x207 - MSM8996
+ * - 0x300 to 0x302 - MSM8996 @tablebulletend
+ */
+ uint32_t clk_id;
+
+ /*
+ * Clock frequency (in Hertz) to be set.
+ * @values
+ * - >= 0 for clock frequency to set @tablebulletend
+ */
+ uint32_t clk_freq_in_hz;
+
+ /* Use to specific divider for two clocks if needed.
+ * Set to Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO for no divider
+ * relation clocks
+ * @values
+ * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO
+ * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND
+ * - #Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR @tablebulletend
+ */
+ uint16_t clk_attri;
+
+ /*
+ * Specifies the root clock source.
+ * Currently, only Q6AFE_LPASS_CLK_ROOT_DEFAULT is valid
+ * @values
+ * - 0 @tablebulletend
+ */
+ uint16_t clk_root;
+
+ /*
+ * for enable and disable clock.
+ * "clk_freq_in_hz", "clk_attri", and "clk_root"
+ * are ignored in disable clock case.
+ * @values
+ * - 0 -- Disabled
+ * - 1 -- Enabled @tablebulletend
+ */
+ uint32_t enable;
+};
+
+struct afe_clk_cfg {
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+ u32 i2s_cfg_minor_version;
+
+/* clk value 1 in MHz. */
+ u32 clk_val1;
+
+/* clk value 2 in MHz. */
+ u32 clk_val2;
+
+/* clk_src
+ * #Q6AFE_LPASS_CLK_SRC_EXTERNAL
+ * #Q6AFE_LPASS_CLK_SRC_INTERNAL
+ */
+
+ u16 clk_src;
+
+/* clk_root -0 for default */
+ u16 clk_root;
+
+/* clk_set_mode
+ * #Q6AFE_LPASS_MODE_BOTH_INVALID
+ * #Q6AFE_LPASS_MODE_CLK1_VALID
+ * #Q6AFE_LPASS_MODE_CLK2_VALID
+ * #Q6AFE_LPASS_MODE_BOTH_VALID
+ */
+ u16 clk_set_mode;
+
+/* This param id is used to configure I2S clk */
+ u16 reserved;
+} __packed;
+
+/* This param id is used to configure I2S clk */
+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
+#define AFE_MODULE_CLOCK_SET 0x0001028F
+#define AFE_PARAM_ID_CLOCK_SET 0x00010290
+
+struct afe_lpass_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_clk_cfg clk_cfg;
+} __packed;
+
+enum afe_lpass_digital_clk_src {
+ Q6AFE_LPASS_DIGITAL_ROOT_INVALID,
+ Q6AFE_LPASS_DIGITAL_ROOT_PRI_MI2S_OSR,
+ Q6AFE_LPASS_DIGITAL_ROOT_SEC_MI2S_OSR,
+ Q6AFE_LPASS_DIGITAL_ROOT_TER_MI2S_OSR,
+ Q6AFE_LPASS_DIGITAL_ROOT_QUAD_MI2S_OSR,
+ Q6AFE_LPASS_DIGITAL_ROOT_CDC_ROOT_CLK,
+} __packed;
+
+/* This param id is used to configure internal clk */
+#define AFE_PARAM_ID_INTERNAL_DIGIATL_CDC_CLK_CONFIG 0x00010239
+
+struct afe_digital_clk_cfg {
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+ u32 i2s_cfg_minor_version;
+
+/* clk value in MHz. */
+ u32 clk_val;
+
+/* INVALID
+ * PRI_MI2S_OSR
+ * SEC_MI2S_OSR
+ * TER_MI2S_OSR
+ * QUAD_MI2S_OSR
+ * DIGT_CDC_ROOT
+ */
+ u16 clk_root;
+
+/* This field must be set to zero. */
+ u16 reserved;
+} __packed;
+
+
+struct afe_lpass_digital_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_digital_clk_cfg clk_cfg;
+} __packed;
+
+/*
+ * Opcode for AFE to start DTMF.
+ */
+#define AFE_PORTS_CMD_DTMF_CTL 0x00010102
+
+/** DTMF payload.*/
+struct afe_dtmf_generation_command {
+ struct apr_hdr hdr;
+
+ /*
+ * Duration of the DTMF tone in ms.
+ * -1 -> continuous,
+ * 0 -> disable
+ */
+ int64_t duration_in_ms;
+
+ /*
+ * The DTMF high tone frequency.
+ */
+ uint16_t high_freq;
+
+ /*
+ * The DTMF low tone frequency.
+ */
+ uint16_t low_freq;
+
+ /*
+ * The DTMF volume setting
+ */
+ uint16_t gain;
+
+ /*
+ * The number of ports to enable/disable on.
+ */
+ uint16_t num_ports;
+
+ /*
+ * The Destination ports - array .
+ * For DTMF on multiple ports, portIds needs to
+ * be populated numPorts times.
+ */
+ uint16_t port_ids;
+
+ /*
+ * variable for 32 bit alignment of APR packet.
+ */
+ uint16_t reserved;
+} __packed;
+
+enum afe_config_type {
+ AFE_SLIMBUS_SLAVE_PORT_CONFIG,
+ AFE_SLIMBUS_SLAVE_CONFIG,
+ AFE_CDC_REGISTERS_CONFIG,
+ AFE_AANC_VERSION,
+ AFE_CDC_CLIP_REGISTERS_CONFIG,
+ AFE_CLIP_BANK_SEL,
+ AFE_CDC_REGISTER_PAGE_CONFIG,
+ AFE_MAX_CONFIG_TYPES,
+};
+
+struct afe_param_slimbus_slave_port_cfg {
+ uint32_t minor_version;
+ uint16_t slimbus_dev_id;
+ uint16_t slave_dev_pgd_la;
+ uint16_t slave_dev_intfdev_la;
+ uint16_t bit_width;
+ uint16_t data_format;
+ uint16_t num_channels;
+ uint16_t slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+} __packed;
+
+struct afe_param_cdc_slimbus_slave_cfg {
+ uint32_t minor_version;
+ uint32_t device_enum_addr_lsw;
+ uint32_t device_enum_addr_msw;
+ uint16_t tx_slave_port_offset;
+ uint16_t rx_slave_port_offset;
+} __packed;
+
+struct afe_param_cdc_reg_cfg {
+ uint32_t minor_version;
+ uint32_t reg_logical_addr;
+ uint32_t reg_field_type;
+ uint32_t reg_field_bit_mask;
+ uint16_t reg_bit_width;
+ uint16_t reg_offset_scale;
+} __packed;
+
+#define AFE_API_VERSION_CDC_REG_PAGE_CFG 1
+
+enum {
+ AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_0 = 0,
+ AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_1,
+ AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_2,
+ AFE_CDC_REG_PAGE_ASSIGN_PROC_ID_3,
+};
+
+struct afe_param_cdc_reg_page_cfg {
+ uint32_t minor_version;
+ uint32_t enable;
+ uint32_t proc_id;
+} __packed;
+
+struct afe_param_cdc_reg_cfg_data {
+ uint32_t num_registers;
+ struct afe_param_cdc_reg_cfg *reg_data;
+} __packed;
+
+struct afe_svc_cmd_set_param {
+ uint32_t payload_size;
+ uint32_t payload_address_lsw;
+ uint32_t payload_address_msw;
+ uint32_t mem_map_handle;
+} __packed;
+
+struct afe_svc_param_data {
+ uint32_t module_id;
+ uint32_t param_id;
+ uint16_t param_size;
+ uint16_t reserved;
+} __packed;
+
+struct afe_param_hw_mad_ctrl {
+ uint32_t minor_version;
+ uint16_t mad_type;
+ uint16_t mad_enable;
+} __packed;
+
+struct afe_cmd_hw_mad_ctrl {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_hw_mad_ctrl payload;
+} __packed;
+
+struct afe_cmd_hw_mad_slimbus_slave_port_cfg {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_slimbus_slave_port_cfg sb_port_cfg;
+} __packed;
+
+struct afe_cmd_sw_mad_enable {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+} __packed;
+
+struct afe_param_cdc_reg_cfg_payload {
+ struct afe_svc_param_data common;
+ struct afe_param_cdc_reg_cfg reg_cfg;
+} __packed;
+
+struct afe_lpass_clk_config_command_v2 {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_svc_param_data pdata;
+ struct afe_clk_set clk_cfg;
+} __packed;
+
+/*
+ * reg_data's size can be up to AFE_MAX_CDC_REGISTERS_TO_CONFIG
+ */
+struct afe_svc_cmd_cdc_reg_cfg {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_param_cdc_reg_cfg_payload reg_data[0];
+} __packed;
+
+struct afe_svc_cmd_init_cdc_reg_cfg {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 init;
+} __packed;
+
+struct afe_svc_cmd_sb_slave_cfg {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_cdc_slimbus_slave_cfg sb_slave_cfg;
+} __packed;
+
+struct afe_svc_cmd_cdc_reg_page_cfg {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_cdc_reg_page_cfg cdc_reg_page_cfg;
+} __packed;
+
+struct afe_svc_cmd_cdc_aanc_version {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_cdc_aanc_version version;
+} __packed;
+
+struct afe_port_cmd_set_aanc_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union {
+ struct afe_param_aanc_port_cfg aanc_port_cfg;
+ struct afe_mod_enable_param mod_enable;
+ } __packed data;
+} __packed;
+
+struct afe_port_cmd_set_aanc_acdb_table {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+} __packed;
+
+/* Dolby DAP topology */
+#define DOLBY_ADM_COPP_TOPOLOGY_ID 0x0001033B
+#define DS2_ADM_COPP_TOPOLOGY_ID 0x1301033B
+
+/* RMS value from DSP */
+#define RMS_MODULEID_APPI_PASSTHRU 0x10009011
+#define RMS_PARAM_FIRST_SAMPLE 0x10009012
+#define RMS_PAYLOAD_LEN 4
+
+/* Customized mixing in matix mixer */
+#define MTMX_MODULE_ID_DEFAULT_CHMIXER 0x00010341
+#define DEFAULT_CHMIXER_PARAM_ID_COEFF 0x00010342
+#define CUSTOM_STEREO_PAYLOAD_SIZE 9
+#define CUSTOM_STEREO_CMD_PARAM_SIZE 24
+#define CUSTOM_STEREO_NUM_OUT_CH 0x0002
+#define CUSTOM_STEREO_NUM_IN_CH 0x0002
+#define CUSTOM_STEREO_INDEX_PARAM 0x0002
+#define Q14_GAIN_ZERO_POINT_FIVE 0x2000
+#define Q14_GAIN_UNITY 0x4000
+
+struct afe_svc_cmd_set_clip_bank_selection {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_clip_bank_sel bank_sel;
+} __packed;
+
+/* Ultrasound supported formats */
+#define US_POINT_EPOS_FORMAT_V2 0x0001272D
+#define US_RAW_FORMAT_V2 0x0001272C
+#define US_PROX_FORMAT_V4 0x0001273B
+#define US_RAW_SYNC_FORMAT 0x0001272F
+#define US_GES_SYNC_FORMAT 0x00012730
+
+#define AFE_MODULE_GROUP_DEVICE 0x00010254
+#define AFE_PARAM_ID_GROUP_DEVICE_CFG 0x00010255
+#define AFE_PARAM_ID_GROUP_DEVICE_ENABLE 0x00010256
+#define AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX 0x1102
+
+/* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_CFG
+ * parameter, which configures max of 8 AFE ports
+ * into a group.
+ * The fixed size of this structure is sixteen bytes.
+ */
+struct afe_group_device_group_cfg {
+ u32 minor_version;
+ u16 group_id;
+ u16 num_channels;
+ u16 port_id[8];
+} __packed;
+
+#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX \
+ (AFE_PORT_ID_PRIMARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX \
+ (AFE_PORT_ID_PRIMARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX \
+ (AFE_PORT_ID_SECONDARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX \
+ (AFE_PORT_ID_SECONDARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX \
+ (AFE_PORT_ID_TERTIARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX \
+ (AFE_PORT_ID_TERTIARY_TDM_TX + 0x100)
+#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX \
+ (AFE_PORT_ID_QUATERNARY_TDM_RX + 0x100)
+#define AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX \
+ (AFE_PORT_ID_QUATERNARY_TDM_TX + 0x100)
+
+/* ID of the parameter used by #AFE_MODULE_GROUP_DEVICE to configure the
+ * group device. #AFE_SVC_CMD_SET_PARAM can use this parameter ID.
+ *
+ * Requirements:
+ * - Configure the group before the member ports in the group are
+ * configured and started.
+ * - Enable the group only after it is configured.
+ * - Stop all member ports in the group before disabling the group.
+ */
+#define AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG 0x0001029E
+
+/* Version information used to handle future additions to
+ * AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG processing (for backward compatibility).
+ */
+#define AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG 0x1
+
+/* Number of AFE ports in group device */
+#define AFE_GROUP_DEVICE_NUM_PORTS 8
+
+/* Payload of the AFE_PARAM_ID_GROUP_DEVICE_TDM_CONFIG parameter ID
+ * used by AFE_MODULE_GROUP_DEVICE.
+ */
+struct afe_param_id_group_device_tdm_cfg {
+ u32 group_device_cfg_minor_version;
+ /* Minor version used to track group device configuration.
+ * @values #AFE_API_VERSION_GROUP_DEVICE_TDM_CONFIG
+ */
+
+ u16 group_id;
+ /* ID for the group device.
+ * @values
+ * - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_RX
+ * - #AFE_GROUP_DEVICE_ID_PRIMARY_TDM_TX
+ * - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_RX
+ * - #AFE_GROUP_DEVICE_ID_SECONDARY_TDM_TX
+ * - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_RX
+ * - #AFE_GROUP_DEVICE_ID_TERTIARY_TDM_TX
+ * - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_RX
+ * - #AFE_GROUP_DEVICE_ID_QUATERNARY_TDM_TX
+ */
+
+ u16 reserved;
+ /* 0 */
+
+ u16 port_id[AFE_GROUP_DEVICE_NUM_PORTS];
+ /* Array of member port IDs of this group.
+ * @values
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_1
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_2
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_3
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_4
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_5
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_6
+ * - #AFE_PORT_ID_PRIMARY_TDM_RX_7
+
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_1
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_2
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_3
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_4
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_5
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_6
+ * - #AFE_PORT_ID_PRIMARY_TDM_TX_7
+
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_1
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_2
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_3
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_4
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_5
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_6
+ * - #AFE_PORT_ID_SECONDARY_TDM_RX_7
+
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_1
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_2
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_3
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_4
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_5
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_6
+ * - #AFE_PORT_ID_SECONDARY_TDM_TX_7
+
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_1
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_2
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_3
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_4
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_5
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_6
+ * - #AFE_PORT_ID_TERTIARY_TDM_RX_7
+
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_1
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_2
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_3
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_4
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_5
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_6
+ * - #AFE_PORT_ID_TERTIARY_TDM_TX_7
+
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_1
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_2
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_3
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_4
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_5
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_6
+ * - #AFE_PORT_ID_QUATERNARY_TDM_RX_7
+
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_1
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_2
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_3
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_4
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_5
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_6
+ * - #AFE_PORT_ID_QUATERNARY_TDM_TX_7
+ * @tablebulletend
+ */
+
+ u32 num_channels;
+ /* Number of enabled slots for TDM frame.
+ * @values 1 to 8
+ */
+
+ u32 sample_rate;
+ /* Sampling rate of the port.
+ * @values
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_24K
+ * - #AFE_PORT_SAMPLE_RATE_32K
+ * - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend
+ */
+
+ u32 bit_width;
+ /* Bit width of the sample.
+ * @values 16, 24, (32)
+ */
+
+ u16 nslots_per_frame;
+ /* Number of slots per frame. Typical : 1, 2, 4, 8, 16, 32.
+ * @values 1 - 32
+ */
+
+ u16 slot_width;
+ /* Slot width of the slot in a TDM frame. (slot_width >= bit_width)
+ * have to be satisfied.
+ * @values 16, 24, 32
+ */
+
+ u32 slot_mask;
+ /* Position of active slots. When that bit is set, that paricular
+ * slot is active.
+ * Number of active slots can be inferred by number of bits set in
+ * the mask. Only 8 individual bits can be enabled.
+ * Bits 0..31 corresponding to slot 0..31
+ * @values 1 to 2^32 -1
+ */
+} __packed;
+
+/* Payload of the #AFE_PARAM_ID_GROUP_DEVICE_ENABLE
+ * parameter, which enables or
+ * disables any module.
+ * The fixed size of this structure is four bytes.
+ */
+
+struct afe_group_device_enable {
+ u16 group_id;
+ /* valid value is AFE_GROUP_DEVICE_ID_SECONDARY_MI2S_RX */
+ u16 enable;
+ /* Enables (1) or disables (0) the module. */
+} __packed;
+
+union afe_port_group_config {
+ struct afe_group_device_group_cfg group_cfg;
+ struct afe_group_device_enable group_enable;
+ struct afe_param_id_group_device_tdm_cfg tdm_cfg;
+} __packed;
+
+struct afe_port_group_create {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ union afe_port_group_config data;
+} __packed;
+
+/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to specify
+ * the timing statistics of the corresponding device interface.
+ * Client can periodically query for the device time statistics to help adjust
+ * the PLL based on the drift value. The get param command must be sent to
+ * AFE port ID corresponding to device interface
+
+ * This parameter ID supports following get param commands:
+ * #AFE_PORT_CMD_GET_PARAM_V2 and
+ * #AFE_PORT_CMD_GET_PARAM_V3.
+ */
+#define AFE_PARAM_ID_DEV_TIMING_STATS 0x000102AD
+
+/* Version information used to handle future additions to AFE device
+ * interface timing statistics (for backward compatibility).
+ */
+#define AFE_API_VERSION_DEV_TIMING_STATS 0x1
+
+/* Enumeration for specifying a sink(Rx) device */
+#define AFE_SINK_DEVICE 0x0
+
+/* Enumeration for specifying a source(Tx) device */
+#define AFE_SOURCE_DEVICE 0x1
+
+/* Enumeration for specifying the drift reference is of type AV Timer */
+#define AFE_REF_TIMER_TYPE_AVTIMER 0x0
+
+/* Message payload structure for the
+ * AFE_PARAM_ID_DEV_TIMING_STATS parameter.
+ */
+struct afe_param_id_dev_timing_stats {
+ /* Minor version used to track the version of device interface timing
+ * statistics. Currently, the supported version is 1.
+ * @values #AFE_API_VERSION_DEV_TIMING_STATS
+ */
+ u32 minor_version;
+
+ /* Indicates the device interface direction as either
+ * source (Tx) or sink (Rx).
+ * @values
+ * #AFE_SINK_DEVICE
+ * #AFE_SOURCE_DEVICE
+ */
+ u16 device_direction;
+
+ /* Reference timer for drift accumulation and time stamp information.
+ * @values
+ * #AFE_REF_TIMER_TYPE_AVTIMER @tablebulletend
+ */
+ u16 reference_timer;
+
+ /*
+ * Flag to indicate if resync is required on the client side for
+ * drift correction. Flag is set to TRUE for the first get_param
+ * response after device interface starts. This flag value can be
+ * used by client to identify if device interface restart has
+ * happened and if any re-sync is required at their end for drift
+ * correction.
+ * @values
+ * 0: FALSE (Resync not required)
+ * 1: TRUE (Resync required) @tablebulletend
+ */
+ u32 resync_flag;
+
+ /* Accumulated drift value in microseconds. This value is updated
+ * every 100th ms.
+ * Positive drift value indicates AV timer is running faster than device
+ * Negative drift value indicates AV timer is running slower than device
+ * @values Any valid int32 number
+ */
+ s32 acc_drift_value;
+
+ /* Lower 32 bits of the 64-bit absolute timestamp of reference
+ * timer in microseconds.
+
+ * This timestamp corresponds to the time when the drift values
+ * are accumlated for every 100th ms.
+ * @values Any valid uint32 number
+ */
+ u32 ref_timer_abs_ts_lsw;
+
+ /* Upper 32 bits of the 64-bit absolute timestamp of reference
+ * timer in microseconds.
+ * This timestamp corresponds to the time when the drift values
+ * are accumlated for every 100th ms.
+ * @values Any valid uint32 number
+ */
+ u32 ref_timer_abs_ts_msw;
+} __packed;
+
+struct afe_av_dev_drift_get_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_get_param_v2 get_param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_dev_timing_stats timing_stats;
+} __packed;
+
+struct afe_av_dev_drift_get_param_resp {
+ uint32_t status;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_param_id_dev_timing_stats timing_stats;
+} __packed;
+
+/* Command for Matrix or Stream Router */
+#define ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2 0x00010DCE
+/* Module for AVSYNC */
+#define ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC 0x00010DC6
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
+ * render window start value. This parameter is supported only for a Set
+ * command (not a Get command) in the Rx direction
+ * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2).
+ * Render window start is a value (session time minus timestamp, or ST-TS)
+ * below which frames are held, and after which frames are immediately
+ * rendered.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2 0x00010DD1
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC to specify the
+ * render window end value. This parameter is supported only for a Set
+ * command (not a Get command) in the Rx direction
+ * (#ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2). Render window end is a value
+ * (session time minus timestamp) above which frames are dropped, and below
+ * which frames are immediately rendered.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2 0x00010DD2
+
+/* Generic payload of the window parameters in the
+ * #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC module.
+ * This payload is supported only for a Set command
+ * (not a Get command) on the Rx path.
+ */
+struct asm_session_mtmx_strtr_param_window_v2_t {
+ u32 window_lsw;
+ /* Lower 32 bits of the render window start value. */
+
+ u32 window_msw;
+ /* Upper 32 bits of the render window start value.
+ *
+ * The 64-bit number formed by window_lsw and window_msw specifies a
+ * signed 64-bit window value in microseconds. The sign extension is
+ * necessary. This value is used by the following parameter IDs:
+ * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2
+ * #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2
+ * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_START_V2
+ * #ASM_SESSION_MTMX_STRTR_PARAM_STAT_WINDOW_END_V2
+ * The value depends on which parameter ID is used.
+ * The aDSP honors the windows at a granularity of 1 ms.
+ */
+};
+
+struct asm_session_cmd_set_mtmx_strstr_params_v2 {
+ uint32_t data_payload_addr_lsw;
+ /* Lower 32 bits of the 64-bit data payload address. */
+
+ uint32_t data_payload_addr_msw;
+ /* Upper 32 bits of the 64-bit data payload address.
+ * If the address is not sent (NULL), the message is in the payload.
+ * If the address is sent (non-NULL), the parameter data payloads
+ * begin at the specified address.
+ */
+
+ uint32_t mem_map_handle;
+ /* Unique identifier for an address. This memory map handle is returned
+ * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+ * values
+ * - NULL -- Parameter data payloads are within the message payload
+ * (in-band).
+ * - Non-NULL -- Parameter data payloads begin at the address specified
+ * in the data_payload_addr_lsw and data_payload_addr_msw fields
+ * (out-of-band).
+ */
+
+ uint32_t data_payload_size;
+ /* Actual size of the variable payload accompanying the message, or in
+ * shared memory. This field is used for parsing the parameter payload.
+ * values > 0 bytes
+ */
+
+ uint32_t direction;
+ /* Direction of the entity (matrix mixer or stream router) on which
+ * the parameter is to be set.
+ * values
+ * - 0 -- Rx (for Rx stream router or Rx matrix mixer)
+ * - 1 -- Tx (for Tx stream router or Tx matrix mixer)
+ */
+};
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
+ * audio client choose the rendering decision that the audio DSP should use.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD 0x00012F0D
+
+/* Indicates that rendering decision will be based on default rate
+ * (session clock based rendering, device driven).
+ * 1. The default session clock based rendering is inherently driven
+ * by the timing of the device.
+ * 2. After the initial decision is made (first buffer after a run
+ * command), subsequent data rendering decisions are made with
+ * respect to the rate at which the device is rendering, thus deriving
+ * its timing from the device.
+ * 3. While this decision making is simple, it has some inherent limitations
+ * (mentioned in the next section).
+ * 4. If this API is not set, the session clock based rendering will be assumed
+ * and this will ensure that the DSP is backward compatible.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT 0
+
+/* Indicates that rendering decision will be based on local clock rate.
+ * 1. In the DSP loopback/client loopback use cases (frame based
+ * inputs), the incoming data into audio DSP is time-stamped at the
+ * local clock rate (STC).
+ * 2. This TS rate may match the incoming data rate or maybe different
+ * from the incoming data rate.
+ * 3. Regardless, the data will be time-stamped with local STC and
+ * therefore, the client is recommended to set this mode for these
+ * use cases. This method is inherently more robust to sequencing
+ * (AFE Start/Stop) and device switches, among other benefits.
+ * 4. This API will inform the DSP to compare every incoming buffer TS
+ * against local STC.
+ * 5. DSP will continue to honor render windows APIs, as before.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC 1
+
+/* Structure for rendering decision parameter */
+struct asm_session_mtmx_strtr_param_render_mode_t {
+ /* Specifies the type of rendering decision the audio DSP should use.
+ *
+ * @values
+ * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT
+ * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC
+ */
+ u32 flags;
+} __packed;
+
+/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the
+ * audio client to specify the clock recovery mechanism that the audio DSP
+ * should use.
+ */
+
+#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD 0x00012F0E
+
+/* Indicates that default clock recovery will be used (no clock recovery).
+ * If the client wishes that no clock recovery be done, the client can
+ * choose this. This means that no attempt will made by the DSP to try and
+ * match the rates of the input and output audio.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE 0
+
+/* Indicates that independent clock recovery needs to be used.
+ * 1. In the DSP loopback/client loopback use cases (frame based inputs),
+ * the client should choose the independent clock recovery option.
+ * 2. This basically de-couples the audio and video from knowing each others
+ * clock sources and lets the audio DSP independently rate match the input
+ * and output rates.
+ * 3. After drift detection, the drift correction is achieved by either pulling
+ * the PLLs (if applicable) or by stream to device rate matching
+ * (for PCM use cases) by comparing drift with respect to STC.
+ * 4. For passthrough use cases, since the PLL pulling is the only option,
+ * a best effort will be made.
+ * If PLL pulling is not possible / available, the rendering will be
+ * done without rate matching.
+ */
+#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO 1
+
+/* Payload of the #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC parameter.
+ */
+struct asm_session_mtmx_strtr_param_clk_rec_t {
+ /* Specifies the type of clock recovery that the audio DSP should
+ * use for rate matching.
+ */
+
+ /* @values
+ * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_DEFAULT
+ * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_INDEPENDENT
+ */
+ u32 flags;
+} __packed;
+
+
+/* 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 asm_stream_param_data_v2 data;
+ union asm_session_mtmx_strtr_param_config config;
+} __packed;
+
+#define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF
+#define ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2 0x00010DD0
+
+#define ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3 0x00012F0B
+#define ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK (0x80000000UL)
+
+struct asm_session_cmd_get_mtmx_strstr_params_v2 {
+ uint32_t data_payload_addr_lsw;
+ /* Lower 32 bits of the 64-bit data payload address. */
+
+ uint32_t data_payload_addr_msw;
+ /*
+ * Upper 32 bits of the 64-bit data payload address.
+ * If the address is not sent (NULL), the message is in the payload.
+ * If the address is sent (non-NULL), the parameter data payloads
+ * begin at the specified address.
+ */
+
+ uint32_t mem_map_handle;
+ /*
+ * Unique identifier for an address. This memory map handle is returned
+ * by the aDSP through the #ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+ * values
+ * - NULL -- Parameter data payloads are within the message payload
+ * (in-band).
+ * - Non-NULL -- Parameter data payloads begin at the address specified
+ * in the data_payload_addr_lsw and data_payload_addr_msw fields
+ * (out-of-band).
+ */
+ uint32_t direction;
+ /*
+ * Direction of the entity (matrix mixer or stream router) on which
+ * the parameter is to be set.
+ * values
+ * - 0 -- Rx (for Rx stream router or Rx matrix mixer)
+ * - 1 -- Tx (for Tx stream router or Tx matrix mixer)
+ */
+ uint32_t module_id;
+ /* Unique module ID. */
+
+ uint32_t param_id;
+ /* Unique parameter ID. */
+
+ uint32_t param_max_size;
+};
+
+struct asm_session_mtmx_strtr_param_session_time_v3_t {
+ uint32_t session_time_lsw;
+ /* Lower 32 bits of the current session time in microseconds */
+
+ uint32_t session_time_msw;
+ /*
+ * Upper 32 bits of the current session time in microseconds.
+ * The 64-bit number formed by session_time_lsw and session_time_msw
+ * is treated as signed.
+ */
+
+ uint32_t absolute_time_lsw;
+ /*
+ * Lower 32 bits of the 64-bit absolute time in microseconds.
+ * This is the time when the sample corresponding to the
+ * session_time_lsw is rendered to the hardware. This absolute
+ * time can be slightly in the future or past.
+ */
+
+ uint32_t absolute_time_msw;
+ /*
+ * Upper 32 bits of the 64-bit absolute time in microseconds.
+ * This is the time when the sample corresponding to the
+ * session_time_msw is rendered to hardware. This absolute
+ * time can be slightly in the future or past. The 64-bit number
+ * formed by absolute_time_lsw and absolute_time_msw is treated as
+ * unsigned.
+ */
+
+ uint32_t time_stamp_lsw;
+ /* Lower 32 bits of the last processed timestamp in microseconds */
+
+ uint32_t time_stamp_msw;
+ /*
+ * Upper 32 bits of the last processed timestamp in microseconds.
+ * The 64-bit number formed by time_stamp_lsw and time_stamp_lsw
+ * is treated as unsigned.
+ */
+
+ uint32_t flags;
+ /*
+ * Keeps track of any additional flags needed.
+ * @values{for bit 31}
+ * - 0 -- Uninitialized/invalid
+ * - 1 -- Valid
+ * All other bits are reserved; clients must set them to zero.
+ */
+};
+
+union asm_session_mtmx_strtr_data_type {
+ struct asm_session_mtmx_strtr_param_session_time_v3_t session_time;
+};
+
+struct asm_mtmx_strtr_get_params {
+ struct apr_hdr hdr;
+ struct asm_session_cmd_get_mtmx_strstr_params_v2 param_info;
+} __packed;
+
+struct asm_mtmx_strtr_get_params_cmdrsp {
+ uint32_t err_code;
+ struct asm_stream_param_data_v2 param_info;
+ union asm_session_mtmx_strtr_data_type param_data;
+} __packed;
+
+#define AUDPROC_MODULE_ID_RESAMPLER 0x00010719
+
+enum {
+ LEGACY_PCM = 0,
+ COMPRESSED_PASSTHROUGH,
+ COMPRESSED_PASSTHROUGH_CONVERT,
+ COMPRESSED_PASSTHROUGH_DSD,
+ LISTEN,
+ COMPRESSED_PASSTHROUGH_GEN,
+ COMPRESSED_PASSTHROUGH_IEC61937
+};
+
+#define AUDPROC_MODULE_ID_COMPRESSED_MUTE 0x00010770
+#define AUDPROC_PARAM_ID_COMPRESSED_MUTE 0x00010771
+
+struct adm_set_compressed_device_mute {
+ struct adm_cmd_set_pp_params_v5 command;
+ struct adm_param_data_v5 params;
+ u32 mute_on;
+} __packed;
+
+#define AUDPROC_MODULE_ID_COMPRESSED_LATENCY 0x0001076E
+#define AUDPROC_PARAM_ID_COMPRESSED_LATENCY 0x0001076F
+
+struct adm_set_compressed_device_latency {
+ struct adm_cmd_set_pp_params_v5 command;
+ struct adm_param_data_v5 params;
+ u32 latency;
+} __packed;
+
+#define VOICEPROC_MODULE_ID_GENERIC_TX 0x00010EF6
+#define VOICEPROC_PARAM_ID_FLUENCE_SOUNDFOCUS 0x00010E37
+#define VOICEPROC_PARAM_ID_FLUENCE_SOURCETRACKING 0x00010E38
+#define MAX_SECTORS 8
+#define MAX_NOISE_SOURCE_INDICATORS 3
+#define MAX_POLAR_ACTIVITY_INDICATORS 360
+
+struct sound_focus_param {
+ uint16_t start_angle[MAX_SECTORS];
+ uint8_t enable[MAX_SECTORS];
+ uint16_t gain_step;
+} __packed;
+
+struct source_tracking_param {
+ uint8_t vad[MAX_SECTORS];
+ uint16_t doa_speech;
+ uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
+ uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
+} __packed;
+
+struct adm_param_fluence_soundfocus_t {
+ uint16_t start_angles[MAX_SECTORS];
+ uint8_t enables[MAX_SECTORS];
+ uint16_t gain_step;
+ uint16_t reserved;
+} __packed;
+
+struct adm_set_fluence_soundfocus_param {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+ struct adm_param_fluence_soundfocus_t soundfocus_data;
+} __packed;
+
+struct adm_param_fluence_sourcetracking_t {
+ uint8_t vad[MAX_SECTORS];
+ uint16_t doa_speech;
+ uint16_t doa_noise[MAX_NOISE_SOURCE_INDICATORS];
+ uint8_t polar_activity[MAX_POLAR_ACTIVITY_INDICATORS];
+} __packed;
+
+#define AUDPROC_MODULE_ID_AUDIOSPHERE 0x00010916
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_ENABLE 0x00010917
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_STRENGTH 0x00010918
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_CONFIG_MODE 0x00010919
+
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_STEREO_INPUT 0x0001091A
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_COEFFS_MULTICHANNEL_INPUT 0x0001091B
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_STEREO_INPUT 0x0001091C
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_DESIGN_MULTICHANNEL_INPUT 0x0001091D
+
+#define AUDPROC_PARAM_ID_AUDIOSPHERE_OPERATING_INPUT_MEDIA_INFO 0x0001091E
+
+#define AUDPROC_MODULE_ID_VOICE_TX_SECNS 0x10027059
+#define AUDPROC_PARAM_IDX_SEC_PRIMARY_MIC_CH 0x10014444
+
+struct admx_sec_primary_mic_ch {
+ uint16_t version;
+ uint16_t reserved;
+ uint16_t sec_primary_mic_ch;
+ uint16_t reserved1;
+} __packed;
+
+
+struct adm_set_sec_primary_ch_params {
+ struct adm_cmd_set_pp_params_v5 params;
+ struct adm_param_data_v5 data;
+ struct admx_sec_primary_mic_ch sec_primary_mic_ch_data;
+} __packed;
+#endif /*_APR_AUDIO_V2_H_ */