Re: [PATCH v6 1/7] ASoC: qcom: qdsp6: add topology-driven Audio IF support

On 8/3/26 8:56 PM, Prasad Kumpatla wrote:
> Add topology parsing and media-format programming for Audio IF source and
> sink modules.
> 
> Add the Audio IF module IDs, the required topology tokens, and a dedicated
> topology loader that stores the parsed interface configuration in the
> AudioReach module state. Also add the Audio IF media-format path that
> sends the interface configuration, hardware endpoint media format, and
> frame-duration parameters for Audio IF modules.
> 
> This keeps the serial-interface configuration topology-driven while still
> allowing the machine driver to provide runtime slot and media format
> settings. The same Audio IF path can then be reused for TDM, PCM, and I2S
> style backends.
> 
> The new UAPI tokens (AR_TKN_U16_MODULE_SYNC_SRC=262 through
> AR_TKN_U8_MODULE_INV_EXT_BIT_CLK=276) are added, together with the value
> defines used by the sync source, sync mode, data delay, interface mode,
> bit clock type, and polarity tokens.
> 
> MODULE_ID_AUDIO_IF_SINK (0x0700117C) and MODULE_ID_AUDIO_IF_SOURCE
> (0x0700117D) are introduced in this patch.
> 
> This Module is validated on Hawi and Shikra platforms.
> 
> Signed-off-by: Prasad Kumpatla <[email protected]>

Reviewed-by: Srinivas Kandagatla <[email protected]>

--srini


> ---
>  include/uapi/sound/snd_ar_tokens.h |  93 ++++++++++++++++++++++++++
>  sound/soc/qcom/qdsp6/audioreach.c  |  93 ++++++++++++++++++++++++++
>  sound/soc/qcom/qdsp6/audioreach.h  |  90 +++++++++++++++++++++++++
>  sound/soc/qcom/qdsp6/topology.c    | 103 +++++++++++++++++++++++++++++
>  4 files changed, 379 insertions(+)
> 
> diff --git a/include/uapi/sound/snd_ar_tokens.h b/include/uapi/sound/snd_ar_tokens.h
> index 6b8102eaa121..1700e3f5cb64 100644
> --- a/include/uapi/sound/snd_ar_tokens.h
> +++ b/include/uapi/sound/snd_ar_tokens.h
> @@ -168,6 +168,60 @@ enum ar_event_types {
>   *						LOG_WAIT = 0,
>   *						LOG_IMMEDIATELY = 1
>   *
> + * %AR_TKN_U16_MODULE_SYNC_SRC:			Frame sync source
> + *						AR_AUDIO_IF_SYNC_SRC_EXTERNAL = 0,
> + *						AR_AUDIO_IF_SYNC_SRC_INTERNAL = 1
> + *
> + * %AR_TKN_U16_MODULE_CTRL_DATA_OUT_ENABLE:	Enable data-out tri-state control
> + *						AR_AUDIO_IF_CTRL_DATA_OE_DISABLE = 0,
> + *						AR_AUDIO_IF_CTRL_DATA_OE_ENABLE = 1
> + *
> + * %AR_TKN_U32_MODULE_SLOT_MASK:			Active TDM slot bitmask
> + *
> + * %AR_TKN_U16_MODULE_NSLOTS_PER_FRAME:		Number of slots per TDM frame
> + *
> + * %AR_TKN_U16_MODULE_SLOT_WIDTH:		Slot width in bits (16 or 32)
> + *
> + * %AR_TKN_U16_MODULE_SYNC_MODE:			Frame sync mode
> + *						AR_AUDIO_IF_FRAME_SYNC_MODE_SHORT = 0,
> + *						AR_AUDIO_IF_FRAME_SYNC_MODE_ONE_SLOT = 1,
> + *						AR_AUDIO_IF_FRAME_SYNC_MODE_LONG = 2
> + *
> + * %AR_TKN_U16_MODULE_CTRL_INVERT_SYNC_PULSE:	Invert frame sync pulse polarity
> + *						AR_AUDIO_IF_SYNC_NORMAL = 0,
> + *						AR_AUDIO_IF_SYNC_INVERTED = 1
> + *
> + * %AR_TKN_U16_MODULE_CTRL_SYNC_DATA_DELAY:	Data delay relative to frame sync
> + *						AR_AUDIO_IF_DATA_DELAY_NONE = 0,
> + *						AR_AUDIO_IF_DATA_DELAY_1_CYCLE = 1,
> + *						AR_AUDIO_IF_DATA_DELAY_2_CYCLE = 2
> + *
> + * %AR_TKN_U16_MODULE_INTF_MODE:			Audio IF interface mode
> + *						AR_AUDIO_IF_INTF_MODE_TDM = 0,
> + *						AR_AUDIO_IF_INTF_MODE_PCM = 1,
> + *						AR_AUDIO_IF_INTF_MODE_I2S = 2
> + *
> + * %AR_TKN_U16_MODULE_QAIF_TYPE:			QAIF hardware port type index
> + *						AR_AUDIO_IF_QAIF = 0,
> + *						AR_AUDIO_IF_QAIF_VA = 1
> + *
> + * %AR_TKN_U32_MODULE_ACTIVE_LANE_MASK:		Active lane bitmask for multi-lane
> + *
> + * %AR_TKN_U32_MODULE_FRAME_SYNC_RATE:		Frame sync rate in Hz
> + *
> + * %AR_TKN_U16_MODULE_BIT_CLK_TYPE:		Bit clock type
> + *						AR_AUDIO_IF_BIT_CLK_INTERNAL = 0,
> + *						AR_AUDIO_IF_BIT_CLK_EXTERNAL = 1,
> + *						AR_AUDIO_IF_BIT_CLK_SKIP = 2
> + *
> + * %AR_TKN_U8_MODULE_INV_INT_BIT_CLK:		Invert internal bit clock
> + *						AR_AUDIO_IF_CLK_NORMAL = 0,
> + *						AR_AUDIO_IF_CLK_INVERTED = 1
> + *
> + * %AR_TKN_U8_MODULE_INV_EXT_BIT_CLK:		Invert external bit clock
> + *						AR_AUDIO_IF_CLK_NORMAL = 0,
> + *						AR_AUDIO_IF_CLK_INVERTED = 1
> + *
>   * %AR_TKN_DAI_INDEX:				dai index
>   *
>   */
> @@ -240,6 +294,45 @@ enum ar_event_types {
>  #define AR_TKN_U32_MODULE_LOG_TAP_POINT_ID	260
>  #define AR_TKN_U32_MODULE_LOG_MODE		261
>  
> +#define AR_TKN_U16_MODULE_SYNC_SRC		262
> +#define AR_TKN_U16_MODULE_CTRL_DATA_OUT_ENABLE	263
> +#define AR_TKN_U32_MODULE_SLOT_MASK		264
> +#define AR_TKN_U16_MODULE_NSLOTS_PER_FRAME	265
> +#define AR_TKN_U16_MODULE_SLOT_WIDTH		266
> +#define AR_TKN_U16_MODULE_SYNC_MODE		267
> +#define AR_TKN_U16_MODULE_CTRL_INVERT_SYNC_PULSE	268
> +#define AR_TKN_U16_MODULE_CTRL_SYNC_DATA_DELAY	269
> +#define AR_TKN_U16_MODULE_INTF_MODE		270
> +#define AR_TKN_U16_MODULE_QAIF_TYPE		271
> +#define AR_TKN_U32_MODULE_ACTIVE_LANE_MASK	272
> +#define AR_TKN_U32_MODULE_FRAME_SYNC_RATE	273
> +#define AR_TKN_U16_MODULE_BIT_CLK_TYPE		274
> +#define AR_TKN_U8_MODULE_INV_INT_BIT_CLK	275
> +#define AR_TKN_U8_MODULE_INV_EXT_BIT_CLK	276
> +
> +#define AR_AUDIO_IF_SYNC_SRC_EXTERNAL		0
> +#define AR_AUDIO_IF_SYNC_SRC_INTERNAL		1
> +#define AR_AUDIO_IF_CTRL_DATA_OE_DISABLE	0
> +#define AR_AUDIO_IF_CTRL_DATA_OE_ENABLE		1
> +#define AR_AUDIO_IF_INTF_MODE_TDM		0
> +#define AR_AUDIO_IF_INTF_MODE_PCM		1
> +#define AR_AUDIO_IF_INTF_MODE_I2S		2
> +#define AR_AUDIO_IF_QAIF			0
> +#define AR_AUDIO_IF_QAIF_VA			1
> +#define AR_AUDIO_IF_FRAME_SYNC_MODE_SHORT	0
> +#define AR_AUDIO_IF_FRAME_SYNC_MODE_ONE_SLOT	1
> +#define AR_AUDIO_IF_FRAME_SYNC_MODE_LONG	2
> +#define AR_AUDIO_IF_SYNC_NORMAL			0
> +#define AR_AUDIO_IF_SYNC_INVERTED		1
> +#define AR_AUDIO_IF_DATA_DELAY_NONE		0
> +#define AR_AUDIO_IF_DATA_DELAY_1_CYCLE		1
> +#define AR_AUDIO_IF_DATA_DELAY_2_CYCLE		2
> +#define AR_AUDIO_IF_BIT_CLK_INTERNAL		0
> +#define AR_AUDIO_IF_BIT_CLK_EXTERNAL		1
> +#define AR_AUDIO_IF_BIT_CLK_SKIP		2
> +#define AR_AUDIO_IF_CLK_NORMAL			0
> +#define AR_AUDIO_IF_CLK_INVERTED		1
> +
>  #define SND_SOC_AR_TPLG_MODULE_CFG_TYPE 0x01001006
>  struct audioreach_module_priv_data {
>  	__le32 size;	/* size in bytes of the array, including all elements */
> diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
> index e6e9eb2e85aa..6a11f2932ba5 100644
> --- a/sound/soc/qcom/qdsp6/audioreach.c
> +++ b/sound/soc/qcom/qdsp6/audioreach.c
> @@ -152,6 +152,13 @@ struct apm_i2s_module_intf_cfg {
>  
>  #define APM_I2S_INTF_CFG_PSIZE ALIGN(sizeof(struct apm_i2s_module_intf_cfg), 8)
>  
> +struct apm_audio_if_module_intf_cfg {
> +	struct apm_module_param_data param_data;
> +	struct param_id_audio_if_intf_cfg cfg;
> +} __packed;
> +
> +#define APM_AUDIO_IF_INTF_CFG_PSIZE ALIGN(sizeof(struct apm_audio_if_module_intf_cfg), 8)
> +
>  struct apm_module_hw_ep_mf_cfg {
>  	struct apm_module_param_data param_data;
>  	struct param_id_hw_ep_mf mf;
> @@ -168,6 +175,13 @@ struct apm_module_frame_size_factor_cfg {
>  
>  #define APM_FS_CFG_PSIZE ALIGN(sizeof(struct apm_module_frame_size_factor_cfg), 8)
>  
> +struct apm_module_hw_ep_frame_duration_cfg {
> +	struct apm_module_param_data param_data;
> +	struct param_id_hw_ep_frame_duration frame_duration;
> +} __packed;
> +
> +#define APM_HW_EP_FRAME_DURATION_PSIZE ALIGN(sizeof(struct apm_module_hw_ep_frame_duration_cfg), 8)
> +
>  struct apm_module_hw_ep_power_mode_cfg {
>  	struct apm_module_param_data param_data;
>  	struct param_id_hw_ep_power_mode_cfg power_mode;
> @@ -1042,6 +1056,81 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
>  	return q6apm_send_cmd_sync(graph->apm, pkt, 0);
>  }
>  
> +static int audioreach_audio_if_set_media_format(struct q6apm_graph *graph,
> +						const struct audioreach_module *module,
> +						const struct audioreach_module_config *cfg)
> +{
> +	struct apm_module_hw_ep_frame_duration_cfg *fd_cfg;
> +	struct apm_module_param_data *param_data;
> +	struct apm_audio_if_module_intf_cfg *intf_cfg;
> +	struct apm_module_hw_ep_mf_cfg *hw_cfg;
> +	int ic_sz = APM_AUDIO_IF_INTF_CFG_PSIZE;
> +	int ep_sz = APM_HW_EP_CFG_PSIZE;
> +	int fd_sz = APM_HW_EP_FRAME_DURATION_PSIZE;
> +	int size = ic_sz + ep_sz + fd_sz;
> +	u32 slot_mask = cfg->slot_mask ? cfg->slot_mask : module->slot_mask;
> +	u16 nslots_per_frame = cfg->nslots_per_frame ?
> +				 (u16)cfg->nslots_per_frame : module->nslots_per_frame;
> +	u16 slot_width = cfg->slot_width ? (u16)cfg->slot_width : module->slot_width;
> +	void *p;
> +
> +	struct gpr_pkt *pkt __free(kfree) = audioreach_alloc_apm_cmd_pkt(size, APM_CMD_SET_CFG, 0);
> +	if (IS_ERR(pkt))
> +		return PTR_ERR(pkt);
> +
> +	p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
> +	intf_cfg = p;
> +
> +	param_data = &intf_cfg->param_data;
> +	param_data->module_instance_id = module->instance_id;
> +	param_data->error_code = 0;
> +	param_data->param_id = PARAM_ID_AUDIO_IF_INTF_CFG;
> +	param_data->param_size = ic_sz - APM_MODULE_PARAM_DATA_SIZE;
> +	intf_cfg->cfg.qaif_type = module->qaif_type;
> +	intf_cfg->cfg.intf_idx = (u16)module->hw_interface_idx;
> +	intf_cfg->cfg.intf_mode = module->intf_mode;
> +	intf_cfg->cfg.ctrl_data_out_enable = module->ctrl_data_out_enable;
> +	intf_cfg->cfg.active_slot_mask = slot_mask;
> +	intf_cfg->cfg.nslots_per_frame = nslots_per_frame;
> +	intf_cfg->cfg.slot_width = slot_width;
> +	intf_cfg->cfg.active_lane_mask = module->active_lane_mask;
> +	intf_cfg->cfg.frame_sync_rate = module->frame_sync_rate;
> +	intf_cfg->cfg.frame_sync_src = module->sync_src;
> +	intf_cfg->cfg.frame_sync_mode = module->sync_mode;
> +	intf_cfg->cfg.invert_frame_sync_pulse = module->ctrl_invert_sync_pulse;
> +	intf_cfg->cfg.frame_sync_data_delay = module->ctrl_sync_data_delay;
> +	intf_cfg->cfg.bit_clk_type = module->bit_clk_type;
> +	intf_cfg->cfg.inv_int_bit_clk = module->inv_int_bit_clk;
> +	intf_cfg->cfg.inv_ext_bit_clk = module->inv_ext_bit_clk;
> +
> +	p += ic_sz;
> +	hw_cfg = p;
> +	param_data = &hw_cfg->param_data;
> +	param_data->module_instance_id = module->instance_id;
> +	param_data->error_code = 0;
> +	param_data->param_id = PARAM_ID_HW_EP_MF_CFG;
> +	param_data->param_size = ep_sz - APM_MODULE_PARAM_DATA_SIZE;
> +
> +	hw_cfg->mf.sample_rate = cfg->sample_rate;
> +	hw_cfg->mf.bit_width = cfg->bit_width;
> +	hw_cfg->mf.num_channels = cfg->num_channels;
> +	hw_cfg->mf.data_format = module->data_format;
> +
> +	p += ep_sz;
> +	fd_cfg = p;
> +	param_data = &fd_cfg->param_data;
> +	param_data->module_instance_id = module->instance_id;
> +	param_data->error_code = 0;
> +	param_data->param_id = PARAM_ID_HW_EP_FRAME_DURATION;
> +	param_data->param_size = fd_sz - APM_MODULE_PARAM_DATA_SIZE;
> +	fd_cfg->frame_duration.frame_duration_in_us = AUDIO_IF_FRAME_DURATION_US;
> +	fd_cfg->frame_duration.allow_frame_duration_normalization = AUDIO_IF_FRAME_DURATION_NORMALIZATION_ENABLE;
> +	fd_cfg->frame_duration.min_normalized_frame_dur_us = AUDIO_IF_FRAME_DURATION_MIN_US;
> +	fd_cfg->frame_duration.max_normalized_frame_dur_us = AUDIO_IF_FRAME_DURATION_MAX_US;
> +
> +	return q6apm_send_cmd_sync(graph->apm, pkt, 0);
> +}
> +
>  static int audioreach_logging_set_media_format(struct q6apm_graph *graph,
>  					       const struct audioreach_module *module)
>  {
> @@ -1411,6 +1500,10 @@ int audioreach_set_media_format(struct q6apm_graph *graph,
>  		if (!rc)
>  			rc = audioreach_module_enable(graph, module, true);
>  		break;
> +	case MODULE_ID_AUDIO_IF_SOURCE:
> +	case MODULE_ID_AUDIO_IF_SINK:
> +		rc = audioreach_audio_if_set_media_format(graph, module, cfg);
> +		break;
>  
>  	default:
>  		rc = 0;
> diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
> index 62a2fd79bbcb..35541b2d8c99 100644
> --- a/sound/soc/qcom/qdsp6/audioreach.h
> +++ b/sound/soc/qcom/qdsp6/audioreach.h
> @@ -36,6 +36,8 @@ struct q6apm_graph;
>  #define MODULE_ID_SPEAKER_PROTECTION	0x070010E2
>  #define MODULE_ID_SPEAKER_PROTECTION_VI	0x070010E3
>  #define MODULE_ID_OPUS_DEC		0x07001174
> +#define MODULE_ID_AUDIO_IF_SINK		0x0700117C
> +#define MODULE_ID_AUDIO_IF_SOURCE	0x0700117D
>  
>  #define APM_CMD_GET_SPF_STATE		0x01001021
>  #define APM_CMD_RSP_GET_SPF_STATE	0x02001007
> @@ -544,6 +546,74 @@ struct param_id_i2s_intf_cfg {
>  #define PORT_ID_I2S_OUPUT		1
>  #define I2S_STACK_SIZE			2048
>  
> +#define PARAM_ID_AUDIO_IF_INTF_CFG	0x08001B11
> +
> +/*
> + * struct param_id_audio_if_intf_cfg - Audio interface configuration
> + * @qaif_type: Audio interface type (e.g. QAIF, QAIF_VA)
> + * @intf_idx: Interface instance index
> + * @intf_mode: Interface operating mode (TDM/PCM/I2S)
> + * @ctrl_data_out_enable: Enable sharing of data-out signal with other masters
> + * @active_slot_mask: Bitmask indicating active slots
> + * @nslots_per_frame: Number of slots per audio frame
> + * @slot_width: Width of each slot in bits
> + * @active_lane_mask: Bitmask of active data lanes
> + * @frame_sync_rate: Frame sync rate in Hz
> + * @frame_sync_src: Frame sync source selection
> + * @frame_sync_mode: Frame sync mode configuration
> + * @invert_frame_sync_pulse: Invert frame sync polarity when set
> + * @frame_sync_data_delay: Data delay from frame sync in bit clocks
> + * @bit_clk_type: Bit clock type (internal / external)
> + * @inv_int_bit_clk: Invert internal bit clock when set
> + * @inv_ext_bit_clk: Invert external bit clock when set
> + *
> + * This structure defines configuration parameters for the Qualcomm
> + * Audio Interface (QAIF) block. It is used to program interface
> + * characteristics such as slot configuration, clocking and frame
> + * synchronization behaviour.
> + */
> +struct param_id_audio_if_intf_cfg {
> +	uint16_t qaif_type;
> +	uint16_t intf_idx;
> +	uint16_t intf_mode;
> +	uint16_t ctrl_data_out_enable;
> +	uint32_t active_slot_mask;
> +	uint16_t nslots_per_frame;
> +	uint16_t slot_width;
> +	uint32_t active_lane_mask;
> +	uint32_t frame_sync_rate;
> +	uint16_t frame_sync_src;
> +	uint16_t frame_sync_mode;
> +	uint16_t invert_frame_sync_pulse;
> +	uint16_t frame_sync_data_delay;
> +	uint16_t bit_clk_type;
> +	uint8_t inv_int_bit_clk;
> +	uint8_t inv_ext_bit_clk;
> +} __packed;
> +
> +#define PARAM_ID_HW_EP_FRAME_DURATION	0x08001B2F
> +#define AUDIO_IF_FRAME_DURATION_US			1000
> +#define AUDIO_IF_FRAME_DURATION_NORMALIZATION_ENABLE	1
> +#define AUDIO_IF_FRAME_DURATION_MIN_US			1
> +#define AUDIO_IF_FRAME_DURATION_MAX_US			100000
> +
> +/**
> + * struct param_id_hw_ep_frame_duration - Hardware endpoint frame duration
> + * @frame_duration_in_us: Frame duration in microseconds.
> + * @allow_frame_duration_normalization: Permit SPF to normalize frame duration.
> + * @min_normalized_frame_dur_us: Minimum normalized frame duration in microseconds.
> + * @max_normalized_frame_dur_us: Maximum normalized frame duration in microseconds.
> + *
> + * This structure configures the frame duration for the Audio IF hardware
> + * endpoint and, when enabled, the allowed normalization range.
> + */
> +struct param_id_hw_ep_frame_duration {
> +	uint32_t frame_duration_in_us;
> +	uint32_t allow_frame_duration_normalization;
> +	uint32_t min_normalized_frame_dur_us;
> +	uint32_t max_normalized_frame_dur_us;
> +} __packed;
> +
>  #define PARAM_ID_DISPLAY_PORT_INTF_CFG		0x08001154
>  
>  struct param_id_display_port_intf_cfg {
> @@ -877,6 +947,23 @@ struct audioreach_module {
>  	uint32_t data_format;
>  	uint32_t hw_interface_type;
>  
> +	/* Audio IF module (TDM/PCM/I2S) */
> +	u32 slot_mask;
> +	u32 active_lane_mask;
> +	u32 frame_sync_rate;
> +	u16 qaif_type;
> +	u16 sync_src;
> +	u16 ctrl_data_out_enable;
> +	u16 nslots_per_frame;
> +	u16 slot_width;
> +	u16 intf_mode;
> +	u16 sync_mode;
> +	u16 ctrl_invert_sync_pulse;
> +	u16 ctrl_sync_data_delay;
> +	u16 bit_clk_type;
> +	u8 inv_int_bit_clk;
> +	u8 inv_ext_bit_clk;
> +
>  	/* PCM module specific */
>  	uint32_t interleave_type;
>  
> @@ -907,6 +994,9 @@ struct audioreach_module_config {
>  	u32	channel_allocation;
>  	u32	sd_line_mask;
>  	int	fmt;
> +	u32	slot_mask;
> +	u16	nslots_per_frame;
> +	u16	slot_width;
>  	struct snd_codec codec;
>  	u8 channel_map[AR_PCM_MAX_NUM_CHANNEL];
>  };
> diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c
> index 1f69fba6de26..54661bcb006c 100644
> --- a/sound/soc/qcom/qdsp6/topology.c
> +++ b/sound/soc/qcom/qdsp6/topology.c
> @@ -753,6 +753,103 @@ static int audioreach_widget_i2s_module_load(struct audioreach_module *mod,
>  	return 0;
>  }
>  
> +static int audioreach_widget_audio_if_module_load(struct audioreach_module *mod,
> +						  const struct snd_soc_tplg_vendor_array *mod_array)
> +{
> +	const struct snd_soc_tplg_vendor_value_elem *mod_elem;
> +	int tkn_count = 0;
> +	u32 val;
> +
> +	mod_elem = mod_array->value;
> +
> +	while (tkn_count < le32_to_cpu(mod_array->num_elems)) {
> +		val = le32_to_cpu(mod_elem->value);
> +		switch (le32_to_cpu(mod_elem->token)) {
> +		case AR_TKN_U32_MODULE_HW_IF_IDX:
> +			mod->hw_interface_idx = val;
> +			break;
> +		case AR_TKN_U32_MODULE_FMT_DATA:
> +			mod->data_format = val;
> +			break;
> +		case AR_TKN_U16_MODULE_SYNC_SRC:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->sync_src = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_CTRL_DATA_OUT_ENABLE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->ctrl_data_out_enable = (u16)val;
> +			break;
> +		case AR_TKN_U32_MODULE_SLOT_MASK:
> +			mod->slot_mask = val;
> +			break;
> +		case AR_TKN_U16_MODULE_NSLOTS_PER_FRAME:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->nslots_per_frame = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_SLOT_WIDTH:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->slot_width = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_INTF_MODE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->intf_mode = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_SYNC_MODE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->sync_mode = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_CTRL_INVERT_SYNC_PULSE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->ctrl_invert_sync_pulse = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_CTRL_SYNC_DATA_DELAY:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->ctrl_sync_data_delay = (u16)val;
> +			break;
> +		case AR_TKN_U16_MODULE_QAIF_TYPE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->qaif_type = (u16)val;
> +			break;
> +		case AR_TKN_U32_MODULE_ACTIVE_LANE_MASK:
> +			mod->active_lane_mask = val;
> +			break;
> +		case AR_TKN_U32_MODULE_FRAME_SYNC_RATE:
> +			mod->frame_sync_rate = val;
> +			break;
> +		case AR_TKN_U16_MODULE_BIT_CLK_TYPE:
> +			if (val > U16_MAX)
> +				return -EINVAL;
> +			mod->bit_clk_type = (u16)val;
> +			break;
> +		case AR_TKN_U8_MODULE_INV_INT_BIT_CLK:
> +			if (val > U8_MAX)
> +				return -EINVAL;
> +			mod->inv_int_bit_clk = (u8)val;
> +			break;
> +		case AR_TKN_U8_MODULE_INV_EXT_BIT_CLK:
> +			if (val > U8_MAX)
> +				return -EINVAL;
> +			mod->inv_ext_bit_clk = (u8)val;
> +			break;
> +		default:
> +			break;
> +		}
> +		tkn_count++;
> +		mod_elem++;
> +	}
> +
> +	return 0;
> +}
> +
>  static int audioreach_widget_dp_module_load(struct audioreach_module *mod,
>  					    const struct snd_soc_tplg_vendor_array *mod_array)
>  {
> @@ -806,6 +903,12 @@ static int audioreach_widget_load_buffer(struct snd_soc_component *component,
>  	case MODULE_ID_I2S_SOURCE:
>  		audioreach_widget_i2s_module_load(mod, mod_array);
>  		break;
> +	case MODULE_ID_AUDIO_IF_SINK:
> +	case MODULE_ID_AUDIO_IF_SOURCE:
> +		ret = audioreach_widget_audio_if_module_load(mod, mod_array);
> +		if (ret)
> +			return ret;
> +		break;
>  	case MODULE_ID_DISPLAY_PORT_SINK:
>  		audioreach_widget_dp_module_load(mod, mod_array);
>  		break;