Re: [PATCH v4 1/7] ASoC: qcom: qdsp6: add topology-driven Audio IF support
From: Srinivas Kandagatla
Date: Tue Jul 14 2026 - 04:30:41 EST
On 7/12/26 2:41 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 <prasad.kumpatla@xxxxxxxxxxxxxxxx>
> ---
> include/uapi/sound/snd_ar_tokens.h | 89 ++++++++++++++++++++++++
> sound/soc/qcom/qdsp6/audioreach.c | 97 ++++++++++++++++++++++++++
> sound/soc/qcom/qdsp6/audioreach.h | 81 ++++++++++++++++++++++
> sound/soc/qcom/qdsp6/topology.c | 105 +++++++++++++++++++++++++++++
> 4 files changed, 372 insertions(+)
>
> diff --git a/include/uapi/sound/snd_ar_tokens.h b/include/uapi/sound/snd_ar_tokens.h
> index 6b8102eaa..bda65311c 100644
> --- a/include/uapi/sound/snd_ar_tokens.h
> +++ b/include/uapi/sound/snd_ar_tokens.h
> @@ -168,6 +168,58 @@ 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_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 +292,43 @@ 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_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 e6e9eb2e8..62140ce8e 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 +1056,85 @@ 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;
> + /*
> + * A zero machine override means use the topology default; zero is not
> + * a valid override for these TDM slot fields.
> + */
What is this comment about, it does not make any sense w.r.t to where
its placed, either remove it or put it correct place.
> + 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 = 1;
> + fd_cfg->frame_duration.min_normalized_frame_dur_us = 1;
> + fd_cfg->frame_duration.max_normalized_frame_dur_us = 100000;
Can we define these magic values somewhere as default values.
> +
> + 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 +1504,10 @@ 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 62a2fd79b..d57ba4e6a 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 +546,65 @@ 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
> +
> +#define AUDIO_IF_INTF_MODE_TDM AR_AUDIO_IF_INTF_MODE_TDM
> +#define AUDIO_IF_INTF_MODE_PCM AR_AUDIO_IF_INTF_MODE_PCM
> +#define AUDIO_IF_INTF_MODE_I2S AR_AUDIO_IF_INTF_MODE_I2S
> +
> +/*
> + * 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 {
> + u16 qaif_type;
> + u16 intf_idx;
> + u16 intf_mode;
> + u16 ctrl_data_out_enable;
> + u32 active_slot_mask;
this header uses uint32_t types, please stick to such style for consistency.
> + u16 nslots_per_frame;
> + u16 slot_width;
> + u32 active_lane_mask;
> + u32 frame_sync_rate;
> + u16 frame_sync_src;
> + u16 frame_sync_mode;
> + u16 invert_frame_sync_pulse;
> + u16 frame_sync_data_delay;
> + u16 bit_clk_type;
> + u8 inv_int_bit_clk;
> + u8 inv_ext_bit_clk;
> +} __packed;
> +
> +#define PARAM_ID_HW_EP_FRAME_DURATION 0x08001B2F
> +#define AUDIO_IF_FRAME_DURATION_US 1000
> +
> +struct param_id_hw_ep_frame_duration {
> + u32 frame_duration_in_us;
> + u32 allow_frame_duration_normalization;
> + u32 min_normalized_frame_dur_us;
> + u32 max_normalized_frame_dur_us;
> +} __packed;
> +
> #define PARAM_ID_DISPLAY_PORT_INTF_CFG 0x08001154
>
> struct param_id_display_port_intf_cfg {
> @@ -877,6 +938,23 @@ 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 +985,9 @@ 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 1f69fba6d..3dd0a5c29 100644
> --- a/sound/soc/qcom/qdsp6/topology.c
> +++ b/sound/soc/qcom/qdsp6/topology.c
> @@ -753,6 +753,105 @@ 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:
> + if (val > U16_MAX)
> + return -EINVAL;
> + 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 +905,12 @@ 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;