From: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
To: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>,
Srinivas Kandagatla <srini@kernel.org>,
Liam Girdwood <lgirdwood@gmail.com>,
Mark Brown <broonie@kernel.org>, Rob Herring <robh@kernel.org>,
Krzysztof Kozlowski <krzk+dt@kernel.org>,
Conor Dooley <conor+dt@kernel.org>,
Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.com>
Cc: linux-arm-msm@vger.kernel.org, linux-sound@vger.kernel.org,
devicetree@vger.kernel.org, linux-kernel@vger.kernel.org
Subject: Re: [PATCH v5 1/7] ASoC: qcom: qdsp6: add topology-driven Audio IF support
Date: Mon, 3 Aug 2026 21:13:05 +0530 [thread overview]
Message-ID: <084f16eb-2d7e-4cb7-843b-e2dc1722aeb0@oss.qualcomm.com> (raw)
In-Reply-To: <ce580ddc-6bd0-4e34-b399-ef2919e6bf6a@oss.qualcomm.com>
On 8/2/2026 6:29 PM, Srinivas Kandagatla wrote:
> On 8/1/26 6:46 AM, 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@oss.qualcomm.com>
>> ---
> Small nits below.
>
>> include/uapi/sound/snd_ar_tokens.h | 89 ++++++++++++++++++++++++
>> sound/soc/qcom/qdsp6/audioreach.c | 93 +++++++++++++++++++++++++
>> sound/soc/qcom/qdsp6/audioreach.h | 84 +++++++++++++++++++++++
>> sound/soc/qcom/qdsp6/topology.c | 105 +++++++++++++++++++++++++++++
>> 4 files changed, 371 insertions(+)
>>
>> diff --git a/include/uapi/sound/snd_ar_tokens.h b/include/uapi/sound/snd_ar_tokens.h
>> index 6b8102eaa121..bda65311cfa9 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
> What are the supported types for this?
Hi Srini,
Thanks for the review.
Below are supported types. I will add them as well in next revision.
Range: 0x0 - QAIF
0x1 - QAIF_VA
>
>> + *
>> + * %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 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..924e85f4fdf6 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 +546,68 @@ 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
> Dead code.
Ack, will remove.
>
>
>> +
>> +/*
>> + * 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 {
> Any reason why no kernel doc for this?
Will Update in Next revision.
>
>> + 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 +941,23 @@ 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 +988,9 @@ 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..3dd0a5c29a6e 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)
> this check is inconsistent with supported range.
Good catch. The token represents a U32 value and the |U16_MAX| check is
unnecessarily restrictive.
I'll update the validation accordingly.
Thanks,
Prasad
>
>
>> + 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;
next prev parent reply other threads:[~2026-08-03 15:43 UTC|newest]
Thread overview: 17+ messages / expand[flat|nested] mbox.gz Atom feed top
2026-08-01 5:46 [PATCH v5 0/7] ASoC: qcom: add AudioReach TDM backend support Prasad Kumpatla
2026-08-01 5:46 ` [PATCH v5 1/7] ASoC: qcom: qdsp6: add topology-driven Audio IF support Prasad Kumpatla
2026-08-02 12:59 ` Srinivas Kandagatla
2026-08-03 15:43 ` Prasad Kumpatla [this message]
2026-08-01 5:46 ` [PATCH v5 2/7] ASoC: qcom: q6apm-lpass-dais: add TDM DAI operations Prasad Kumpatla
2026-08-02 13:08 ` Srinivas Kandagatla
2026-08-01 5:46 ` [PATCH v5 3/7] dt-bindings: sound: qcom,q6dsp-lpass-ports: add Audio IF clocks Prasad Kumpatla
2026-08-02 13:30 ` Srinivas Kandagatla
2026-08-03 15:50 ` Prasad Kumpatla
2026-08-03 14:10 ` Rob Herring
2026-08-03 16:07 ` Prasad Kumpatla
2026-08-01 5:46 ` [PATCH v5 4/7] ASoC: qcom: q6prm: add Audio IF clock IDs Prasad Kumpatla
2026-08-01 5:46 ` [PATCH v5 5/7] dt-bindings: sound: qcom,sm8250: allow TDM slot properties Prasad Kumpatla
2026-08-01 5:46 ` [PATCH v5 6/7] ASoC: qcom: common: add DAI-node TDM slot helpers Prasad Kumpatla
2026-08-01 5:46 ` [PATCH v5 7/7] ASoC: qcom: sc8280xp: add TDM hw_params support Prasad Kumpatla
2026-08-03 13:07 ` [PATCH v5 0/7] ASoC: qcom: add AudioReach TDM backend support Mark Brown
2026-08-03 18:09 ` Prasad Kumpatla
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