ASoC: qcom: audioreach: add compress offload

Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:

This patchset adds compressed offload support to Qualcomm audioreach drivers.
Currently it supports AAC, MP3 and FALC along with gapless.

Tested this on SM8450 and sc7280.

sound/soc/qcom/qdsp6/audioreach.c

@@ -732,33 +732,32 @@ static int audioreach_codec_dma_set_media_format(struct q6apm_graph *graph,
 	return rc;
 }
 
-static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
-					 struct audioreach_module *module, bool enable)
+int audioreach_send_u32_param(struct q6apm_graph *graph, struct audioreach_module *module,
+			      uint32_t param_id, uint32_t param_val)
 {
 	struct apm_module_param_data *param_data;
-	struct param_id_sal_limiter_enable *limiter_enable;
-	int payload_size;
 	struct gpr_pkt *pkt;
-	int rc;
+	uint32_t *param;
+	int rc, payload_size;
 	void *p;
 
-	payload_size = sizeof(*limiter_enable) + APM_MODULE_PARAM_DATA_SIZE;
-
-	pkt = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
-	if (IS_ERR(pkt))
-		return PTR_ERR(pkt);
+	payload_size = sizeof(uint32_t) + APM_MODULE_PARAM_DATA_SIZE;
+	p = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
+	if (IS_ERR(p))
+		return -ENOMEM;
 
-	p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
+	pkt = p;
+	p = p + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
 
 	param_data = p;
 	param_data->module_instance_id = module->instance_id;
 	param_data->error_code = 0;
-	param_data->param_id = PARAM_ID_SAL_LIMITER_ENABLE;
-	param_data->param_size = sizeof(*limiter_enable);
-	p = p + APM_MODULE_PARAM_DATA_SIZE;
-	limiter_enable = p;
+	param_data->param_id = param_id;
+	param_data->param_size = sizeof(uint32_t);
 
-	limiter_enable->enable_lim = enable;
+	p = p + APM_MODULE_PARAM_DATA_SIZE;
+	param = p;
+	*param = param_val;
 
 	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
 
@@ -766,77 +765,34 @@ static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
 
 	return rc;
 }
+EXPORT_SYMBOL_GPL(audioreach_send_u32_param);
+
+static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
+					 struct audioreach_module *module, bool enable)
+{
+	return audioreach_send_u32_param(graph, module, PARAM_ID_SAL_LIMITER_ENABLE, enable);
+}
 
 static int audioreach_sal_set_media_format(struct q6apm_graph *graph,
 					   struct audioreach_module *module,
 					   struct audioreach_module_config *cfg)
 {
-	struct apm_module_param_data *param_data;
-	struct param_id_sal_output_config *media_format;
-	int payload_size;
-	struct gpr_pkt *pkt;
-	int rc;
-	void *p;
-
-	payload_size = sizeof(*media_format) + APM_MODULE_PARAM_DATA_SIZE;
-
-	pkt = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
-	if (IS_ERR(pkt))
-		return PTR_ERR(pkt);
-
-	p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
-
-	param_data = p;
-	param_data->module_instance_id = module->instance_id;
-	param_data->error_code = 0;
-	param_data->param_id = PARAM_ID_SAL_OUTPUT_CFG;
-	param_data->param_size = sizeof(*media_format);
-	p = p + APM_MODULE_PARAM_DATA_SIZE;
-	media_format = p;
-
-	media_format->bits_per_sample = cfg->bit_width;
-
-	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
-
-	kfree(pkt);
-
-	return rc;
+	return audioreach_send_u32_param(graph, module, PARAM_ID_SAL_OUTPUT_CFG,  cfg->bit_width);
 }
 
 static int audioreach_module_enable(struct q6apm_graph *graph,
 				    struct audioreach_module *module,
 				    bool enable)
 {
-	struct apm_module_param_data *param_data;
-	struct param_id_module_enable *param;
-	int payload_size;
-	struct gpr_pkt *pkt;
-	int rc;
-	void *p;
-
-	payload_size = sizeof(*param) + APM_MODULE_PARAM_DATA_SIZE;
-
-	pkt = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
-	if (IS_ERR(pkt))
-		return PTR_ERR(pkt);
-
-	p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
-
-	param_data = p;
-	param_data->module_instance_id = module->instance_id;
-	param_data->error_code = 0;
-	param_data->param_id = PARAM_ID_MODULE_ENABLE;
-	param_data->param_size = sizeof(*param);
-	p = p + APM_MODULE_PARAM_DATA_SIZE;
-	param = p;
-
-	param->enable = enable;
-
-	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
-
-	kfree(pkt);
+	return audioreach_send_u32_param(graph, module, PARAM_ID_MODULE_ENABLE, enable);
+}
 
-	return rc;
+static int audioreach_gapless_set_media_format(struct q6apm_graph *graph,
+					       struct audioreach_module *module,
+					       struct audioreach_module_config *cfg)
+{
+	return audioreach_send_u32_param(graph, module, PARAM_ID_EARLY_EOS_DELAY,
+					 EARLY_EOS_DELAY_MS);
 }
 
 static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
@@ -886,6 +842,99 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
 	return rc;
 }
 
+static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
+					     void *p, struct audioreach_module_config *mcfg)
+{
+	struct payload_media_fmt_aac_t *aac_cfg;
+	struct payload_media_fmt_pcm *mp3_cfg;
+	struct payload_media_fmt_flac_t *flac_cfg;
+
+	switch (mcfg->fmt) {
+	case SND_AUDIOCODEC_MP3:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_MP3;
+		media_fmt_hdr->payload_size = 0;
+		p = p + sizeof(*media_fmt_hdr);
+		mp3_cfg = p;
+		mp3_cfg->sample_rate = mcfg->sample_rate;
+		mp3_cfg->bit_width = mcfg->bit_width;
+		mp3_cfg->alignment = PCM_LSB_ALIGNED;
+		mp3_cfg->bits_per_sample = mcfg->bit_width;
+		mp3_cfg->q_factor = mcfg->bit_width - 1;
+		mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
+		mp3_cfg->num_channels = mcfg->num_channels;
+
+		if (mcfg->num_channels == 1) {
+			mp3_cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+		} else if (mcfg->num_channels == 2) {
+			mp3_cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+			mp3_cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+		}
+		break;
+	case SND_AUDIOCODEC_AAC:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_AAC;
+		media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_aac_t);
+		p = p + sizeof(*media_fmt_hdr);
+		aac_cfg = p;
+		aac_cfg->aac_fmt_flag = 0;
+		aac_cfg->audio_obj_type = 5;
+		aac_cfg->num_channels = mcfg->num_channels;
+		aac_cfg->total_size_of_PCE_bits = 0;
+		aac_cfg->sample_rate = mcfg->sample_rate;
+		break;
+	case SND_AUDIOCODEC_FLAC:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_FLAC;
+		media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_flac_t);
+		p = p + sizeof(*media_fmt_hdr);
+		flac_cfg = p;
+		flac_cfg->sample_size = mcfg->codec.options.flac_d.sample_size;
+		flac_cfg->num_channels = mcfg->num_channels;
+		flac_cfg->min_blk_size = mcfg->codec.options.flac_d.min_blk_size;
+		flac_cfg->max_blk_size = mcfg->codec.options.flac_d.max_blk_size;
+		flac_cfg->sample_rate = mcfg->sample_rate;
+		flac_cfg->min_frame_size = mcfg->codec.options.flac_d.min_frame_size;
+		flac_cfg->max_frame_size = mcfg->codec.options.flac_d.max_frame_size;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
+{
+	struct media_format *header;
+	struct gpr_pkt *pkt;
+	int iid, payload_size, rc;
+	void *p;
+
+	payload_size = sizeof(struct apm_sh_module_media_fmt_cmd);
+
+	iid = q6apm_graph_get_rx_shmem_module_iid(graph);
+	pkt = audioreach_alloc_cmd_pkt(payload_size, DATA_CMD_WR_SH_MEM_EP_MEDIA_FORMAT,
+			0, graph->port->id, iid);
+
+	if (IS_ERR(pkt))
+		return -ENOMEM;
+
+	p = (void *)pkt + GPR_HDR_SIZE;
+	header = p;
+	rc = audioreach_set_compr_media_format(header, p, mcfg);
+	if (rc) {
+		kfree(pkt);
+		return rc;
+	}
+
+	rc = gpr_send_port_pkt(graph->port, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(audioreach_compr_set_param);
+
 static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
 					   struct audioreach_module *module,
 					   struct audioreach_module_config *cfg)
@@ -1089,25 +1138,33 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
 	p = p + APM_MODULE_PARAM_DATA_SIZE;
 
 	header = p;
-	header->data_format = DATA_FORMAT_FIXED_POINT;
-	header->fmt_id = MEDIA_FMT_ID_PCM;
-	header->payload_size = payload_size - sizeof(*header);
-
-	p = p + sizeof(*header);
-	cfg = p;
-	cfg->sample_rate = mcfg->sample_rate;
-	cfg->bit_width = mcfg->bit_width;
-	cfg->alignment = PCM_LSB_ALIGNED;
-	cfg->bits_per_sample = mcfg->bit_width;
-	cfg->q_factor = mcfg->bit_width - 1;
-	cfg->endianness = PCM_LITTLE_ENDIAN;
-	cfg->num_channels = mcfg->num_channels;
-
-	if (mcfg->num_channels == 1) {
-		cfg->channel_mapping[0] =  PCM_CHANNEL_L;
-	} else if (num_channels == 2) {
-		cfg->channel_mapping[0] =  PCM_CHANNEL_L;
-		cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+	if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
+		header->data_format = DATA_FORMAT_FIXED_POINT;
+		header->fmt_id =  MEDIA_FMT_ID_PCM;
+		header->payload_size = payload_size - sizeof(*header);
+
+		p = p + sizeof(*header);
+		cfg = p;
+		cfg->sample_rate = mcfg->sample_rate;
+		cfg->bit_width = mcfg->bit_width;
+		cfg->alignment = PCM_LSB_ALIGNED;
+		cfg->bits_per_sample = mcfg->bit_width;
+		cfg->q_factor = mcfg->bit_width - 1;
+		cfg->endianness = PCM_LITTLE_ENDIAN;
+		cfg->num_channels = mcfg->num_channels;
+
+		if (mcfg->num_channels == 1)
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+		else if (num_channels == 2) {
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+			cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+		}
+	} else {
+		rc = audioreach_set_compr_media_format(header, p, mcfg);
+		if (rc) {
+			kfree(pkt);
+			return rc;
+		}
 	}
 
 	rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);
@@ -1192,6 +1249,8 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
 	case MODULE_ID_PCM_DEC:
 	case MODULE_ID_PCM_ENC:
 	case MODULE_ID_PCM_CNV:
+	case MODULE_ID_PLACEHOLDER_DECODER:
+	case MODULE_ID_PLACEHOLDER_ENCODER:
 		rc = audioreach_pcm_set_media_format(graph, module, cfg);
 		break;
 	case MODULE_ID_DISPLAY_PORT_SINK:
@@ -1219,6 +1278,9 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
 	case MODULE_ID_MFC:
 		rc = audioreach_mfc_set_media_format(graph, module, cfg);
 		break;
+	case MODULE_ID_GAPLESS:
+		rc = audioreach_gapless_set_media_format(graph, module, cfg);
+		break;
 	default:
 		rc = 0;
 	}

sound/soc/qcom/qdsp6/audioreach.h

@@ -15,6 +15,8 @@ struct q6apm_graph;
 #define MODULE_ID_PCM_CNV		0x07001003
 #define MODULE_ID_PCM_ENC		0x07001004
 #define MODULE_ID_PCM_DEC		0x07001005
+#define MODULE_ID_PLACEHOLDER_ENCODER	0x07001008
+#define MODULE_ID_PLACEHOLDER_DECODER	0x07001009
 #define MODULE_ID_SAL			0x07001010
 #define MODULE_ID_MFC			0x07001015
 #define MODULE_ID_CODEC_DMA_SINK	0x07001023
@@ -22,6 +24,10 @@ struct q6apm_graph;
 #define MODULE_ID_I2S_SINK		0x0700100A
 #define MODULE_ID_I2S_SOURCE		0x0700100B
 #define MODULE_ID_DATA_LOGGING		0x0700101A
+#define MODULE_ID_AAC_DEC		0x0700101F
+#define MODULE_ID_FLAC_DEC		0x0700102F
+#define MODULE_ID_MP3_DECODE		0x0700103B
+#define MODULE_ID_GAPLESS		0x0700104D
 #define MODULE_ID_DISPLAY_PORT_SINK	0x07001069
 
 #define APM_CMD_GET_SPF_STATE		0x01001021
@@ -143,12 +149,15 @@ struct param_id_enc_bitrate_param {
 } __packed;
 
 #define DATA_FORMAT_FIXED_POINT		1
+#define DATA_FORMAT_GENERIC_COMPRESSED	5
+#define DATA_FORMAT_RAW_COMPRESSED	6
 #define PCM_LSB_ALIGNED			1
 #define PCM_MSB_ALIGNED			2
 #define PCM_LITTLE_ENDIAN		1
 #define PCM_BIT_ENDIAN			2
 
 #define MEDIA_FMT_ID_PCM	0x09001000
+#define MEDIA_FMT_ID_MP3	0x09001009
 #define PCM_CHANNEL_L		1
 #define PCM_CHANNEL_R		2
 #define SAMPLE_RATE_48K		48000
@@ -226,6 +235,28 @@ struct apm_media_format {
 	uint32_t payload_size;
 } __packed;
 
+#define MEDIA_FMT_ID_FLAC	0x09001004
+
+struct payload_media_fmt_flac_t {
+	uint16_t num_channels;
+	uint16_t sample_size;
+	uint16_t min_blk_size;
+	uint16_t max_blk_size;
+	uint32_t sample_rate;
+	uint32_t min_frame_size;
+	uint32_t max_frame_size;
+} __packed;
+
+#define MEDIA_FMT_ID_AAC	0x09001001
+
+struct payload_media_fmt_aac_t {
+	uint16_t aac_fmt_flag;
+	uint16_t audio_obj_type;
+	uint16_t num_channels;
+	uint16_t total_size_of_PCE_bits;
+	uint32_t sample_rate;
+} __packed;
+
 #define DATA_CMD_WR_SH_MEM_EP_EOS			0x04001002
 #define WR_SH_MEM_EP_EOS_POLICY_LAST	1
 #define WR_SH_MEM_EP_EOS_POLICY_EACH	2
@@ -522,6 +553,8 @@ struct param_id_sal_limiter_enable {
 } __packed;
 
 #define PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT	0x08001024
+#define PARAM_ID_EARLY_EOS_DELAY		0x0800114C
+#define EARLY_EOS_DELAY_MS			150
 
 struct param_id_mfc_media_format {
 	uint32_t sample_rate;
@@ -530,6 +563,10 @@ struct param_id_mfc_media_format {
 	uint16_t channel_mapping[];
 } __packed;
 
+struct param_id_gapless_early_eos_delay_t {
+	uint32_t early_eos_delay_ms;
+} __packed;
+
 struct media_format {
 	uint32_t data_format;
 	uint32_t fmt_id;
@@ -608,6 +645,15 @@ struct param_id_vol_ctrl_master_gain {
 } __packed;
 
 
+#define PARAM_ID_REMOVE_INITIAL_SILENCE		0x0800114B
+#define PARAM_ID_REMOVE_TRAILING_SILENCE	0x0800115D
+
+#define PARAM_ID_REAL_MODULE_ID	0x0800100B
+
+struct param_id_placeholder_real_module_id {
+	uint32_t real_module_id;
+} __packed;
+
 /* Graph */
 struct audioreach_connection {
 	/* Connections */
@@ -716,6 +762,7 @@ struct audioreach_module_config {
 	u32	channel_allocation;
 	u32	sd_line_mask;
 	int	fmt;
+	struct snd_codec codec;
 	u8 channel_map[AR_PCM_MAX_NUM_CHANNEL];
 };
 
@@ -752,4 +799,8 @@ int audioreach_set_media_format(struct q6apm_graph *graph,
 int audioreach_shared_memory_send_eos(struct q6apm_graph *graph);
 int audioreach_gain_set_vol_ctrl(struct q6apm *apm,
 				 struct audioreach_module *module, int vol);
+int audioreach_send_u32_param(struct q6apm_graph *graph, struct audioreach_module *module,
+			      uint32_t param_id, uint32_t param_val);
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
+
 #endif /* __AUDIOREACH_H__ */

sound/soc/qcom/qdsp6/q6apm.c

@@ -298,6 +298,71 @@ int q6apm_unmap_memory_regions(struct q6apm_graph *graph, unsigned int dir)
 }
 EXPORT_SYMBOL_GPL(q6apm_unmap_memory_regions);
 
+int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
+{
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_send_u32_param(graph, module, PARAM_ID_REMOVE_INITIAL_SILENCE, samples);
+}
+EXPORT_SYMBOL_GPL(q6apm_remove_initial_silence);
+
+int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
+{
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_send_u32_param(graph, module, PARAM_ID_REMOVE_TRAILING_SILENCE, samples);
+}
+EXPORT_SYMBOL_GPL(q6apm_remove_trailing_silence);
+
+int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en)
+{
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_send_u32_param(graph, module, PARAM_ID_MODULE_ENABLE, en);
+}
+EXPORT_SYMBOL_GPL(q6apm_enable_compress_module);
+
+int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph,
+			     uint32_t codec_id)
+{
+	struct audioreach_module *module;
+	uint32_t module_id;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	switch (codec_id) {
+	case SND_AUDIOCODEC_MP3:
+		module_id = MODULE_ID_MP3_DECODE;
+		break;
+	case SND_AUDIOCODEC_AAC:
+		module_id = MODULE_ID_AAC_DEC;
+		break;
+	case SND_AUDIOCODEC_FLAC:
+		module_id = MODULE_ID_FLAC_DEC;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return audioreach_send_u32_param(graph, module, PARAM_ID_REAL_MODULE_ID,
+					 module_id);
+}
+EXPORT_SYMBOL_GPL(q6apm_set_real_module_id);
+
 int q6apm_graph_media_format_pcm(struct q6apm_graph *graph, struct audioreach_module_config *cfg)
 {
 	struct audioreach_graph_info *info = graph->info;
@@ -497,6 +562,9 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		}
 		break;
 	case DATA_CMD_WR_SH_MEM_EP_EOS_RENDERED:
+		client_event = APM_CLIENT_EVENT_CMD_EOS_DONE;
+		if (graph->cb)
+			graph->cb(client_event, hdr->token, data->payload, graph->priv);
 		break;
 	case GPR_BASIC_RSP_RESULT:
 		switch (result->opcode) {

sound/soc/qcom/qdsp6/q6apm.h

@@ -45,6 +45,8 @@
 #define APM_WRITE_TOKEN_LEN_SHIFT              16
 
 #define APM_MAX_SESSIONS			8
+#define APM_LAST_BUFFER_FLAG			BIT(30)
+#define NO_TIMESTAMP				0xFF00
 
 struct q6apm {
 	struct device *dev;
@@ -147,4 +149,8 @@ int q6apm_graph_get_rx_shmem_module_iid(struct q6apm_graph *graph);
 
 bool q6apm_is_adsp_ready(void);
 
+int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en);
+int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
+int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
+int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id);
 #endif /* __APM_GRAPH_ */

sound/soc/qcom/sc7280.c

@@ -14,6 +14,7 @@
 #include <sound/soc.h>
 #include <sound/rt5682s.h>
 #include <linux/soundwire/sdw.h>
+#include <sound/pcm_params.h>
 
 #include "../codecs/rt5682.h"
 #include "../codecs/rt5682s.h"
@@ -196,8 +197,10 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
 	struct sdw_stream_runtime *sruntime;
 	int i;
 
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, 48000, 48000);
+	if (!rtd->dai_link->no_pcm) {
+		snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+		snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, 48000, 48000);
+	}
 
 	switch (cpu_dai->id) {
 	case LPASS_CDC_DMA_TX3:
@@ -358,6 +361,20 @@ static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = {
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 };
 
+static int sc7280_snd_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+					 struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
 static int sc7280_snd_platform_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card;
@@ -387,6 +404,8 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev)
 	for_each_card_prelinks(card, i, link) {
 		link->init = sc7280_init;
 		link->ops = &sc7280_ops;
+		if (link->no_pcm == 1)
+			link->be_hw_params_fixup = sc7280_snd_be_hw_params_fixup;
 	}
 
 	return devm_snd_soc_register_card(dev, card);