ASoC: Fixes for v6.12

An uncomfortably large set of fixes due to me not getting round to
 sending them for longer than I should due to travel and illness.  This
 is mostly smaller driver specific changes, but there are a couple of
 generic changes:
 
  - Bumping the minimal topology ABI we check for during validation, the
    code had support for v4 removed previously but the update of the
    define used for initial validation was missed.
  - Fix the assumption that DAPM structs will be embedded in a component
    which isn't true for card widgets when doing name comparisons, though
    fortunately this is rarely triggered.
 
 We've pulled in one Soundwire fix which was part of a larger series
 fixing cleanup issues in on Intel Soundwire systems.
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Merge tag 'asoc-fix-v6.12-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.12

An uncomfortably large set of fixes due to me not getting round to
sending them for longer than I should due to travel and illness.  This
is mostly smaller driver specific changes, but there are a couple of
generic changes:

 - Bumping the minimal topology ABI we check for during validation, the
   code had support for v4 removed previously but the update of the
   define used for initial validation was missed.
 - Fix the assumption that DAPM structs will be embedded in a component
   which isn't true for card widgets when doing name comparisons, though
   fortunately this is rarely triggered.

We've pulled in one Soundwire fix which was part of a larger series
fixing cleanup issues in on Intel Soundwire systems.

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml

@@ -102,21 +102,21 @@ properties:
     default: 2
 
   interrupts:
-    oneOf:
+    minItems: 1
-      - minItems: 1
+    maxItems: 2
-        items:
-          - description: TX interrupt
-          - description: RX interrupt
-      - items:
-          - description: common/combined interrupt
 
   interrupt-names:
     oneOf:
-      - minItems: 1
+      - description: TX interrupt
+        const: tx
+      - description: RX interrupt
+        const: rx
+      - description: TX and RX interrupts
         items:
           - const: tx
           - const: rx
-      - const: common
+      - description: Common/combined interrupt
+        const: common
 
   fck_parent:
     $ref: /schemas/types.yaml#/definitions/string

MAINTAINERS

@@ -15089,6 +15089,7 @@ F:	drivers/spi/spi-at91-usart.c
 
 MICROCHIP AUDIO ASOC DRIVERS
 M:	Claudiu Beznea <claudiu.beznea@tuxon.dev>
+M:	Andrei Simion <andrei.simion@microchip.com>
 L:	linux-sound@vger.kernel.org
 S:	Supported
 F:	Documentation/devicetree/bindings/sound/atmel*
@@ -15197,6 +15198,7 @@ F:	include/video/atmel_lcdc.h
 
 MICROCHIP MCP16502 PMIC DRIVER
 M:	Claudiu Beznea <claudiu.beznea@tuxon.dev>
+M:	Andrei Simion <andrei.simion@microchip.com>
 L:	linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
 S:	Supported
 F:	Documentation/devicetree/bindings/regulator/microchip,mcp16502.yaml
@@ -15328,6 +15330,7 @@ F:	drivers/spi/spi-atmel.*
 
 MICROCHIP SSC DRIVER
 M:	Claudiu Beznea <claudiu.beznea@tuxon.dev>
+M:	Andrei Simion <andrei.simion@microchip.com>
 L:	linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
 S:	Supported
 F:	Documentation/devicetree/bindings/misc/atmel-ssc.txt
@@ -23287,7 +23290,7 @@ F:	Documentation/devicetree/bindings/iio/adc/ti,lmp92064.yaml
 F:	drivers/iio/adc/ti-lmp92064.c
 
 TI PCM3060 ASoC CODEC DRIVER
-M:	Kirill Marinushkin <kmarinushkin@birdec.com>
+M:	Kirill Marinushkin <k.marinushkin@gmail.com>
 L:	linux-sound@vger.kernel.org
 S:	Maintained
 F:	Documentation/devicetree/bindings/sound/pcm3060.txt

drivers/soundwire/intel_ace2x.c

@@ -376,11 +376,12 @@ static int intel_hw_params(struct snd_pcm_substream *substream,
 static int intel_prepare(struct snd_pcm_substream *substream,
 			 struct snd_soc_dai *dai)
 {
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
 	struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai);
 	struct sdw_intel *sdw = cdns_to_intel(cdns);
 	struct sdw_cdns_dai_runtime *dai_runtime;
+	struct snd_pcm_hw_params *hw_params;
 	int ch, dir;
-	int ret = 0;
 
 	dai_runtime = cdns->dai_runtime_array[dai->id];
 	if (!dai_runtime) {
@@ -389,12 +390,8 @@ static int intel_prepare(struct snd_pcm_substream *substream,
 		return -EIO;
 	}
 
+	hw_params = &rtd->dpcm[substream->stream].hw_params;
 	if (dai_runtime->suspended) {
-		struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
-		struct snd_pcm_hw_params *hw_params;
-
-		hw_params = &rtd->dpcm[substream->stream].hw_params;
-
 		dai_runtime->suspended = false;
 
 		/*
@@ -415,15 +412,11 @@ static int intel_prepare(struct snd_pcm_substream *substream,
 		/* the SHIM will be configured in the callback functions */
 
 		sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi);
-
-		/* Inform DSP about PDI stream number */
-		ret = intel_params_stream(sdw, substream, dai,
-					  hw_params,
-					  sdw->instance,
-					  dai_runtime->pdi->intel_alh_id);
 	}
 
-	return ret;
+	/* Inform DSP about PDI stream number */
+	return intel_params_stream(sdw, substream, dai, hw_params, sdw->instance,
+				   dai_runtime->pdi->intel_alh_id);
 }
 
 static int

include/uapi/sound/asoc.h

@@ -88,7 +88,7 @@
 
 /* ABI version */
 #define SND_SOC_TPLG_ABI_VERSION	0x5	/* current version */
-#define SND_SOC_TPLG_ABI_VERSION_MIN	0x4	/* oldest version supported */
+#define SND_SOC_TPLG_ABI_VERSION_MIN	0x5	/* oldest version supported */
 
 /* Max size of TLV data */
 #define SND_SOC_TPLG_TLV_SIZE		32

sound/soc/amd/yc/acp6x-mach.c

@@ -325,6 +325,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "E1404FA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
@@ -339,6 +346,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M7600RE"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "M3502RA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/codecs/aw88399.c

@@ -656,7 +656,7 @@ static int aw_dev_get_dsp_status(struct aw_device *aw_dev)
 	if (ret)
 		return ret;
 	if (!(reg_val & (~AW88399_WDT_CNT_MASK)))
-		ret = -EPERM;
+		return -EPERM;
 
 	return 0;
 }

sound/soc/codecs/lpass-rx-macro.c

@@ -202,12 +202,14 @@
 #define CDC_RX_RXn_RX_PATH_SEC3(rx, n)	(0x042c  + rx->rxn_reg_stride * n)
 #define CDC_RX_RX0_RX_PATH_SEC4		(0x0430)
 #define CDC_RX_RX0_RX_PATH_SEC7		(0x0434)
-#define CDC_RX_RXn_RX_PATH_SEC7(rx, n)	(0x0434  + rx->rxn_reg_stride * n)
+#define CDC_RX_RXn_RX_PATH_SEC7(rx, n)		\
+	(0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0))
 #define CDC_RX_DSM_OUT_DELAY_SEL_MASK	GENMASK(2, 0)
 #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE	0x2
 #define CDC_RX_RX0_RX_PATH_MIX_SEC0	(0x0438)
 #define CDC_RX_RX0_RX_PATH_MIX_SEC1	(0x043C)
-#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)	(0x0440  + rx->rxn_reg_stride * n)
+#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)	\
+	(0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0))
 #define CDC_RX_RXn_DSM_CLK_EN_MASK	BIT(0)
 #define CDC_RX_RX0_RX_PATH_DSM_CTL	(0x0440)
 #define CDC_RX_RX0_RX_PATH_DSM_DATA1	(0x0444)
@@ -645,6 +647,7 @@ struct rx_macro {
 	int rx_mclk_cnt;
 	enum lpass_codec_version codec_version;
 	int rxn_reg_stride;
+	int rxn_reg_stride2;
 	bool is_ear_mode_on;
 	bool hph_pwr_mode;
 	bool hph_hd2_mode;
@@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 							      CDC_RX_PATH_PGA_MUTE_MASK, 0x0);
 			}
 
-			if (j == INTERP_AUX)
-				dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2);
-
 			int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8;
 			int_mux_cfg1 = int_mux_cfg0 + 4;
 			int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0);
@@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component,
 
 	main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx);
 	dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx);
-	if (interp_idx == INTERP_AUX)
-		dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2);
-
 	rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx);
 
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
@@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev)
 	case LPASS_CODEC_VERSION_2_0:
 	case LPASS_CODEC_VERSION_2_1:
 		rx->rxn_reg_stride = 0x80;
+		rx->rxn_reg_stride2 = 0xc;
 		def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults);
 		reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL);
 		if (!reg_defaults)
@@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev)
 	case LPASS_CODEC_VERSION_2_7:
 	case LPASS_CODEC_VERSION_2_8:
 		rx->rxn_reg_stride = 0xc0;
+		rx->rxn_reg_stride2 = 0x0;
 		def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults);
 		reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL);
 		if (!reg_defaults)

sound/soc/codecs/max98388.c

@@ -763,6 +763,7 @@ static int max98388_dai_tdm_slot(struct snd_soc_dai *dai,
 			addr = MAX98388_R2044_PCM_TX_CTRL1 + (cnt / 8);
 			bits = cnt % 8;
 			regmap_update_bits(max98388->regmap, addr, bits, bits);
+			slot_found++;
 			if (slot_found >= MAX_NUM_CH)
 				break;
 		}

sound/soc/codecs/pcm3060-i2c.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 I2C driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
+// Copyright (C) 2018 Kirill Marinushkin <k.marinushkin@gmail.com>
 
 #include <linux/i2c.h>
 #include <linux/module.h>
@@ -55,5 +55,5 @@ static struct i2c_driver pcm3060_i2c_driver = {
 module_i2c_driver(pcm3060_i2c_driver);
 
 MODULE_DESCRIPTION("PCM3060 I2C driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
+MODULE_AUTHOR("Kirill Marinushkin <k.marinushkin@gmail.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060-spi.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 SPI driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
+// Copyright (C) 2018 Kirill Marinushkin <k.marinushkin@gmail.com>
 
 #include <linux/module.h>
 #include <linux/spi/spi.h>
@@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = {
 module_spi_driver(pcm3060_spi_driver);
 
 MODULE_DESCRIPTION("PCM3060 SPI driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
+MODULE_AUTHOR("Kirill Marinushkin <k.marinushkin@gmail.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 codec driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
+// Copyright (C) 2018 Kirill Marinushkin <k.marinushkin@gmail.com>
 
 #include <linux/module.h>
 #include <sound/pcm_params.h>
@@ -343,5 +343,5 @@ int pcm3060_probe(struct device *dev)
 EXPORT_SYMBOL(pcm3060_probe);
 
 MODULE_DESCRIPTION("PCM3060 codec driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
+MODULE_AUTHOR("Kirill Marinushkin <k.marinushkin@gmail.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.h

@@ -2,7 +2,7 @@
 /*
  * PCM3060 codec driver
  *
- * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
+ * Copyright (C) 2018 Kirill Marinushkin <k.marinushkin@gmail.com>
  */
 
 #ifndef _SND_SOC_PCM3060_H

sound/soc/codecs/rt722-sdca-sdw.c

@@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave)
 	}
 
 	/* set the timeout values */
-	prop->clk_stop_timeout = 200;
+	prop->clk_stop_timeout = 900;
 
 	/* wake-up event */
 	prop->wake_capable = 1;

sound/soc/fsl/fsl_esai.c

@@ -119,10 +119,10 @@ static irqreturn_t esai_isr(int irq, void *devid)
 		dev_dbg(&pdev->dev, "isr: Transmission Initialized\n");
 
 	if (esr & ESAI_ESR_RFF_MASK)
-		dev_warn(&pdev->dev, "isr: Receiving overrun\n");
+		dev_dbg(&pdev->dev, "isr: Receiving overrun\n");
 
 	if (esr & ESAI_ESR_TFE_MASK)
-		dev_warn(&pdev->dev, "isr: Transmission underrun\n");
+		dev_dbg(&pdev->dev, "isr: Transmission underrun\n");
 
 	if (esr & ESAI_ESR_TLS_MASK)
 		dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n");

sound/soc/fsl/fsl_micfil.c

@@ -28,6 +28,13 @@
 
 #define MICFIL_OSR_DEFAULT	16
 
+#define MICFIL_NUM_RATES	7
+#define MICFIL_CLK_SRC_NUM	3
+/* clock source ids */
+#define MICFIL_AUDIO_PLL1	0
+#define MICFIL_AUDIO_PLL2	1
+#define MICFIL_CLK_EXT3		2
+
 enum quality {
 	QUALITY_HIGH,
 	QUALITY_MEDIUM,
@@ -45,9 +52,12 @@ struct fsl_micfil {
 	struct clk *mclk;
 	struct clk *pll8k_clk;
 	struct clk *pll11k_clk;
+	struct clk *clk_src[MICFIL_CLK_SRC_NUM];
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct sdma_peripheral_config sdmacfg;
 	struct snd_soc_card *card;
+	struct snd_pcm_hw_constraint_list constraint_rates;
+	unsigned int constraint_rates_list[MICFIL_NUM_RATES];
 	unsigned int dataline;
 	char name[32];
 	int irq[MICFIL_IRQ_LINES];
@@ -67,6 +77,7 @@ struct fsl_micfil_soc_data {
 	bool imx;
 	bool use_edma;
 	bool use_verid;
+	bool volume_sx;
 	u64  formats;
 };
 
@@ -76,6 +87,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = {
 	.fifo_depth = 8,
 	.dataline =  0xf,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.volume_sx = true,
 };
 
 static struct fsl_micfil_soc_data fsl_micfil_imx8mp = {
@@ -84,6 +96,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = {
 	.fifo_depth = 32,
 	.dataline =  0xf,
 	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.volume_sx = false,
 };
 
 static struct fsl_micfil_soc_data fsl_micfil_imx93 = {
@@ -94,6 +107,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = {
 	.formats = SNDRV_PCM_FMTBIT_S32_LE,
 	.use_edma = true,
 	.use_verid = true,
+	.volume_sx = false,
 };
 
 static const struct of_device_id fsl_micfil_dt_ids[] = {
@@ -317,7 +331,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = {
+static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = {
+	SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv),
+	SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL,
+		       MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv),
+};
+
+static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = {
 	SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL,
 			  MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL,
@@ -334,6 +367,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = {
 			  MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL,
 			  MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv),
+};
+
+static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = {
 	SOC_ENUM_EXT("MICFIL Quality Select",
 		     fsl_micfil_quality_enum,
 		     micfil_quality_get, micfil_quality_set),
@@ -449,12 +485,34 @@ static int fsl_micfil_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct fsl_micfil *micfil = snd_soc_dai_get_drvdata(dai);
+	unsigned int rates[MICFIL_NUM_RATES] = {8000, 11025, 16000, 22050, 32000, 44100, 48000};
+	int i, j, k = 0;
+	u64 clk_rate;
 
 	if (!micfil) {
 		dev_err(dai->dev, "micfil dai priv_data not set\n");
 		return -EINVAL;
 	}
 
+	micfil->constraint_rates.list = micfil->constraint_rates_list;
+	micfil->constraint_rates.count = 0;
+
+	for (j = 0; j < MICFIL_NUM_RATES; j++) {
+		for (i = 0; i < MICFIL_CLK_SRC_NUM; i++) {
+			clk_rate = clk_get_rate(micfil->clk_src[i]);
+			if (clk_rate != 0 && do_div(clk_rate, rates[j]) == 0) {
+				micfil->constraint_rates_list[k++] = rates[j];
+				micfil->constraint_rates.count++;
+				break;
+			}
+		}
+	}
+
+	if (micfil->constraint_rates.count > 0)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   &micfil->constraint_rates);
+
 	return 0;
 }
 
@@ -801,6 +859,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
+static int fsl_micfil_component_probe(struct snd_soc_component *component)
+{
+	struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component);
+
+	if (micfil->soc->volume_sx)
+		snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls,
+					       ARRAY_SIZE(fsl_micfil_volume_sx_controls));
+	else
+		snd_soc_add_component_controls(component, fsl_micfil_volume_controls,
+					       ARRAY_SIZE(fsl_micfil_volume_controls));
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops fsl_micfil_dai_ops = {
 	.probe		= fsl_micfil_dai_probe,
 	.startup	= fsl_micfil_startup,
@@ -821,6 +893,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = {
 
 static const struct snd_soc_component_driver fsl_micfil_component = {
 	.name		= "fsl-micfil-dai",
+	.probe		= fsl_micfil_component_probe,
 	.controls       = fsl_micfil_snd_controls,
 	.num_controls   = ARRAY_SIZE(fsl_micfil_snd_controls),
 	.legacy_dai_naming      = 1,
@@ -1134,6 +1207,12 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	fsl_asoc_get_pll_clocks(&pdev->dev, &micfil->pll8k_clk,
 				&micfil->pll11k_clk);
 
+	micfil->clk_src[MICFIL_AUDIO_PLL1] = micfil->pll8k_clk;
+	micfil->clk_src[MICFIL_AUDIO_PLL2] = micfil->pll11k_clk;
+	micfil->clk_src[MICFIL_CLK_EXT3] = devm_clk_get(&pdev->dev, "clkext3");
+	if (IS_ERR(micfil->clk_src[MICFIL_CLK_EXT3]))
+		micfil->clk_src[MICFIL_CLK_EXT3] = NULL;
+
 	/* init regmap */
 	regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res);
 	if (IS_ERR(regs))

sound/soc/intel/avs/core.c

@@ -28,6 +28,7 @@
 #include "avs.h"
 #include "cldma.h"
 #include "messages.h"
+#include "pcm.h"
 
 static u32 pgctl_mask = AZX_PGCTL_LSRMD_MASK;
 module_param(pgctl_mask, uint, 0444);
@@ -247,7 +248,7 @@ static void hdac_stream_update_pos(struct hdac_stream *stream, u64 buffer_size)
 static void hdac_update_stream(struct hdac_bus *bus, struct hdac_stream *stream)
 {
 	if (stream->substream) {
-		snd_pcm_period_elapsed(stream->substream);
+		avs_period_elapsed(stream->substream);
 	} else if (stream->cstream) {
 		u64 buffer_size = stream->cstream->runtime->buffer_size;
 

sound/soc/intel/avs/pcm.c

@@ -16,6 +16,7 @@
 #include <sound/soc-component.h>
 #include "avs.h"
 #include "path.h"
+#include "pcm.h"
 #include "topology.h"
 #include "../../codecs/hda.h"
 
@@ -30,6 +31,7 @@ struct avs_dma_data {
 		struct hdac_ext_stream *host_stream;
 	};
 
+	struct work_struct period_elapsed_work;
 	struct snd_pcm_substream *substream;
 };
 
@@ -56,6 +58,22 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction)
 	return dw->priv;
 }
 
+static void avs_period_elapsed_work(struct work_struct *work)
+{
+	struct avs_dma_data *data = container_of(work, struct avs_dma_data, period_elapsed_work);
+
+	snd_pcm_period_elapsed(data->substream);
+}
+
+void avs_period_elapsed(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+	struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(rtd, 0);
+	struct avs_dma_data *data = snd_soc_dai_get_dma_data(dai, substream);
+
+	schedule_work(&data->period_elapsed_work);
+}
+
 static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
@@ -77,6 +95,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
 	data->substream = substream;
 	data->template = template;
 	data->adev = adev;
+	INIT_WORK(&data->period_elapsed_work, avs_period_elapsed_work);
 	snd_soc_dai_set_dma_data(dai, substream, data);
 
 	if (rtd->dai_link->ignore_suspend)

sound/soc/intel/avs/pcm.h

@@ -0,0 +1,16 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2024 Intel Corporation
+ *
+ * Authors: Cezary Rojewski <cezary.rojewski@intel.com>
+ *          Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
+ */
+
+#ifndef __SOUND_SOC_INTEL_AVS_PCM_H
+#define __SOUND_SOC_INTEL_AVS_PCM_H
+
+#include <sound/pcm.h>
+
+void avs_period_elapsed(struct snd_pcm_substream *substream);
+
+#endif

sound/soc/intel/common/soc-acpi-intel-lnl-match.c

@@ -225,6 +225,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = {
 	}
 };
 
+static const struct snd_soc_acpi_adr_device rt1318_1_adr[] = {
+	{
+		.adr = 0x000133025D131801ull,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+		.name_prefix = "rt1318-1"
+	}
+};
+
 static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = {
 	{
 		.adr = 0x000130025D131801ull,
@@ -243,6 +252,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = {
 	}
 };
 
+static const struct snd_soc_acpi_adr_device rt713_0_adr[] = {
+	{
+		.adr = 0x000031025D071301ull,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+		.name_prefix = "rt713"
+	}
+};
+
 static const struct snd_soc_acpi_adr_device rt714_0_adr[] = {
 	{
 		.adr = 0x000030025D071401ull,
@@ -378,6 +396,20 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = {
 	{}
 };
 
+static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = {
+	{
+		.mask = BIT(0),
+		.num_adr = ARRAY_SIZE(rt713_0_adr),
+		.adr_d = rt713_0_adr,
+	},
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(rt1318_1_adr),
+		.adr_d = rt1318_1_adr,
+	},
+	{}
+};
+
 /* this table is used when there is no I2S codec present */
 struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
 	/* mockup tests need to be first */
@@ -447,6 +479,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg"
 	},
+	{
+		.link_mask = BIT(0) | BIT(1),
+		.links = lnl_sdw_rt713_l0_rt1318_l1,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg"
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines);

sound/soc/loongson/loongson_card.c

@@ -144,6 +144,7 @@ static int loongson_card_parse_of(struct loongson_card_data *data)
 			dev_err(dev, "getting cpu dlc error (%d)\n", ret);
 			goto err;
 		}
+		loongson_dai_links[i].platforms->of_node = loongson_dai_links[i].cpus->of_node;
 
 		ret = snd_soc_of_get_dlc(codec, NULL, loongson_dai_links[i].codecs, 0);
 		if (ret < 0) {

sound/soc/qcom/Kconfig

@@ -157,6 +157,7 @@ config SND_SOC_SDM845
 	depends on COMMON_CLK
 	select SND_SOC_QDSP6
 	select SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_SDW
 	select SND_SOC_RT5663
 	select SND_SOC_MAX98927
 	imply SND_SOC_CROS_EC_CODEC
@@ -208,6 +209,7 @@ config SND_SOC_SC7280
 	tristate "SoC Machine driver for SC7280 boards"
 	depends on I2C && SOUNDWIRE
 	select SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_SDW
 	select SND_SOC_LPASS_SC7280
 	select SND_SOC_MAX98357A
 	select SND_SOC_WCD938X_SDW

sound/soc/qcom/lpass-cpu.c

@@ -1242,6 +1242,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
 	/* Allocation for i2sctl regmap fields */
 	drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl),
 					GFP_KERNEL);
+	if (!drvdata->i2sctl)
+		return -ENOMEM;
 
 	/* Initialize bitfields for dai I2SCTL register */
 	ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl,

sound/soc/qcom/sc7280.c

@@ -23,6 +23,7 @@
 #include "common.h"
 #include "lpass.h"
 #include "qdsp6/q6afe.h"
+#include "sdw.h"
 
 #define DEFAULT_MCLK_RATE              19200000
 #define RT5682_PLL_FREQ (48000 * 512)
@@ -316,6 +317,7 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream)
 	struct snd_soc_card *card = rtd->card;
 	struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card);
 	struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+	struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
 
 	switch (cpu_dai->id) {
 	case MI2S_PRIMARY:
@@ -333,6 +335,9 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream)
 	default:
 		break;
 	}
+
+	data->sruntime[cpu_dai->id] = NULL;
+	sdw_release_stream(sruntime);
 }
 
 static int sc7280_snd_startup(struct snd_pcm_substream *substream)
@@ -347,6 +352,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream)
 	switch (cpu_dai->id) {
 	case MI2S_PRIMARY:
 		ret = sc7280_rt5682_init(rtd);
+		if (ret)
+			return ret;
 		break;
 	case SECONDARY_MI2S_RX:
 		codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S;
@@ -360,7 +367,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream)
 	default:
 		break;
 	}
-	return ret;
+
+	return qcom_snd_sdw_startup(substream);
 }
 
 static const struct snd_soc_ops sc7280_ops = {

sound/soc/qcom/sdm845.c

@@ -15,6 +15,7 @@
 #include <uapi/linux/input-event-codes.h>
 #include "common.h"
 #include "qdsp6/q6afe.h"
+#include "sdw.h"
 #include "../codecs/rt5663.h"
 
 #define DRIVER_NAME	"sdm845"
@@ -416,7 +417,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
 		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
 		break;
 	}
-	return 0;
+	return qcom_snd_sdw_startup(substream);
 }
 
 static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
@@ -425,6 +426,7 @@ static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
 	struct snd_soc_card *card = rtd->card;
 	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
 	struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+	struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
 
 	switch (cpu_dai->id) {
 	case PRIMARY_MI2S_RX:
@@ -463,6 +465,9 @@ static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
 		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
 		break;
 	}
+
+	data->sruntime[cpu_dai->id] = NULL;
+	sdw_release_stream(sruntime);
 }
 
 static int sdm845_snd_prepare(struct snd_pcm_substream *substream)

sound/soc/sh/rcar/core.c

@@ -1281,7 +1281,9 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph)
 		if (!of_node_name_eq(ports, "ports") &&
 		    !of_node_name_eq(ports, "port"))
 			continue;
-		priv->component_dais[i] = of_graph_get_endpoint_count(ports);
+		priv->component_dais[i] =
+			of_graph_get_endpoint_count(of_node_name_eq(ports, "ports") ?
+						    ports : np);
 		nr += priv->component_dais[i];
 		i++;
 		if (i >= RSND_MAX_COMPONENT) {
@@ -1493,7 +1495,8 @@ static int rsnd_dai_probe(struct rsnd_priv *priv)
 			if (!of_node_name_eq(ports, "ports") &&
 			    !of_node_name_eq(ports, "port"))
 				continue;
-			for_each_endpoint_of_node(ports, dai_np) {
+			for_each_endpoint_of_node(of_node_name_eq(ports, "ports") ?
+						  ports : np, dai_np) {
 				__rsnd_dai_probe(priv, dai_np, dai_np, 0, dai_i);
 				if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) {
 					rdai = rsnd_rdai_get(priv, dai_i);

sound/soc/soc-dapm.c

@@ -2785,10 +2785,10 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai);
 
 int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s)
 {
-	struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm);
+	struct snd_soc_component *component = widget->dapm->component;
 	const char *wname = widget->name;
 
-	if (component->name_prefix)
+	if (component && component->name_prefix)
 		wname += strlen(component->name_prefix) + 1; /* plus space */
 
 	return strcmp(wname, s);

sound/soc/sof/amd/acp-loader.c

@@ -206,7 +206,10 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev)
 		configure_pte_for_fw_loading(FW_SRAM_DATA_BIN, ACP_SRAM_PAGE_COUNT, adata);
 		src_addr = ACP_SYSTEM_MEMORY_WINDOW + ACP_DEFAULT_SRAM_LENGTH +
 			   (page_count * ACP_PAGE_SIZE);
-		dest_addr = ACP_SRAM_BASE_ADDRESS;
+		if (adata->pci_rev > ACP63_PCI_ID)
+			dest_addr = ACP7X_SRAM_BASE_ADDRESS;
+		else
+			dest_addr = ACP_SRAM_BASE_ADDRESS;
 
 		ret = configure_and_run_dma(adata, src_addr, dest_addr,
 					    adata->fw_sram_data_bin_size);

sound/soc/sof/amd/acp.c

@@ -329,7 +329,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr,
 					    fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE,
 					    ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US);
 	if (ret < 0) {
-		dev_err(sdev->dev, "PSP validation failed\n");
+		val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SHA_PSP_ACK);
+		dev_err(sdev->dev, "PSP validation failed: fw_qualifier = %#x, ACP_SHA_PSP_ACK = %#x\n",
+			fw_qualifier, val);
 		return ret;
 	}
 

sound/soc/sof/intel/hda-dai-ops.c

@@ -346,20 +346,21 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_hdac_ext_stream_start(hext_stream);
 		break;
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		snd_hdac_ext_stream_clear(hext_stream);
-
 		/*
-		 * Save the LLP registers in case the stream is
+		 * Save the LLP registers since in case of PAUSE the LLP
-		 * restarting due PAUSE_RELEASE, or START without a pcm
+		 * register are not reset to 0, the delay calculation will use
-		 * close/open since in this case the LLP register is not reset
+		 * the saved offsets for compensating the delay calculation.
-		 * to 0 and the delay calculation will return with invalid
-		 * results.
 		 */
 		hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
 		hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
+		snd_hdac_ext_stream_clear(hext_stream);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+		hext_stream->pplcllpl = 0;
+		hext_stream->pplcllpu = 0;
+		snd_hdac_ext_stream_clear(hext_stream);
 		break;
 	default:
 		dev_err(sdev->dev, "unknown trigger command %d\n", cmd);
@@ -512,7 +513,6 @@ static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = {
 static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 				 struct snd_pcm_substream *substream, int cmd)
 {
-	struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream);
 	struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
 
 	switch (cmd) {
@@ -527,9 +527,6 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c
 		if (ret < 0)
 			return ret;
 
-		if (cmd == SNDRV_PCM_TRIGGER_STOP)
-			return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
-
 		break;
 	}
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:

sound/soc/sof/intel/hda-dai.c

@@ -302,6 +302,7 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i
 	}
 
 	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ret = hda_link_dma_cleanup(substream, hext_stream, dai);
 		if (ret < 0) {
@@ -370,6 +371,13 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	sdev = widget_to_sdev(w);
+	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
+
+	/* nothing more to do if the link is already prepared */
+	if (hext_stream && hext_stream->link_prepared)
+		return 0;
+
 	/* use HDaudio stream handling */
 	ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags);
 	if (ret < 0) {
@@ -377,7 +385,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	sdev = widget_to_sdev(w);
 	if (sdev->dspless_mode_selected)
 		return 0;
 
@@ -482,6 +489,31 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 	int i;
 
+	ops = hda_dai_get_ops(substream, cpu_dai);
+	if (!ops) {
+		dev_err(cpu_dai->dev, "DAI widget ops not set\n");
+		return -EINVAL;
+	}
+
+	sdev = widget_to_sdev(w);
+	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
+
+	/* nothing more to do if the link is already prepared */
+	if (hext_stream && hext_stream->link_prepared)
+		return 0;
+
+	/*
+	 * reset the PCMSyCM registers to handle a prepare callback when the PCM is restarted
+	 * due to xruns or after a call to snd_pcm_drain/drop()
+	 */
+	ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id,
+					     0, 0, substream->stream);
+	if (ret < 0) {
+		dev_err(cpu_dai->dev, "%s:  hdac_bus_eml_sdw_map_stream_ch failed %d\n",
+			__func__, ret);
+		return ret;
+	}
+
 	data.dai_index = (link_id << 8) | cpu_dai->id;
 	data.dai_node_id = intel_alh_id;
 	ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags);
@@ -490,10 +522,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	ops = hda_dai_get_ops(substream, cpu_dai);
-	sdev = widget_to_sdev(w);
 	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
-
 	if (!hext_stream)
 		return -ENODEV;
 

sound/soc/sof/intel/hda-loader.c

@@ -294,14 +294,9 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
 {
 	struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
 	const struct sof_intel_dsp_desc *chip = hda->desc;
-	struct sof_intel_hda_stream *hda_stream;
-	unsigned long time_left;
 	unsigned int reg;
 	int ret, status;
 
-	hda_stream = container_of(hext_stream, struct sof_intel_hda_stream,
-				  hext_stream);
-
 	dev_dbg(sdev->dev, "Code loader DMA starting\n");
 
 	ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START);
@@ -310,18 +305,6 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
 		return ret;
 	}
 
-	if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) {
-		/* Wait for completion of transfer */
-		time_left = wait_for_completion_timeout(&hda_stream->ioc,
-							msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
-
-		if (!time_left) {
-			dev_err(sdev->dev, "Code loader DMA did not complete\n");
-			return -ETIMEDOUT;
-		}
-		dev_dbg(sdev->dev, "Code loader DMA done\n");
-	}
-
 	dev_dbg(sdev->dev, "waiting for FW_ENTERED status\n");
 
 	status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,

sound/soc/sof/ipc4-topology.c

@@ -3129,9 +3129,20 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		 * group_id during copier's ipc_prepare op.
 		 */
 		if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+			struct sof_ipc4_alh_configuration_blob *blob;
+
+			blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config;
 			ipc4_copier->dai_index = data->dai_node_id;
-			copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+
-			copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id);
+			/*
+			 * no need to set the node_id for aggregated DAI's. These will be assigned
+			 * a group_id during widget ipc_prepare
+			 */
+			if (blob->alh_cfg.device_count == 1) {
+				copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+				copier_data->gtw_cfg.node_id |=
+					SOF_IPC4_NODE_INDEX(data->dai_node_id);
+			}
 		}
 
 		break;