Merge tag 'asoc-fix-v6.4-rc2' of...

Merge tag 'asoc-fix-v6.4-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.4

More fixes that came in since the merge window, the bulk of which are
for the SOF code, I suspect as a result of the wide usage, active
development and large code size rather than huge quality problems.

There's also a couple of MAINTAINERS updates and some new device quirks.

Documentation/devicetree/bindings/sound/tas2562.yaml

@@ -55,7 +55,9 @@ properties:
     description: TDM TX current sense time slot.
 
   '#sound-dai-cells':
-    const: 1
+    # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+    # compatibility but is deprecated.
+    enum: [0, 1]
 
 required:
   - compatible
@@ -72,7 +74,7 @@ examples:
      codec: codec@4c {
        compatible = "ti,tas2562";
        reg = <0x4c>;
-       #sound-dai-cells = <1>;
+       #sound-dai-cells = <0>;
        interrupt-parent = <&gpio1>;
        interrupts = <14>;
        shutdown-gpios = <&gpio1 15 0>;

Documentation/devicetree/bindings/sound/tas2770.yaml

@@ -57,7 +57,9 @@ properties:
       - 1 # Falling edge
 
   '#sound-dai-cells':
-    const: 1
+    # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+    # compatibility but is deprecated.
+    enum: [0, 1]
 
 required:
   - compatible
@@ -74,7 +76,7 @@ examples:
      codec: codec@41 {
        compatible = "ti,tas2770";
        reg = <0x41>;
-       #sound-dai-cells = <1>;
+       #sound-dai-cells = <0>;
        interrupt-parent = <&gpio1>;
        interrupts = <14>;
        reset-gpio = <&gpio1 15 0>;

Documentation/devicetree/bindings/sound/tas27xx.yaml

@@ -50,7 +50,9 @@ properties:
     description: TDM TX voltage sense time slot.
 
   '#sound-dai-cells':
-    const: 1
+    # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+    # compatibility but is deprecated.
+    enum: [0, 1]
 
 required:
   - compatible
@@ -67,7 +69,7 @@ examples:
      codec: codec@38 {
        compatible = "ti,tas2764";
        reg = <0x38>;
-       #sound-dai-cells = <1>;
+       #sound-dai-cells = <0>;
        interrupt-parent = <&gpio1>;
        interrupts = <14>;
        reset-gpios = <&gpio1 15 0>;

MAINTAINERS

@@ -4921,7 +4921,6 @@ F:	drivers/media/cec/i2c/ch7322.c
 CIRRUS LOGIC AUDIO CODEC DRIVERS
 M:	James Schulman <james.schulman@cirrus.com>
 M:	David Rhodes <david.rhodes@cirrus.com>
-M:	Lucas Tanure <tanureal@opensource.cirrus.com>
 M:	Richard Fitzgerald <rf@opensource.cirrus.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	patches@opensource.cirrus.com
@@ -18568,10 +18567,9 @@ F:	Documentation/admin-guide/LSM/SafeSetID.rst
 F:	security/safesetid/
 
 SAMSUNG AUDIO (ASoC) DRIVERS
-M:	Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
 M:	Sylwester Nawrocki <s.nawrocki@samsung.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
-S:	Supported
+S:	Maintained
 B:	mailto:linux-samsung-soc@vger.kernel.org
 F:	Documentation/devicetree/bindings/sound/samsung*
 F:	sound/soc/samsung/

include/sound/hda-mlink.h

@@ -44,6 +44,9 @@ int hdac_bus_eml_sdw_power_down_unlocked(struct hdac_bus *bus, int sublink);
 
 int hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num);
 
+int hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
+				   int channel_mask, int stream_id, int dir);
+
 void hda_bus_ml_put_all(struct hdac_bus *bus);
 void hda_bus_ml_reset_losidv(struct hdac_bus *bus);
 int hda_bus_ml_resume(struct hdac_bus *bus);
@@ -51,6 +54,7 @@ int hda_bus_ml_suspend(struct hdac_bus *bus);
 
 struct hdac_ext_link *hdac_bus_eml_ssp_get_hlink(struct hdac_bus *bus);
 struct hdac_ext_link *hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus);
+struct hdac_ext_link *hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus);
 
 struct mutex *hdac_bus_eml_get_mutex(struct hdac_bus *bus, bool alt, int elid);
 
@@ -144,6 +148,13 @@ hdac_bus_eml_sdw_power_down_unlocked(struct hdac_bus *bus, int sublink) { return
 static inline int
 hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num) { return 0; }
 
+static inline int
+hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
+			       int channel_mask, int stream_id, int dir)
+{
+	return 0;
+}
+
 static inline void hda_bus_ml_put_all(struct hdac_bus *bus) { }
 static inline void hda_bus_ml_reset_losidv(struct hdac_bus *bus) { }
 static inline int hda_bus_ml_resume(struct hdac_bus *bus) { return 0; }
@@ -155,6 +166,9 @@ hdac_bus_eml_ssp_get_hlink(struct hdac_bus *bus) { return NULL; }
 static inline struct hdac_ext_link *
 hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus) { return NULL; }
 
+static inline struct hdac_ext_link *
+hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus) { return NULL; }
+
 static inline struct mutex *
 hdac_bus_eml_get_mutex(struct hdac_bus *bus, bool alt, int elid) { return NULL; }
 

include/uapi/sound/sof/tokens.h

@@ -183,7 +183,7 @@
 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_INTERLEAVING_STYLE	1906
 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG	1907
 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_SAMPLE_TYPE	1908
-#define SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX		1909
+#define SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX	1909
 /* intentional token numbering discontinuity, reserved for future use */
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_RATE	1930
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_BIT_DEPTH	1931
@@ -194,6 +194,7 @@
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_INTERLEAVING_STYLE	1936
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG	1937
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_SAMPLE_TYPE	1938
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX	1939
 /* intentional token numbering discontinuity, reserved for future use */
 #define SOF_TKN_CAVS_AUDIO_FORMAT_IBS		1970
 #define SOF_TKN_CAVS_AUDIO_FORMAT_OBS		1971

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

@@ -311,6 +311,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_BOARD_NAME, "8A22"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "System76"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"),
+		}
+	},
 	{}
 };
 

sound/soc/codecs/cs35l56.c

@@ -852,10 +852,11 @@ static void cs35l56_dsp_work(struct work_struct *work)
 	 */
 	if (cs35l56->sdw_peripheral) {
 		cs35l56->sdw_irq_no_unmask = true;
-		cancel_work_sync(&cs35l56->sdw_irq_work);
+		flush_work(&cs35l56->sdw_irq_work);
 		sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0);
 		sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1);
 		sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF);
+		flush_work(&cs35l56->sdw_irq_work);
 	}
 
 	ret = cs35l56_mbox_send(cs35l56, CS35L56_MBOX_CMD_SHUTDOWN);

sound/soc/codecs/ssm2602.c

@@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = {
 	{ .reg = 0x09, .def = 0x0000 }
 };
 
+/*
+ * ssm2602 register patch
+ * Workaround for playback distortions after power up: activates digital
+ * core, and then powers on output, DAC, and whole chip at the same time
+ */
+
+static const struct reg_sequence ssm2602_patch[] = {
+	{ SSM2602_ACTIVE, 0x01 },
+	{ SSM2602_PWR,    0x07 },
+	{ SSM2602_RESET,  0x00 },
+};
+
 
 /*Appending several "None"s just for OSS mixer use*/
 static const char *ssm2602_input_select[] = {
@@ -598,6 +610,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component)
 		return ret;
 	}
 
+	regmap_register_patch(ssm2602->regmap, ssm2602_patch,
+			      ARRAY_SIZE(ssm2602_patch));
+
 	/* set the update bits */
 	regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
 			    LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);

sound/soc/dwc/dwc-i2s.c

@@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
 
 		/* Error Handling: TX */
 		if (isr[i] & ISR_TXFO) {
-			dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
+			dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i);
 			irq_valid = true;
 		}
 
 		/* Error Handling: TX */
 		if (isr[i] & ISR_RXFO) {
-			dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
+			dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i);
 			irq_valid = true;
 		}
 	}

sound/soc/fsl/fsl_micfil.c

@@ -1159,7 +1159,7 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to pcm register\n");
-		return ret;
+		goto err_pm_disable;
 	}
 
 	fsl_micfil_dai.capture.formats = micfil->soc->formats;
@@ -1169,9 +1169,20 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register component %s\n",
 			fsl_micfil_component.name);
+		goto err_pm_disable;
 	}
 
 	return ret;
+
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+
+	return ret;
+}
+
+static void fsl_micfil_remove(struct platform_device *pdev)
+{
+	pm_runtime_disable(&pdev->dev);
 }
 
 static int __maybe_unused fsl_micfil_runtime_suspend(struct device *dev)
@@ -1232,6 +1243,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = {
 
 static struct platform_driver fsl_micfil_driver = {
 	.probe = fsl_micfil_probe,
+	.remove_new = fsl_micfil_remove,
 	.driver = {
 		.name = "fsl-micfil-dai",
 		.pm = &fsl_micfil_pm_ops,

sound/soc/jz4740/jz4740-i2s.c

@@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate,
+				 unsigned long i2sdiv_max)
+{
+	unsigned long div, rate1, rate2, err1, err2;
+
+	div = mclk / (64 * rate);
+	if (div == 0)
+		div = 1;
+
+	rate1 = mclk / (64 * div);
+	rate2 = mclk / (64 * (div + 1));
+
+	err1 = abs(rate1 - rate);
+	err2 = abs(rate2 - rate);
+
+	/*
+	 * Choose the divider that produces the smallest error in the
+	 * output rate and reject dividers with a 5% or higher error.
+	 * In the event that both dividers are outside the acceptable
+	 * error margin, reject the rate to prevent distorted audio.
+	 * (The number 5% is arbitrary.)
+	 */
+	if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20)
+		return div;
+	if (div < i2sdiv_max && err2 < rate/20)
+		return div + 1;
+
+	return -EINVAL;
+}
+
 static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	struct regmap_field *div_field;
+	unsigned long i2sdiv_max;
 	unsigned int sample_size;
-	uint32_t ctrl;
-	int div;
+	uint32_t ctrl, conf;
+	int div = 1;
 
 	regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl);
-
-	div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
+	regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
@@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 			ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
 
 		div_field = i2s->field_i2sdiv_playback;
+		i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb,
+				     i2s->soc_info->field_i2sdiv_playback.lsb);
 	} else {
 		ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE;
 		ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size);
 
 		div_field = i2s->field_i2sdiv_capture;
+		i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb,
+				     i2s->soc_info->field_i2sdiv_capture.lsb);
+	}
+
+	/*
+	 * Only calculate I2SDIV if we're supplying the bit or frame clock.
+	 * If the codec is supplying both clocks then the divider output is
+	 * unused, and we don't want it to limit the allowed sample rates.
+	 */
+	if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) {
+		div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s),
+					    params_rate(params), i2sdiv_max);
+		if (div < 0)
+			return div;
 	}
 
 	regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl);

sound/soc/mediatek/mt8186/mt8186-afe-clk.c

@@ -644,9 +644,3 @@ int mt8186_init_clock(struct mtk_base_afe *afe)
 
 	return 0;
 }
-
-void mt8186_deinit_clock(void *priv)
-{
-	struct mtk_base_afe *afe = priv;
-	mt8186_audsys_clk_unregister(afe);
-}

sound/soc/mediatek/mt8186/mt8186-afe-clk.h

@@ -81,7 +81,6 @@ enum {
 struct mtk_base_afe;
 int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id);
 int mt8186_init_clock(struct mtk_base_afe *afe);
-void mt8186_deinit_clock(void *priv);
 int mt8186_afe_enable_cgs(struct mtk_base_afe *afe);
 void mt8186_afe_disable_cgs(struct mtk_base_afe *afe);
 int mt8186_afe_enable_clock(struct mtk_base_afe *afe);

sound/soc/mediatek/mt8186/mt8186-afe-pcm.c

@@ -2848,10 +2848,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = devm_add_action_or_reset(dev, mt8186_deinit_clock, (void *)afe);
-	if (ret)
-		return ret;
-
 	/* init memif */
 	afe->memif_32bit_supported = 0;
 	afe->memif_size = MT8186_MEMIF_NUM;

sound/soc/mediatek/mt8186/mt8186-audsys-clk.c

@@ -84,6 +84,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
 	GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24),
 };
 
+static void mt8186_audsys_clk_unregister(void *data)
+{
+	struct mtk_base_afe *afe = data;
+	struct mt8186_afe_private *afe_priv = afe->platform_priv;
+	struct clk *clk;
+	struct clk_lookup *cl;
+	int i;
+
+	if (!afe_priv)
+		return;
+
+	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+		cl = afe_priv->lookup[i];
+		if (!cl)
+			continue;
+
+		clk = cl->clk;
+		clk_unregister_gate(clk);
+
+		clkdev_drop(cl);
+	}
+}
+
 int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
 {
 	struct mt8186_afe_private *afe_priv = afe->platform_priv;
@@ -124,27 +147,6 @@ int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
 		afe_priv->lookup[i] = cl;
 	}
 
-	return 0;
+	return devm_add_action_or_reset(afe->dev, mt8186_audsys_clk_unregister, afe);
 }
 
-void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
-	struct mt8186_afe_private *afe_priv = afe->platform_priv;
-	struct clk *clk;
-	struct clk_lookup *cl;
-	int i;
-
-	if (!afe_priv)
-		return;
-
-	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
-		cl = afe_priv->lookup[i];
-		if (!cl)
-			continue;
-
-		clk = cl->clk;
-		clk_unregister_gate(clk);
-
-		clkdev_drop(cl);
-	}
-}

sound/soc/mediatek/mt8186/mt8186-audsys-clk.h

@@ -10,6 +10,5 @@
 #define _MT8186_AUDSYS_CLK_H_
 
 int mt8186_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe);
 
 #endif

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

@@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp
 		acp_mailbox_read(sdev, offset, p, sz);
 	} else {
 		struct snd_pcm_substream *substream = sps->substream;
-		struct acp_dsp_stream *stream = substream->runtime->private_data;
+		struct acp_dsp_stream *stream;
+
+		if (!substream || !substream->runtime)
+			return -ESTRPIPE;
+
+		stream = substream->runtime->private_data;
 
 		if (!stream)
 			return -ESTRPIPE;

sound/soc/sof/debug.c

@@ -438,8 +438,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg)
 		/* should we prevent DSP entering D3 ? */
 		if (!sdev->ipc_dump_printed)
 			dev_info(sdev->dev,
-				 "preventing DSP entering D3 state to preserve context\n");
-		pm_runtime_get_noresume(sdev->dev);
+				 "Attempting to prevent DSP from entering D3 state to preserve context\n");
+		pm_runtime_get_if_in_use(sdev->dev);
 	}
 
 	/* dump vital information to the logs */

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

@@ -19,6 +19,9 @@
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_MLINK)
 
+/* worst-case number of sublinks is used for sublink refcount array allocation only */
+#define HDAML_MAX_SUBLINKS (AZX_ML_LCTL_CPA_SHIFT - AZX_ML_LCTL_SPA_SHIFT)
+
 /**
  * struct hdac_ext2_link - HDAudio extended+alternate link
  *
@@ -33,6 +36,7 @@
  * @leptr:		extended link pointer
  * @eml_lock:		mutual exclusion to access shared registers e.g. CPA/SPA bits
  * in LCTL register
+ * @sublink_ref_count:	array of refcounts, required to power-manage sublinks independently
  * @base_ptr:		pointer to shim/ip/shim_vs space
  * @instance_offset:	offset between each of @slcount instances managed by link
  * @shim_offset:	offset to SHIM register base
@@ -53,6 +57,7 @@ struct hdac_ext2_link {
 	u32 leptr;
 
 	struct mutex eml_lock; /* prevent concurrent access to e.g. CPA/SPA */
+	int sublink_ref_count[HDAML_MAX_SUBLINKS];
 
 	/* internal values computed from LCAP contents */
 	void __iomem *base_ptr;
@@ -68,6 +73,7 @@ struct hdac_ext2_link {
 #define AZX_REG_SDW_SHIM_OFFSET				0x0
 #define AZX_REG_SDW_IP_OFFSET				0x100
 #define AZX_REG_SDW_VS_SHIM_OFFSET			0x6000
+#define AZX_REG_SDW_SHIM_PCMSyCM(y)			(0x16 + 0x4 * (y))
 
 /* only one instance supported */
 #define AZX_REG_INTEL_DMIC_SHIM_OFFSET			0x0
@@ -91,7 +97,7 @@ struct hdac_ext2_link {
  */
 
 static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
-			  void __iomem *ml_addr, int link_idx)
+			  void __iomem *remap_addr, void __iomem *ml_addr, int link_idx)
 {
 	struct hdac_ext_link *hlink = &h2link->hext_link;
 	u32 base_offset;
@@ -126,15 +132,16 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
 		link_idx, h2link->slcount);
 
 	/* find IP ID and offsets */
-	h2link->leptr = readl(hlink->ml_addr + AZX_REG_ML_LEPTR);
+	h2link->leptr = readl(ml_addr + AZX_REG_ML_LEPTR);
 
 	h2link->elid = FIELD_GET(AZX_REG_ML_LEPTR_ID, h2link->leptr);
 
 	base_offset = FIELD_GET(AZX_REG_ML_LEPTR_PTR, h2link->leptr);
-	h2link->base_ptr = hlink->ml_addr + base_offset;
+	h2link->base_ptr = remap_addr + base_offset;
 
 	switch (h2link->elid) {
 	case AZX_REG_ML_LEPTR_ID_SDW:
+		h2link->instance_offset = AZX_REG_SDW_INSTANCE_OFFSET;
 		h2link->shim_offset = AZX_REG_SDW_SHIM_OFFSET;
 		h2link->ip_offset = AZX_REG_SDW_IP_OFFSET;
 		h2link->shim_vs_offset = AZX_REG_SDW_VS_SHIM_OFFSET;
@@ -149,6 +156,7 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
 			link_idx, base_offset);
 		break;
 	case AZX_REG_ML_LEPTR_ID_INTEL_SSP:
+		h2link->instance_offset = AZX_REG_INTEL_SSP_INSTANCE_OFFSET;
 		h2link->shim_offset = AZX_REG_INTEL_SSP_SHIM_OFFSET;
 		h2link->ip_offset = AZX_REG_INTEL_SSP_IP_OFFSET;
 		h2link->shim_vs_offset = AZX_REG_INTEL_SSP_VS_SHIM_OFFSET;
@@ -333,6 +341,21 @@ static void hdaml_link_set_lsdiid(u32 __iomem *lsdiid, int dev_num)
 	writel(val, lsdiid);
 }
 
+static void hdaml_shim_map_stream_ch(u16 __iomem *pcmsycm, int lchan, int hchan,
+				     int stream_id, int dir)
+{
+	u16 val;
+
+	val = readw(pcmsycm);
+
+	u16p_replace_bits(&val, lchan, GENMASK(3, 0));
+	u16p_replace_bits(&val, hchan, GENMASK(7, 4));
+	u16p_replace_bits(&val, stream_id, GENMASK(13, 8));
+	u16p_replace_bits(&val, dir, BIT(15));
+
+	writew(val, pcmsycm);
+}
+
 static void hdaml_lctl_offload_enable(u32 __iomem *lctl, bool enable)
 {
 	u32 val = readl(lctl);
@@ -364,7 +387,7 @@ static int hda_ml_alloc_h2link(struct hdac_bus *bus, int index)
 	hlink->bus = bus;
 	hlink->ml_addr = bus->mlcap + AZX_ML_BASE + (AZX_ML_INTERVAL * index);
 
-	ret = hdaml_lnk_enum(bus->dev, h2link, hlink->ml_addr, index);
+	ret = hdaml_lnk_enum(bus->dev, h2link, bus->remap_addr, hlink->ml_addr, index);
 	if (ret < 0) {
 		kfree(h2link);
 		return ret;
@@ -641,8 +664,13 @@ static int hdac_bus_eml_power_up_base(struct hdac_bus *bus, bool alt, int elid,
 	if (eml_lock)
 		mutex_lock(&h2link->eml_lock);
 
+	if (!alt) {
 		if (++hlink->ref_count > 1)
 			goto skip_init;
+	} else {
+		if (++h2link->sublink_ref_count[sublink] > 1)
+			goto skip_init;
+	}
 
 	ret = hdaml_link_init(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
 
@@ -684,9 +712,13 @@ static int hdac_bus_eml_power_down_base(struct hdac_bus *bus, bool alt, int elid
 	if (eml_lock)
 		mutex_lock(&h2link->eml_lock);
 
+	if (!alt) {
 		if (--hlink->ref_count > 0)
 			goto skip_shutdown;
-
+	} else {
+		if (--h2link->sublink_ref_count[sublink] > 0)
+			goto skip_shutdown;
+	}
 	ret = hdaml_link_shutdown(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
 
 skip_shutdown:
@@ -740,6 +772,40 @@ int hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num)
 	return 0;
 } EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_set_lsdiid, SND_SOC_SOF_HDA_MLINK);
 
+/*
+ * the 'y' parameter comes from the PCMSyCM hardware register naming. 'y' refers to the
+ * PDI index, i.e. the FIFO used for RX or TX
+ */
+int hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
+				   int channel_mask, int stream_id, int dir)
+{
+	struct hdac_ext2_link *h2link;
+	u16 __iomem *pcmsycm;
+	u16 val;
+
+	h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
+	if (!h2link)
+		return -ENODEV;
+
+	pcmsycm = h2link->base_ptr + h2link->shim_offset +
+		h2link->instance_offset * sublink +
+		AZX_REG_SDW_SHIM_PCMSyCM(y);
+
+	mutex_lock(&h2link->eml_lock);
+
+	hdaml_shim_map_stream_ch(pcmsycm, 0, hweight32(channel_mask),
+				 stream_id, dir);
+
+	mutex_unlock(&h2link->eml_lock);
+
+	val = readw(pcmsycm);
+
+	dev_dbg(bus->dev, "channel_mask %#x stream_id %d dir %d pcmscm %#x\n",
+		channel_mask, stream_id, dir, val);
+
+	return 0;
+} EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_map_stream_ch, SND_SOC_SOF_HDA_MLINK);
+
 void hda_bus_ml_put_all(struct hdac_bus *bus)
 {
 	struct hdac_ext_link *hlink;
@@ -836,6 +902,18 @@ struct hdac_ext_link *hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_NS(hdac_bus_eml_dmic_get_hlink, SND_SOC_SOF_HDA_MLINK);
 
+struct hdac_ext_link *hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus)
+{
+	struct hdac_ext2_link *h2link;
+
+	h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
+	if (!h2link)
+		return NULL;
+
+	return &h2link->hext_link;
+}
+EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_get_hlink, SND_SOC_SOF_HDA_MLINK);
+
 int hdac_bus_eml_enable_offload(struct hdac_bus *bus, bool alt, int elid, bool enable)
 {
 	struct hdac_ext2_link *h2link;

sound/soc/sof/ipc3-topology.c

@@ -2103,10 +2103,13 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 	 * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is.
 	 */
 
-	if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+	if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+		/* Clear stale command */
+		config->flags &= ~SOF_DAI_CONFIG_FLAGS_CMD_MASK;
 		config->flags |= flags;
-	else
+	} else {
 		config->flags = flags;
+	}
 
 	/* only send the IPC if the widget is set up in the DSP */
 	if (swidget->use_count > 0) {

sound/soc/sof/ipc4-topology.c

@@ -59,7 +59,7 @@ static const struct sof_topology_token ipc4_in_audio_format_tokens[] = {
 		audio_fmt.interleaving_style)},
 	{SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
-	{SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
+	{SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, pin_index)},
 	{SOF_TKN_CAVS_AUDIO_FORMAT_IBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, buffer_size)},
@@ -79,7 +79,7 @@ static const struct sof_topology_token ipc4_out_audio_format_tokens[] = {
 		audio_fmt.interleaving_style)},
 	{SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
-	{SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
+	{SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, pin_index)},
 	{SOF_TKN_CAVS_AUDIO_FORMAT_OBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
 		offsetof(struct sof_ipc4_pin_format, buffer_size)},

sound/soc/sof/pcm.c

@@ -643,16 +643,17 @@ static int sof_pcm_probe(struct snd_soc_component *component)
 				       "%s/%s",
 				       plat_data->tplg_filename_prefix,
 				       plat_data->tplg_filename);
-	if (!tplg_filename)
-		return -ENOMEM;
+	if (!tplg_filename) {
+		ret = -ENOMEM;
+		goto pm_error;
+	}
 
 	ret = snd_sof_load_topology(component, tplg_filename);
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(component->dev, "error: failed to load DSP topology %d\n",
 			ret);
-		return ret;
-	}
 
+pm_error:
 	pm_runtime_mark_last_busy(component->dev);
 	pm_runtime_put_autosuspend(component->dev);
 

sound/soc/sof/pm.c

@@ -164,7 +164,7 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 		ret = tplg_ops->set_up_all_pipelines(sdev, false);
 		if (ret < 0) {
 			dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret);
-			return ret;
+			goto setup_fail;
 		}
 	}
 
@@ -178,6 +178,18 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 			dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret);
 	}
 
+setup_fail:
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
+	if (ret < 0) {
+		/*
+		 * Debugfs cannot be read in runtime suspend, so cache
+		 * the contents upon failure. This allows to capture
+		 * possible DSP coredump information.
+		 */
+		sof_cache_debugfs(sdev);
+	}
+#endif
+
 	return ret;
 }
 

sound/soc/sof/sof-client-probes.c

@@ -218,12 +218,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
 
 	ret = ipc->points_info(cdev, &desc, &num_desc);
 	if (ret < 0)
-		goto exit;
-
-	pm_runtime_mark_last_busy(dev);
-	err = pm_runtime_put_autosuspend(dev);
-	if (err < 0)
-		dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+		goto pm_error;
 
 	for (i = 0; i < num_desc; i++) {
 		offset = strlen(buf);
@@ -241,6 +236,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
 	ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf));
 
 	kfree(desc);
+
+pm_error:
+	pm_runtime_mark_last_busy(dev);
+	err = pm_runtime_put_autosuspend(dev);
+	if (err < 0)
+		dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+
 exit:
 	kfree(buf);
 	return ret;

sound/soc/sof/topology.c

@@ -586,6 +586,10 @@ static int sof_copy_tuples(struct snd_sof_dev *sdev, struct snd_soc_tplg_vendor_
 				if (*num_copied_tuples == tuples_size)
 					return 0;
 			}
+
+			/* stop when we've found the required token instances */
+			if (found == num_tokens * token_instance_num)
+				return 0;
 		}
 
 		/* next array */
@@ -1261,7 +1265,7 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s
 		if (num_sets > 1) {
 			struct snd_sof_tuple *new_tuples;
 
-			num_tuples += token_list[object_token_list[i]].count * num_sets;
+			num_tuples += token_list[object_token_list[i]].count * (num_sets - 1);
 			new_tuples = krealloc(swidget->tuples,
 					      sizeof(*new_tuples) * num_tuples, GFP_KERNEL);
 			if (!new_tuples) {