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.

Author: tiwai

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 <[email protected]>
 M:	David Rhodes <[email protected]>
-M:	Lucas Tanure <[email protected]>
 M:	Richard Fitzgerald <[email protected]>
 L:	[email protected] (moderated for non-subscribers)
 L:	[email protected]
@@ -18568,10 +18567,9 @@ F:	Documentation/admin-guide/LSM/SafeSetID.rst
 F:	security/safesetid/
 
 SAMSUNG AUDIO (ASoC) DRIVERS
-M:	Krzysztof Kozlowski <[email protected]>
 M:	Sylwester Nawrocki <[email protected]>
 L:	[email protected] (moderated for non-subscribers)
-S:	Supported
+S:	Maintained
 B:	mailto:[email protected]
 F:	Documentation/devicetree/bindings/sound/samsung*
 F:	sound/soc/samsung/

include/sound/hda-mlink.h

@@ -44,13 +44,17 @@ 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);
 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;
 		}
 	}