Merge/sound upstream 20241106

Documentation/devicetree/bindings/sound/awinic,aw88395.yaml

@@ -17,8 +17,9 @@ description:
 properties:
   compatible:
     enum:
-      - awinic,aw88395
+      - awinic,aw88081
       - awinic,aw88261
+      - awinic,aw88395
       - awinic,aw88399
 
   reg:
@@ -56,6 +57,7 @@ allOf:
         compatible:
           contains:
             enum:
+              - awinic,aw88081
               - awinic,aw88261
     then:
       properties:

Documentation/devicetree/bindings/sound/everest,es8326.yaml

@@ -24,6 +24,10 @@ properties:
     items:
       - const: mclk
 
+  interrupts:
+    maxItems: 1
+    description: interrupt output for headset detection
+
   "#sound-dai-cells":
     const: 0
 

Documentation/devicetree/bindings/sound/fsl,esai.yaml

@@ -18,11 +18,15 @@ description:
 
 properties:
   compatible:
-    enum:
-      - fsl,imx35-esai
-      - fsl,imx6ull-esai
-      - fsl,imx8qm-esai
-      - fsl,vf610-esai
+    oneOf:
+      - enum:
+          - fsl,imx35-esai
+          - fsl,imx6ull-esai
+          - fsl,vf610-esai
+      - items:
+          - enum:
+              - fsl,imx8qm-esai
+          - const: fsl,imx6ull-esai
 
   reg:
     maxItems: 1

Documentation/devicetree/bindings/sound/fsl,spdif.yaml

@@ -16,16 +16,23 @@ description: |
 
 properties:
   compatible:
-    enum:
-      - fsl,imx35-spdif
-      - fsl,vf610-spdif
-      - fsl,imx6sx-spdif
-      - fsl,imx8qm-spdif
-      - fsl,imx8qxp-spdif
-      - fsl,imx8mq-spdif
-      - fsl,imx8mm-spdif
-      - fsl,imx8mn-spdif
-      - fsl,imx8ulp-spdif
+    oneOf:
+      - items:
+          - enum:
+              - fsl,imx35-spdif
+              - fsl,imx6sx-spdif
+              - fsl,imx8mm-spdif
+              - fsl,imx8mn-spdif
+              - fsl,imx8mq-spdif
+              - fsl,imx8qm-spdif
+              - fsl,imx8qxp-spdif
+              - fsl,imx8ulp-spdif
+              - fsl,vf610-spdif
+      - items:
+          - enum:
+              - fsl,imx6sl-spdif
+              - fsl,imx6sx-spdif
+          - const: fsl,imx35-spdif
 
   reg:
     maxItems: 1

Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml

@@ -29,6 +29,13 @@ properties:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8188 ASoC platform.
 
+  mediatek,adsp:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of the MT8188 ADSP platform, which is the optional Audio DSP
+      hardware that provides additional audio functionalities if present.
+      The AFE will link to ADSP when the phandle is provided.
+
 patternProperties:
   "^dai-link-[0-9]+$":
     type: object

Documentation/devicetree/bindings/sound/qcom,sm8250.yaml

@@ -25,6 +25,7 @@ properties:
           - enum:
               - qcom,sm8550-sndcard
               - qcom,sm8650-sndcard
+              - qcom,sm8750-sndcard
           - const: qcom,sm8450-sndcard
       - enum:
           - qcom,apq8096-sndcard

Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml

@@ -0,0 +1,56 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/sprd,pcm-platform.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Spreadtrum DMA platform
+
+maintainers:
+  - Orson Zhai <[email protected]>
+  - Baolin Wang <[email protected]>
+  - Chunyan Zhang <[email protected]>
+
+properties:
+  compatible:
+    const: sprd,pcm-platform
+
+  dmas:
+    maxItems: 10
+
+  dma-names:
+    items:
+      - const: normal_p_l
+      - const: normal_p_r
+      - const: normal_c_l
+      - const: normal_c_r
+      - const: voice_c
+      - const: fast_p
+      - const: loop_c
+      - const: loop_p
+      - const: voip_c
+      - const: voip_p
+
+required:
+  - compatible
+  - dmas
+  - dma-names
+
+additionalProperties: false
+
+examples:
+  - |
+    platform {
+      compatible = "sprd,pcm-platform";
+      dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>,
+             <&agcp_dma 3 3>, <&agcp_dma 4 4>,
+             <&agcp_dma 5 5>, <&agcp_dma 6 6>,
+             <&agcp_dma 7 7>, <&agcp_dma 8 8>,
+             <&agcp_dma 9 9>, <&agcp_dma 10 10>;
+      dma-names = "normal_p_l", "normal_p_r",
+                  "normal_c_l", "normal_c_r",
+                  "voice_c", "fast_p",
+                  "loop_c", "loop_p",
+                  "voip_c", "voip_p";
+    };
+...

Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml

@@ -0,0 +1,47 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/sprd,sc9860-mcdt.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Spreadtrum Multi-Channel Data Transfer controller
+
+description:
+  The Multi-channel data transfer controller is used for sound stream
+  transmission between the audio subsystem and other AP/CP subsystem. It
+  supports 10 DAC channels and 10 ADC channels, and each channel can be
+  configured with DMA mode or interrupt mode.
+
+maintainers:
+  - Orson Zhai <[email protected]>
+  - Baolin Wang <[email protected]>
+  - Chunyan Zhang <[email protected]>
+
+properties:
+  compatible:
+    const: sprd,sc9860-mcdt
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - interrupts
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+
+    mcdt@41490000 {
+      compatible = "sprd,sc9860-mcdt";
+      reg = <0x41490000 0x170>;
+      interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>;
+    };
+...

Documentation/sound/soc/clocking.rst

@@ -42,5 +42,17 @@ rate, number of channels and word size) to save on power.
 It is also desirable to use the codec (if possible) to drive (or master) the
 audio clocks as it usually gives more accurate sample rates than the CPU.
 
+ASoC provided clock APIs
+------------------------
 
+.. kernel-doc:: sound/soc/soc-dai.c
+   :identifiers: snd_soc_dai_set_sysclk
 
+.. kernel-doc:: sound/soc/soc-dai.c
+   :identifiers: snd_soc_dai_set_clkdiv
+
+.. kernel-doc:: sound/soc/soc-dai.c
+   :identifiers: snd_soc_dai_set_pll
+
+.. kernel-doc:: sound/soc/soc-dai.c
+   :identifiers: snd_soc_dai_set_bclk_ratio

MAINTAINERS

@@ -21584,6 +21584,15 @@ S:	Supported
 W:	https://github.com/thesofproject/linux/
 F:	sound/soc/sof/
 
+SOUND - GENERIC SOUND CARD (Simple-Audio-Card, Audio-Graph-Card)
+M:	Kuninori Morimoto <[email protected]>
+S:	Supported
+L:	[email protected]
+F:	sound/soc/generic/
+F:	include/sound/simple_card*
+F:	Documentation/devicetree/bindings/sound/simple-card.yaml
+F:	Documentation/devicetree/bindings/sound/audio-graph*.yaml
+
 SOUNDWIRE SUBSYSTEM
 M:	Vinod Koul <[email protected]>
 M:	Bard Liao <[email protected]>

drivers/base/regmap/internal.h

@@ -59,6 +59,7 @@ struct regmap {
 			unsigned long raw_spinlock_flags;
 		};
 	};
+	struct lock_class_key *lock_key;
 	regmap_lock lock;
 	regmap_unlock unlock;
 	void *lock_arg; /* This is passed to lock/unlock functions */

drivers/base/regmap/regcache-maple.c

@@ -317,6 +317,9 @@ static int regcache_maple_init(struct regmap *map)
 
 	mt_init(mt);
 
+	if (!mt_external_lock(mt) && map->lock_key)
+		lockdep_set_class_and_subclass(&mt->ma_lock, map->lock_key, 1);
+
 	for (i = 0; i < map->num_reg_defaults; i++) {
 		ret = regcache_maple_write(map,
 					   map->reg_defaults[i].reg,

drivers/base/regmap/regmap-irq.c

@@ -511,12 +511,16 @@ static irqreturn_t regmap_irq_thread(int irq, void *d)
 		return IRQ_NONE;
 }
 
+static struct lock_class_key regmap_irq_lock_class;
+static struct lock_class_key regmap_irq_request_class;
+
 static int regmap_irq_map(struct irq_domain *h, unsigned int virq,
 			  irq_hw_number_t hw)
 {
 	struct regmap_irq_chip_data *data = h->host_data;
 
 	irq_set_chip_data(virq, data);
+	irq_set_lockdep_class(virq, &regmap_irq_lock_class, &regmap_irq_request_class);
 	irq_set_chip(virq, &data->irq_chip);
 	irq_set_nested_thread(virq, 1);
 	irq_set_parent(virq, data->irq);

drivers/base/regmap/regmap.c

@@ -745,6 +745,7 @@ struct regmap *__regmap_init(struct device *dev,
 						   lock_key, lock_name);
 		}
 		map->lock_arg = map;
+		map->lock_key = lock_key;
 	}
 
 	/*

include/sound/soc-dai.h

@@ -216,8 +216,7 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			  struct snd_pcm_substream *substream, int rollback);
 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 void snd_soc_dai_resume(struct snd_soc_dai *dai);
-int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
-			     struct snd_soc_pcm_runtime *rtd, int num);
+int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd);
 bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream);
 void snd_soc_dai_action(struct snd_soc_dai *dai,
 			int stream, int action);
@@ -275,7 +274,7 @@ struct snd_soc_dai_ops {
 	int (*probe)(struct snd_soc_dai *dai);
 	int (*remove)(struct snd_soc_dai *dai);
 	/* compress dai */
-	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
+	int (*compress_new)(struct snd_soc_pcm_runtime *rtd);
 	/* Optional Callback used at pcm creation*/
 	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
 		       struct snd_soc_dai *dai);

include/sound/soc.h

@@ -486,11 +486,11 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev,
 struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 						   const char *driver_name);
 
-int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd);
 #ifdef CONFIG_SND_SOC_COMPRESS
-int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
+int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd);
 #else
-static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
+static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd)
 {
 	return 0;
 }
@@ -1195,7 +1195,7 @@ struct snd_soc_pcm_runtime {
 	struct dentry *debugfs_dpcm_root;
 #endif
 
-	unsigned int num; /* 0-based and monotonic increasing */
+	unsigned int id; /* 0-based and monotonic increasing */
 	struct list_head list; /* rtd list of the soc card */
 
 	/* function mark */

sound/core/ump.c

@@ -1233,7 +1233,7 @@ static int fill_legacy_mapping(struct snd_ump_endpoint *ump)
 
 	num = 0;
 	for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++)
-		if (group_maps & (1U << i))
+		if ((group_maps & (1U << i)) && ump->groups[i].valid)
 			ump->legacy_mapping[num++] = i;
 
 	return num;

sound/firewire/amdtp-stream.c

@@ -172,6 +172,9 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params,
 			step = max(step, amdtp_syt_intervals[i]);
 	}
 
+	if (step == 0)
+		return -EINVAL;
+
 	t.min = roundup(s->min, step);
 	t.max = rounddown(s->max, step);
 	t.integer = 1;