Merge tag 'sound-6.4-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes that have been gathered since rc1:

   - Lots of small ASoC SOF Intel fixes

   - A couple of UAF and NULL-dereference fixes

   - Quirks and updates for HD-audio, USB-audio and ASoC AMD

   - A few minor build / sparse warning fixes

   - MAINTAINERS and DT updates"

* tag 'sound-6.4-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (38 commits)
  ALSA: hda: Add NVIDIA codec IDs a3 through a7 to patch table
  ALSA: oss: avoid missing-prototype warnings
  ALSA: cs46xx: mark snd_cs46xx_download_image as static
  ALSA: hda: Fix Oops by 9.1 surround channel names
  ASoC: SOF: topology: Fix tuples array allocation
  ASoC: SOF: Separate the tokens for input and output pin index
  MAINTAINERS: Remove self from Cirrus Codec drivers
  ASoC: cs35l56: Prevent unbalanced pm_runtime in dsp_work() on SoundWire
  ASoC: SOF: topology: Fix logic for copying tuples
  ASoC: SOF: pm: save io region state in case of errors in resume
  ASoC: MAINTAINERS: drop Krzysztof Kozlowski from Samsung audio
  ASoC: mediatek: mt8186: Fix use-after-free in driver remove path
  ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_config
  ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling
  ASoC: SOF: pcm: fix pm_runtime imbalance in error handling
  ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions
  ASoC: SOF: Intel: hda-mlink: add helper to program SoundWire PCMSyCM registers
  ASoC: SOF: Intel: hda-mlink: initialize instance_offset member
  ASoC: SOF: Intel: hda-mlink: use 'ml_addr' parameter consistently
  ASoC: SOF: Intel: hda-mlink: fix base_ptr computation
  ...

Author: torvalds

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

@@ -4909,7 +4909,6 @@ 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
@@ -18574,10 +18573,9 @@ 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

@@ -44,13 +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/core/oss/pcm_plugin.h

@@ -141,6 +141,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel,
 
 void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size);
 void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr);
+#else
+
+static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; }
+static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; }
+static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; }
+
+#endif
+
 snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream,
 				     const char *ptr, snd_pcm_uframes_t size,
 				     int in_kernel);
@@ -151,14 +159,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream,
 snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream,
 				     void **bufs, snd_pcm_uframes_t frames);
 
-#else
-
-static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; }
-static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; }
-static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; }
-
-#endif
-
 #ifdef PLUGIN_DEBUG
 #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args)
 #else

sound/firewire/digi00x/digi00x-stream.c

@@ -259,8 +259,10 @@ int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
 		return err;
 
 	err = init_stream(dg00x, &dg00x->tx_stream);
-	if (err < 0)
+	if (err < 0) {
 		destroy_stream(dg00x, &dg00x->rx_stream);
+		return err;
+	}
 
 	err = amdtp_domain_init(&dg00x->domain);
 	if (err < 0) {

sound/pci/cs46xx/cs46xx_lib.c

@@ -531,7 +531,7 @@ static int load_firmware(struct snd_cs46xx *chip)
 	return err;
 }
 
-int snd_cs46xx_download_image(struct snd_cs46xx *chip)
+static __maybe_unused int snd_cs46xx_download_image(struct snd_cs46xx *chip)
 {
 	int idx, err;
 	unsigned int offset = 0;

sound/pci/hda/hda_generic.c

@@ -1155,8 +1155,8 @@ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
 	return path && path->ctls[ctl_type];
 }
 
-static const char * const channel_name[4] = {
-	"Front", "Surround", "CLFE", "Side"
+static const char * const channel_name[] = {
+	"Front", "Surround", "CLFE", "Side", "Back",
 };
 
 /* give some appropriate ctl name prefix for the given line out channel */
@@ -1182,7 +1182,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 
 	/* multi-io channels */
 	if (ch >= cfg->line_outs)
-		return channel_name[ch];
+		goto fixed_name;
 
 	switch (cfg->line_out_type) {
 	case AUTO_PIN_SPEAKER_OUT:
@@ -1234,6 +1234,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 	if (cfg->line_outs == 1 && !spec->multi_ios)
 		return "Line Out";
 
+ fixed_name:
 	if (ch >= ARRAY_SIZE(channel_name)) {
 		snd_BUG();
 		return "PCM";