Merge tag 'sound-6.10-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Lots of small fixes:

   - A race fix for debugfs handling in ALSA core

   - A series of corrections for MIDI2 core format conversions

   - ASoC Intel fixes for 16 bit DMIC config

   - Updates for missing module parameters in ASoC code

   - HD-audio quirk, Cirrus codec fix, etc minor fixes"

* tag 'sound-6.10-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
  ALSA: seq: ump: Fix swapped song position pointer data
  ASoC: SOF: ipc4-topology: Adjust the params based on DAI formats
  ASoC: SOF: ipc4-topology: Improve readability of sof_ipc4_prepare_dai_copier()
  ASoC: SOF: ipc4-topology/pcm: Rename sof_ipc4_copier_is_single_format()
  ASoC: SOF: ipc4-topology: Print out the channel count in sof_ipc4_dbg_audio_format
  ASoC: SOF: ipc4-topology: Add support for NHLT with 16-bit only DMIC blob
  ALSA: seq: Fix yet another spot for system message conversion
  ALSA: ump: Set default protocol when not given explicitly
  ALSA: ump: Don't accept an invalid UMP protocol number
  ASoC: SOF: ipc4-topology: Fix input format query of process modules without base extension
  ASoC: Intel: sof-sdw: fix missing SPI_MASTER dependency
  ALSA: pcm: fix typo in comment
  ALSA: ump: Don't clear bank selection after sending a program change
  ALSA: seq: Fix incorrect UMP type for system messages
  ALSA/hda: intel-dsp-config: reduce log verbosity
  ALSA: seq: Don't clear bank selection at event -> UMP MIDI2 conversion
  ALSA: seq: Fix missing bank setup between MIDI1/MIDI2 UMP conversion
  ASoC: SOF: add missing MODULE_DESCRIPTION()
  ASoC: SOF: reorder MODULE_ definitions
  ASoC: SOF: AMD: group all module related information
  ...

Author: torvalds

MAINTAINERS

@@ -5188,7 +5188,6 @@ F:	Documentation/devicetree/bindings/media/i2c/chrontel,ch7322.yaml
 F:	drivers/media/cec/i2c/ch7322.c
 
 CIRRUS LOGIC AUDIO CODEC DRIVERS
-M:	James Schulman <[email protected]>
 M:	David Rhodes <[email protected]>
 M:	Richard Fitzgerald <[email protected]>
 L:	[email protected] (moderated for non-subscribers)

include/sound/pcm.h

@@ -124,7 +124,7 @@ struct snd_pcm_ops {
 #define SNDRV_PCM_RATE_768000		(1U<<16)	/* 768000Hz */
 
 #define SNDRV_PCM_RATE_CONTINUOUS	(1U<<30)	/* continuous range */
-#define SNDRV_PCM_RATE_KNOT		(1U<<31)	/* supports more non-continuos rates */
+#define SNDRV_PCM_RATE_KNOT		(1U<<31)	/* supports more non-continuous rates */
 
 #define SNDRV_PCM_RATE_8000_44100	(SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\
 					 SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\

sound/core/init.c

@@ -537,6 +537,11 @@ void snd_card_disconnect(struct snd_card *card)
 		synchronize_irq(card->sync_irq);
 
 	snd_info_card_disconnect(card);
+#ifdef CONFIG_SND_DEBUG
+	debugfs_remove(card->debugfs_root);
+	card->debugfs_root = NULL;
+#endif
+
 	if (card->registered) {
 		device_del(&card->card_dev);
 		card->registered = false;
@@ -586,10 +591,6 @@ static int snd_card_do_free(struct snd_card *card)
 		dev_warn(card->dev, "unable to free card info\n");
 		/* Not fatal error */
 	}
-#ifdef CONFIG_SND_DEBUG
-	debugfs_remove(card->debugfs_root);
-	card->debugfs_root = NULL;
-#endif
 	if (card->release_completion)
 		complete(card->release_completion);
 	if (!card->managed)

sound/core/jack.c

@@ -37,11 +37,15 @@ static const int jack_switch_types[SND_JACK_SWITCH_TYPES] = {
 };
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
 
+static void snd_jack_remove_debugfs(struct snd_jack *jack);
+
 static int snd_jack_dev_disconnect(struct snd_device *device)
 {
-#ifdef CONFIG_SND_JACK_INPUT_DEV
 	struct snd_jack *jack = device->device_data;
 
+	snd_jack_remove_debugfs(jack);
+
+#ifdef CONFIG_SND_JACK_INPUT_DEV
 	guard(mutex)(&jack->input_dev_lock);
 	if (!jack->input_dev)
 		return 0;
@@ -381,10 +385,14 @@ static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack,
 	return 0;
 }
 
-static void snd_jack_debugfs_clear_inject_node(struct snd_jack_kctl *jack_kctl)
+static void snd_jack_remove_debugfs(struct snd_jack *jack)
 {
-	debugfs_remove(jack_kctl->jack_debugfs_root);
-	jack_kctl->jack_debugfs_root = NULL;
+	struct snd_jack_kctl *jack_kctl;
+
+	list_for_each_entry(jack_kctl, &jack->kctl_list, list) {
+		debugfs_remove(jack_kctl->jack_debugfs_root);
+		jack_kctl->jack_debugfs_root = NULL;
+	}
 }
 #else /* CONFIG_SND_JACK_INJECTION_DEBUG */
 static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack,
@@ -393,7 +401,7 @@ static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack,
 	return 0;
 }
 
-static void snd_jack_debugfs_clear_inject_node(struct snd_jack_kctl *jack_kctl)
+static void snd_jack_remove_debugfs(struct snd_jack *jack)
 {
 }
 #endif /* CONFIG_SND_JACK_INJECTION_DEBUG */
@@ -404,7 +412,6 @@ static void snd_jack_kctl_private_free(struct snd_kcontrol *kctl)
 
 	jack_kctl = kctl->private_data;
 	if (jack_kctl) {
-		snd_jack_debugfs_clear_inject_node(jack_kctl);
 		list_del(&jack_kctl->list);
 		kfree(jack_kctl);
 	}
@@ -497,8 +504,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 		.dev_free = snd_jack_dev_free,
 #ifdef CONFIG_SND_JACK_INPUT_DEV
 		.dev_register = snd_jack_dev_register,
-		.dev_disconnect = snd_jack_dev_disconnect,
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
+		.dev_disconnect = snd_jack_dev_disconnect,
 	};
 
 	if (initial_kctl) {

sound/core/seq/seq_ump_convert.c

@@ -157,7 +157,7 @@ static void ump_system_to_one_param_ev(const union snd_ump_midi1_msg *val,
 static void ump_system_to_songpos_ev(const union snd_ump_midi1_msg *val,
 				     struct snd_seq_event *ev)
 {
-	ev->data.control.value = (val->system.parm1 << 7) | val->system.parm2;
+	ev->data.control.value = (val->system.parm2 << 7) | val->system.parm1;
 }
 
 /* Encoders for 0xf0 - 0xff */
@@ -368,6 +368,7 @@ static int cvt_ump_midi1_to_midi2(struct snd_seq_client *dest,
 	struct snd_seq_ump_event ev_cvt;
 	const union snd_ump_midi1_msg *midi1 = (const union snd_ump_midi1_msg *)event->ump;
 	union snd_ump_midi2_msg *midi2 = (union snd_ump_midi2_msg *)ev_cvt.ump;
+	struct snd_seq_ump_midi2_bank *cc;
 
 	ev_cvt = *event;
 	memset(&ev_cvt.ump, 0, sizeof(ev_cvt.ump));
@@ -387,11 +388,29 @@ static int cvt_ump_midi1_to_midi2(struct snd_seq_client *dest,
 		midi2->paf.data = upscale_7_to_32bit(midi1->paf.data);
 		break;
 	case UMP_MSG_STATUS_CC:
+		cc = &dest_port->midi2_bank[midi1->note.channel];
+		switch (midi1->cc.index) {
+		case UMP_CC_BANK_SELECT:
+			cc->bank_set = 1;
+			cc->cc_bank_msb = midi1->cc.data;
+			return 0; // skip
+		case UMP_CC_BANK_SELECT_LSB:
+			cc->bank_set = 1;
+			cc->cc_bank_lsb = midi1->cc.data;
+			return 0; // skip
+		}
 		midi2->cc.index = midi1->cc.index;
 		midi2->cc.data = upscale_7_to_32bit(midi1->cc.data);
 		break;
 	case UMP_MSG_STATUS_PROGRAM:
 		midi2->pg.program = midi1->pg.program;
+		cc = &dest_port->midi2_bank[midi1->note.channel];
+		if (cc->bank_set) {
+			midi2->pg.bank_valid = 1;
+			midi2->pg.bank_msb = cc->cc_bank_msb;
+			midi2->pg.bank_lsb = cc->cc_bank_lsb;
+			cc->bank_set = 0;
+		}
 		break;
 	case UMP_MSG_STATUS_CHANNEL_PRESSURE:
 		midi2->caf.data = upscale_7_to_32bit(midi1->caf.data);
@@ -419,6 +438,7 @@ static int cvt_ump_midi2_to_midi1(struct snd_seq_client *dest,
 	struct snd_seq_ump_event ev_cvt;
 	union snd_ump_midi1_msg *midi1 = (union snd_ump_midi1_msg *)ev_cvt.ump;
 	const union snd_ump_midi2_msg *midi2 = (const union snd_ump_midi2_msg *)event->ump;
+	int err;
 	u16 v;
 
 	ev_cvt = *event;
@@ -443,6 +463,24 @@ static int cvt_ump_midi2_to_midi1(struct snd_seq_client *dest,
 		midi1->cc.data = downscale_32_to_7bit(midi2->cc.data);
 		break;
 	case UMP_MSG_STATUS_PROGRAM:
+		if (midi2->pg.bank_valid) {
+			midi1->cc.status = UMP_MSG_STATUS_CC;
+			midi1->cc.index = UMP_CC_BANK_SELECT;
+			midi1->cc.data = midi2->pg.bank_msb;
+			err = __snd_seq_deliver_single_event(dest, dest_port,
+							     (struct snd_seq_event *)&ev_cvt,
+							     atomic, hop);
+			if (err < 0)
+				return err;
+			midi1->cc.index = UMP_CC_BANK_SELECT_LSB;
+			midi1->cc.data = midi2->pg.bank_lsb;
+			err = __snd_seq_deliver_single_event(dest, dest_port,
+							     (struct snd_seq_event *)&ev_cvt,
+							     atomic, hop);
+			if (err < 0)
+				return err;
+			midi1->note.status = midi2->note.status;
+		}
 		midi1->pg.program = midi2->pg.program;
 		break;
 	case UMP_MSG_STATUS_CHANNEL_PRESSURE:
@@ -691,6 +729,7 @@ static int system_ev_to_ump_midi1(const struct snd_seq_event *event,
 				  union snd_ump_midi1_msg *data,
 				  unsigned char status)
 {
+	data->system.type = UMP_MSG_TYPE_SYSTEM; // override
 	data->system.status = status;
 	return 1;
 }
@@ -701,6 +740,7 @@ static int system_1p_ev_to_ump_midi1(const struct snd_seq_event *event,
 				     union snd_ump_midi1_msg *data,
 				     unsigned char status)
 {
+	data->system.type = UMP_MSG_TYPE_SYSTEM; // override
 	data->system.status = status;
 	data->system.parm1 = event->data.control.value & 0x7f;
 	return 1;
@@ -712,9 +752,10 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event,
 				     union snd_ump_midi1_msg *data,
 				     unsigned char status)
 {
+	data->system.type = UMP_MSG_TYPE_SYSTEM; // override
 	data->system.status = status;
-	data->system.parm1 = (event->data.control.value >> 7) & 0x7f;
-	data->system.parm2 = event->data.control.value & 0x7f;
+	data->system.parm1 = event->data.control.value & 0x7f;
+	data->system.parm2 = (event->data.control.value >> 7) & 0x7f;
 	return 1;
 }
 
@@ -854,7 +895,6 @@ static int pgm_ev_to_ump_midi2(const struct snd_seq_event *event,
 		data->pg.bank_msb = cc->cc_bank_msb;
 		data->pg.bank_lsb = cc->cc_bank_lsb;
 		cc->bank_set = 0;
-		cc->cc_bank_msb = cc->cc_bank_lsb = 0;
 	}
 	return 1;
 }

sound/core/ump.c

@@ -685,10 +685,17 @@ static void seq_notify_protocol(struct snd_ump_endpoint *ump)
  */
 int snd_ump_switch_protocol(struct snd_ump_endpoint *ump, unsigned int protocol)
 {
+	unsigned int type;
+
 	protocol &= ump->info.protocol_caps;
 	if (protocol == ump->info.protocol)
 		return 0;
 
+	type = protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK;
+	if (type != SNDRV_UMP_EP_INFO_PROTO_MIDI1 &&
+	    type != SNDRV_UMP_EP_INFO_PROTO_MIDI2)
+		return 0;
+
 	ump->info.protocol = protocol;
 	ump_dbg(ump, "New protocol = %x (caps = %x)\n",
 		protocol, ump->info.protocol_caps);
@@ -960,6 +967,14 @@ int snd_ump_parse_endpoint(struct snd_ump_endpoint *ump)
 	if (err < 0)
 		ump_dbg(ump, "Unable to get UMP EP stream config\n");
 
+	/* If no protocol is set by some reason, assume the valid one */
+	if (!(ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK)) {
+		if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI2)
+			ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI2;
+		else if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI1)
+			ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI1;
+	}
+
 	/* Query and create blocks from Function Blocks */
 	for (blk = 0; blk < ump->info.num_blocks; blk++) {
 		err = create_block_from_fb_info(ump, blk);

sound/core/ump_convert.c

@@ -404,7 +404,6 @@ static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt,
 			midi2->pg.bank_msb = cc->cc_bank_msb;
 			midi2->pg.bank_lsb = cc->cc_bank_lsb;
 			cc->bank_set = 0;
-			cc->cc_bank_msb = cc->cc_bank_lsb = 0;
 		}
 		break;
 	case UMP_MSG_STATUS_CHANNEL_PRESSURE:

sound/hda/intel-dsp-config.c

@@ -668,7 +668,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 		return SND_INTEL_DSP_DRIVER_LEGACY;
 	}
 
-	dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class);
+	dev_dbg(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class);
 
 	/* find the configuration for the specific device */
 	cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table));
@@ -678,12 +678,12 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 	if (cfg->flags & FLAG_SOF) {
 		if (cfg->flags & FLAG_SOF_ONLY_IF_SOUNDWIRE &&
 		    snd_intel_dsp_check_soundwire(pci) > 0) {
-			dev_info(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n");
+			dev_info_once(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n");
 			return SND_INTEL_DSP_DRIVER_SOF;
 		}
 		if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC &&
 		    snd_intel_dsp_check_dmic(pci)) {
-			dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n");
+			dev_info_once(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n");
 			return SND_INTEL_DSP_DRIVER_SOF;
 		}
 		if (!(cfg->flags & FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE))
@@ -694,7 +694,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 	if (cfg->flags & FLAG_SST) {
 		if (cfg->flags & FLAG_SST_ONLY_IF_DMIC) {
 			if (snd_intel_dsp_check_dmic(pci)) {
-				dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SST driver\n");
+				dev_info_once(&pci->dev, "Digital mics found on Skylake+ platform, using SST driver\n");
 				return SND_INTEL_DSP_DRIVER_SST;
 			}
 		} else {

sound/pci/hda/patch_realtek.c

@@ -10310,7 +10310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2),
-	SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),

sound/soc/codecs/cs42l43.c

@@ -310,8 +310,9 @@ static int cs42l43_startup(struct snd_pcm_substream *substream, struct snd_soc_d
 	struct snd_soc_component *component = dai->component;
 	struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component);
 	struct cs42l43 *cs42l43 = priv->core;
-	int provider = !!regmap_test_bits(cs42l43->regmap, CS42L43_ASP_CLK_CONFIG2,
-					  CS42L43_ASP_MASTER_MODE_MASK);
+	int provider = !dai->id || !!regmap_test_bits(cs42l43->regmap,
+						      CS42L43_ASP_CLK_CONFIG2,
+						      CS42L43_ASP_MASTER_MODE_MASK);
 
 	if (provider)
 		priv->constraint.mask = CS42L43_PROVIDER_RATE_MASK;