From fc7c5c208eb7bc2df3a9f4234f14eca250001cb6 Mon Sep 17 00:00:00 2001
From: John Ernberg <john.ernberg@actia.se>
Date: Wed, 3 Mar 2021 18:14:39 +0000
Subject: ALSA: usb: Add Plantronics C320-M USB ctrl msg delay quirk

The microphone in the Plantronics C320-M headset will randomly
fail to initialize properly, at least when using Microsoft Teams.
Introducing a 20ms delay on the control messages appears to
resolve the issue.

Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065
Tested-by: Andreas Kempe <kempe@lysator.liu.se>
Signed-off-by: John Ernberg <john.ernberg@actia.se>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303181405.39835-1-john.ernberg@actia.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 737b2729c0d3..0864692d8a7b 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1670,6 +1670,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	    && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		msleep(20);
 
+	/*
+	 * Plantronics C320-M needs a delay to avoid random
+	 * microhpone failures.
+	 */
+	if (chip->usb_id == USB_ID(0x047f, 0xc025)  &&
+	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+		msleep(20);
+
 	/* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950),
 	 * Jabra 550a, Kingston HyperX needs a tiny delay here,
 	 * otherwise requests like get/set frequency return
-- 
cgit 


From 9799110825dba087c2bdce886977cf84dada2005 Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Thu, 4 Mar 2021 12:34:16 +0800
Subject: ALSA: usb-audio: Disable USB autosuspend properly in
 setup_disable_autosuspend()

Rear audio on Lenovo ThinkStation P620 stops working after commit
1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo
ThinkStation P620"):
[    6.013526] usbcore: registered new interface driver snd-usb-audio
[    6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[    6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[    6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[    6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[    6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[    6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[    6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4

I overlooked the issue because when I was working on the said commit,
only the front audio is tested. Apology for that.

Changing supports_autosuspend in driver is too late for disabling
autosuspend, because it was already used by USB probe routine, so it can
break the balance on the following code that depends on
supports_autosuspend.

Fix it by using usb_disable_autosuspend() helper, and balance the
suspend count in disconnect callback.

Fixes: 1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304043419.287191-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/card.c     | 5 +++++
 sound/usb/quirks.c   | 2 +-
 sound/usb/usbaudio.h | 1 +
 3 files changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 85ed8507e41a..08c794883299 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -830,6 +830,8 @@ static int usb_audio_probe(struct usb_interface *intf,
 		snd_media_device_create(chip, intf);
 	}
 
+	chip->quirk_type = quirk->type;
+
 	usb_chip[chip->index] = chip;
 	chip->intf[chip->num_interfaces] = intf;
 	chip->num_interfaces++;
@@ -912,6 +914,9 @@ static void usb_audio_disconnect(struct usb_interface *intf)
 	} else {
 		mutex_unlock(&register_mutex);
 	}
+
+	if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND)
+		usb_enable_autosuspend(interface_to_usbdev(intf));
 }
 
 /* lock the shutdown (disconnect) task and autoresume */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 0864692d8a7b..66efedaa3aeb 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -547,7 +547,7 @@ static int setup_disable_autosuspend(struct snd_usb_audio *chip,
 				       struct usb_driver *driver,
 				       const struct snd_usb_audio_quirk *quirk)
 {
-	driver->supports_autosuspend = 0;
+	usb_disable_autosuspend(interface_to_usbdev(iface));
 	return 1;	/* Continue with creating streams and mixer */
 }
 
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 215c1771dd57..60b9dd7df6bb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -27,6 +27,7 @@ struct snd_usb_audio {
 	struct snd_card *card;
 	struct usb_interface *intf[MAX_CARD_INTERFACES];
 	u32 usb_id;
+	uint16_t quirk_type;
 	struct mutex mutex;
 	unsigned int system_suspend;
 	atomic_t active;
-- 
cgit 


From a14a6219996ee6f6e858d83b11affc7907633687 Mon Sep 17 00:00:00 2001
From: Mark Pearson <markpearson@lenovo.com>
Date: Tue, 2 Mar 2021 09:10:03 -0500
Subject: ALSA: hda: ignore invalid NHLT table

On some Lenovo systems if the microphone is disabled in the BIOS
only the NHLT table header is created, with no data. This means
the endpoints field is not correctly set to zero - leading to an
unintialised variable and hence invalid descriptors are parsed
leading to page faults.

The Lenovo firmware team is addressing this, but adding a check
preventing invalid tables being parsed is worthwhile.

Tested on a Lenovo T14.

Tested-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Reported-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Signed-off-by: Mark Pearson <markpearson@lenovo.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/intel-nhlt.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c
index d053beccfaec..e2237239d922 100644
--- a/sound/hda/intel-nhlt.c
+++ b/sound/hda/intel-nhlt.c
@@ -39,6 +39,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt)
 	if (!nhlt)
 		return 0;
 
+	if (nhlt->header.length <= sizeof(struct acpi_table_header)) {
+		dev_warn(dev, "Invalid DMIC description table\n");
+		return 0;
+	}
+
 	for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++,
 	     epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) {
 
-- 
cgit 


From fec60c3bc5d1713db2727cdffc638d48f9c07dc3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 4 Mar 2021 09:30:21 +0100
Subject: ALSA: usb-audio: Fix "cannot get freq eq" errors on Dell AE515 sound
 bar

Dell AE515 sound bar (413c:a506) spews the error messages when the
driver tries to read the current sample frequency, hence it needs to
be on the list in snd_usb_get_sample_rate_quirk().

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211551
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304083021.2152-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 66efedaa3aeb..6ef73af00913 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1520,6 +1520,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */
 	case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */
 	case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */
+	case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */
 		return true;
 	}
 
-- 
cgit 


From 06abcb18b3a021ba1a3f2020cbefb3ed04e59e72 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 4 Mar 2021 09:50:09 +0100
Subject: ALSA: usb-audio: Apply the control quirk to Plantronics headsets

Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too.  Apply the
workaround generically for devices with the vendor ID 0x047f.

Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.

BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 6ef73af00913..d3001fb18141 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1672,10 +1672,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 		msleep(20);
 
 	/*
-	 * Plantronics C320-M needs a delay to avoid random
-	 * microhpone failures.
+	 * Plantronics headsets (C320, C320-M, etc) need a delay to avoid
+	 * random microhpone failures.
 	 */
-	if (chip->usb_id == USB_ID(0x047f, 0xc025)  &&
+	if (USB_ID_VENDOR(chip->usb_id) == 0x047f &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		msleep(20);
 
-- 
cgit 


From 56b26497bb4b7ff970612dc25a8a008c34463f7b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 6 Mar 2021 10:50:18 +0100
Subject: ALSA: hda/conexant: Add quirk for mute LED control on HP ZBook G5

The mute and mic-mute LEDs on HP ZBook Studio G5 are controlled via
GPIO bits 0x10 and 0x20, respectively, and we need the extra setup for
those.

As the similar code is already present for other HP models but with
different GPIO pins, this patch factors out the common helper code and
applies those GPIO values for each model.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211893
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210306095018.11746-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 62 ++++++++++++++++++++++++++++++------------
 1 file changed, 45 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f2aa226d1373..c20dad46a7c9 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -149,6 +149,21 @@ static int cx_auto_vmaster_mute_led(struct led_classdev *led_cdev,
 	return 0;
 }
 
+static void cxt_init_gpio_led(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask;
+
+	if (mask) {
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+				    mask);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+				    mask);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_led);
+	}
+}
+
 static int cx_auto_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -156,6 +171,7 @@ static int cx_auto_init(struct hda_codec *codec)
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
 
+	cxt_init_gpio_led(codec);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
 
 	return 0;
@@ -215,6 +231,7 @@ enum {
 	CXT_FIXUP_HP_SPECTRE,
 	CXT_FIXUP_HP_GATE_MIC,
 	CXT_FIXUP_MUTE_LED_GPIO,
+	CXT_FIXUP_HP_ZBOOK_MUTE_LED,
 	CXT_FIXUP_HEADSET_MIC,
 	CXT_FIXUP_HP_MIC_NO_PRESENCE,
 };
@@ -654,31 +671,36 @@ static int cxt_gpio_micmute_update(struct led_classdev *led_cdev,
 	return 0;
 }
 
-
-static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
-				const struct hda_fixup *fix, int action)
+static void cxt_setup_mute_led(struct hda_codec *codec,
+			       unsigned int mute, unsigned int mic_mute)
 {
 	struct conexant_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x03 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 },
-		{}
-	};
 
-	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+	spec->gpio_led = 0;
+	spec->mute_led_polarity = 0;
+	if (mute) {
 		snd_hda_gen_add_mute_led_cdev(codec, cxt_gpio_mute_update);
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x01;
-		spec->gpio_mic_led_mask = 0x02;
+		spec->gpio_mute_led_mask = mute;
+	}
+	if (mic_mute) {
 		snd_hda_gen_add_micmute_led_cdev(codec, cxt_gpio_micmute_update);
+		spec->gpio_mic_led_mask = mic_mute;
 	}
-	snd_hda_add_verbs(codec, gpio_init);
-	if (spec->gpio_led)
-		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-				    spec->gpio_led);
 }
 
+static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		cxt_setup_mute_led(codec, 0x01, 0x02);
+}
+
+static void cxt_fixup_hp_zbook_mute_led(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		cxt_setup_mute_led(codec, 0x10, 0x20);
+}
 
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
@@ -839,6 +861,10 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_mute_led_gpio,
 	},
+	[CXT_FIXUP_HP_ZBOOK_MUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_hp_zbook_mute_led,
+	},
 	[CXT_FIXUP_HEADSET_MIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_headset_mic,
@@ -917,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO),
+	SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE),
@@ -956,6 +983,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = {
 	{ .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" },
 	{ .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" },
 	{ .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" },
+	{ .id = CXT_FIXUP_HP_ZBOOK_MUTE_LED, .name = "hp-zbook-mute-led" },
 	{ .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" },
 	{}
 };
-- 
cgit 


From 28e96c1693ec1cdc963807611f8b5ad400431e82 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 8 Mar 2021 17:07:26 +0100
Subject: ALSA: hda: Drop the BATCH workaround for AMD controllers

The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on
AMD controller (1022:1457)") introduced a few workarounds for the
recent AMD HD-audio controller, and one of them is the forced BATCH
PCM mode so that PulseAudio avoids the timer-based scheduling.  This
was thought to cover for some badly working applications, but this
actually worsens for more others.  In total, this wasn't a good idea
to enforce it.

This is a partial revert of the commit above for dropping the PCM
BATCH enforcement part to recover from the regression again.

Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210308160726.22930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c | 7 -------
 1 file changed, 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 9087981cd1f7..ca2f2ecd1488 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -609,13 +609,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
-	/* by some reason, the playback stream stalls on PulseAudio with
-	 * tsched=1 when a capture stream triggers.  Until we figure out the
-	 * real cause, disable tsched mode by telling the PCM info flag.
-	 */
-	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
-		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
-
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
-- 
cgit 


From f15c5c11abfbf8909eb30598315ecbec2311cfdc Mon Sep 17 00:00:00 2001
From: Simeon Simeonoff <sim.simeonoff@gmail.com>
Date: Mon, 8 Mar 2021 20:48:35 +0200
Subject: ALSA: hda/ca0132: Add Sound BlasterX AE-5 Plus support

The new AE-5 Plus model has a different Subsystem ID compared to the
non-plus model. Adding the new id to the list of quirks.

Signed-off-by: Simeon Simeonoff <sim.simeonoff@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/998cafbe10b648f724ee33570553f2d780a38963.camel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index c966f49fa942..b2b620f6c832 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1309,6 +1309,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+	SND_PCI_QUIRK(0x1102, 0x0191, "Sound Blaster AE-5 Plus", QUIRK_AE5),
 	SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
 	{}
 };
-- 
cgit 


From 30dea07180de3aa0ad613af88431ef4e34b5ef68 Mon Sep 17 00:00:00 2001
From: Pavel Skripkin <paskripkin@gmail.com>
Date: Tue, 9 Mar 2021 01:30:36 +0300
Subject: ALSA: usb-audio: fix NULL ptr dereference in usb_audio_probe

syzbot reported null pointer dereference in usb_audio_probe.
The problem was in case, when quirk == NULL. It's not an
error condition, so quirk must be checked before dereferencing.

Call Trace:
 usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396
 really_probe+0x291/0xe60 drivers/base/dd.c:554
 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740
 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846
 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431
 __device_attach+0x228/0x4a0 drivers/base/dd.c:914
 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491
 device_add+0xbdb/0x1db0 drivers/base/core.c:3242
 usb_set_configuration+0x113f/0x1910 drivers/usb/core/message.c:2164
 usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238
 usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293
 really_probe+0x291/0xe60 drivers/base/dd.c:554
 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740
 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846
 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431
 __device_attach+0x228/0x4a0 drivers/base/dd.c:914
 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491
 device_add+0xbdb/0x1db0 drivers/base/core.c:3242
 usb_new_device.cold+0x721/0x1058 drivers/usb/core/hub.c:2555
 hub_port_connect drivers/usb/core/hub.c:5223 [inline]
 hub_port_connect_change drivers/usb/core/hub.c:5363 [inline]
 port_event drivers/usb/core/hub.c:5509 [inline]
 hub_event+0x2357/0x4320 drivers/usb/core/hub.c:5591
 process_one_work+0x98d/0x1600 kernel/workqueue.c:2275
 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421
 kthread+0x3b1/0x4a0 kernel/kthread.c:292
 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294

Reported-by: syzbot+719da9b149a931f5143f@syzkaller.appspotmail.com
Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/f1ebad6e721412843bd1b12584444c0a63c6b2fb.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/card.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 08c794883299..3fd1743513b5 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -830,7 +830,8 @@ static int usb_audio_probe(struct usb_interface *intf,
 		snd_media_device_create(chip, intf);
 	}
 
-	chip->quirk_type = quirk->type;
+	if (quirk)
+		chip->quirk_type = quirk->type;
 
 	usb_chip[chip->index] = chip;
 	chip->intf[chip->num_interfaces] = intf;
-- 
cgit 


From c5aa956eaeb05fe87e33433d7fd9f5e4d23c7416 Mon Sep 17 00:00:00 2001
From: Pavel Skripkin <paskripkin@gmail.com>
Date: Tue, 9 Mar 2021 01:30:57 +0300
Subject: ALSA: usb-audio: fix use after free in usb_audio_disconnect

The problem was in wrong "if" placement. chip->quirk_type is freed
in snd_card_free_when_closed(), but inside if statement it's accesed.

Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/16da19126ff461e5e64a9aec648cce28fb8ed73e.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/card.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 3fd1743513b5..b6f4c0848e66 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -907,6 +907,9 @@ static void usb_audio_disconnect(struct usb_interface *intf)
 		}
 	}
 
+	if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND)
+		usb_enable_autosuspend(interface_to_usbdev(intf));
+
 	chip->num_interfaces--;
 	if (chip->num_interfaces <= 0) {
 		usb_chip[chip->index] = NULL;
@@ -915,9 +918,6 @@ static void usb_audio_disconnect(struct usb_interface *intf)
 	} else {
 		mutex_unlock(&register_mutex);
 	}
-
-	if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND)
-		usb_enable_autosuspend(interface_to_usbdev(intf));
 }
 
 /* lock the shutdown (disconnect) task and autoresume */
-- 
cgit 


From 13661fc48461282e43fe8f76bf5bf449b3d40687 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 10 Mar 2021 12:28:07 +0100
Subject: ALSA: hda: Flush pending unsolicited events before suspend

The HD-audio controller driver processes the unsolicited events via
its work asynchronously, and this might be pending when the system
goes to suspend.  When a lengthy event handling like ELD byte reads is
running, this might trigger unexpected accesses among suspend/resume
procedure, typically seen with Nvidia driver that still requires the
handling via unsolicited event verbs for ELD updates.

This patch adds the flush of unsol_work to assure that pending events
are processed before going into suspend.

Buglink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 5b492c3f816c..5eea130dcf0a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1026,6 +1026,8 @@ static int azx_prepare(struct device *dev)
 	chip = card->private_data;
 	chip->pm_prepared = 1;
 
+	flush_work(&azx_bus(chip)->unsol_work);
+
 	/* HDA controller always requires different WAKEEN for runtime suspend
 	 * and system suspend, so don't use direct-complete here.
 	 */
-- 
cgit 


From 5ff9dde42e8c72ed8102eb8cb62e03f9dc2103ab Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 10 Mar 2021 12:28:08 +0100
Subject: ALSA: hda: Avoid spurious unsol event handling during S3/S4

When HD-audio bus receives unsolicited events during its system
suspend/resume (S3 and S4) phase, the controller driver may still try
to process events although the codec chips are already (or yet)
powered down.  This might screw up the codec communication, resulting
in CORB/RIRB errors.  Such events should be rather skipped, as the
codec chip status such as the jack status will be fully refreshed at
the system resume time.

Since we're tracking the system suspend/resume state in codec
power.power_state field, let's add the check in the common unsol event
handler entry point to filter out such events.

BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org> # 183ab39eb0ea: ALSA: hda: Initialize power_state
Link: https://lore.kernel.org/r/20210310112809.9215-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_bind.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 6a8564566375..17a25e453f60 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -47,6 +47,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
 	if (codec->bus->shutdown)
 		return;
 
+	/* ignore unsol events during system suspend/resume */
+	if (codec->core.dev.power.power_state.event != PM_EVENT_ON)
+		return;
+
 	if (codec->patch_ops.unsol_event)
 		codec->patch_ops.unsol_event(codec, ev);
 }
-- 
cgit 


From eea46a0879bcca23e15071f9968c0f6e6596e470 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 10 Mar 2021 12:28:09 +0100
Subject: ALSA: hda/hdmi: Cancel pending works before suspend

The per_pin->work might be still floating at the suspend, and this may
hit the access to the hardware at an unexpected timing.  Cancel the
work properly at the suspend callback for avoiding the buggy access.

Note that the bug doesn't trigger easily in the recent kernels since
the work is queued only when the repoll count is set, and usually it's
only at the resume callback, but it's still possible to hit in
theory.

BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 13 +++++++++++++
 1 file changed, 13 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index e6d0843ee9df..45ae845e82df 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2480,6 +2480,18 @@ static void generic_hdmi_free(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_PM
+static int generic_hdmi_suspend(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	int pin_idx;
+
+	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+		struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+		cancel_delayed_work_sync(&per_pin->work);
+	}
+	return 0;
+}
+
 static int generic_hdmi_resume(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -2503,6 +2515,7 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = {
 	.build_controls		= generic_hdmi_build_controls,
 	.unsol_event		= hdmi_unsol_event,
 #ifdef CONFIG_PM
+	.suspend		= generic_hdmi_suspend,
 	.resume			= generic_hdmi_resume,
 #endif
 };
-- 
cgit 


From 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Mon, 22 Feb 2021 17:00:56 +0800
Subject: ASoC: rt1015: fix i2c communication error

Remove 0x100 cache re-sync to solve i2c communication error.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1015.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 37b5795b00d1..90767490af82 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg)
 	case RT1015_VENDOR_ID:
 	case RT1015_DEVICE_ID:
 	case RT1015_PRO_ALT:
+	case RT1015_MAN_I2C:
 	case RT1015_DAC3:
 	case RT1015_VBAT_TEST_OUT1:
 	case RT1015_VBAT_TEST_OUT2:
-- 
cgit 


From 0d2b6e398975bcc6a29f1d466229a312dde71b53 Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Mon, 22 Feb 2021 17:00:57 +0800
Subject: ASoC: rt1015: enable BCLK detection after calibration

Enable BCLK detection after calibration.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-2-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1015.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 90767490af82..844e4079d176 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -514,6 +514,7 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015)
 	msleep(300);
 	regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008);
 	regmap_write(regmap, RT1015_SYS_RST1, 0x05F5);
+	regmap_write(regmap, RT1015_CLK_DET, 0x8000);
 
 	regcache_cache_bypass(regmap, false);
 	regcache_mark_dirty(regmap);
-- 
cgit 


From 0c0a5883783540a56e6a5dbf5868f045dbeaa888 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Mon, 15 Feb 2021 16:33:13 +0000
Subject: ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx

There is potential read of the uninitialized variable ec_tx if the call
to snd_soc_component_read fails or returns an unrecognized w->name. To
avoid this corner case, initialize ec_tx to -1 so that it is caught
later when ec_tx is bounds checked.

Addresses-Coverity: ("Uninitialized scalar variable")
Fixes: 4f692926f562 ("ASoC: codecs: lpass-rx-macro: add dapm widgets and route")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210215163313.84026-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lpass-rx-macro.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index c9c21d22c2c4..8c04b3b2c907 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -2895,7 +2895,7 @@ static int rx_macro_enable_echo(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
 	u16 val, ec_hq_reg;
-	int ec_tx;
+	int ec_tx = -1;
 
 	val = snd_soc_component_read(component,
 			CDC_RX_INP_MUX_RX_MIX_CFG4);
-- 
cgit 


From 87263968516fb9507d6215d53f44052627fae8d8 Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Tue, 16 Feb 2021 14:42:21 +0300
Subject: ASoC: fsl_ssi: Fix TDM slot setup for I2S mode

When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt()
function rewrites the number of slots previously set by the
fsl_ssi_set_dai_tdm_slot() function to 2 by default.
To fix this, let's use the saved slot count value or, if TDM
is not used and the slot count is not set, proceed as before.

Fixes: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_ssi.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 57811743c294..ad8af3f450e2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -878,6 +878,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
 static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 {
 	u32 strcr = 0, scr = 0, stcr, srcr, mask;
+	unsigned int slots;
 
 	ssi->dai_fmt = fmt;
 
@@ -909,10 +910,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 			return -EINVAL;
 		}
 
+		slots = ssi->slots ? : 2;
 		regmap_update_bits(ssi->regs, REG_SSI_STCCR,
-				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
 		regmap_update_bits(ssi->regs, REG_SSI_SRCCR,
-				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
 
 		/* Data on rising edge of bclk, frame low, 1clk before data */
 		strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS;
-- 
cgit 


From d917b5dde660b11abd757bf99a29353c36880b2c Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Mon, 15 Feb 2021 15:21:15 +0100
Subject: ASoC: rt5670: Remove 'OUT Channel Switch' control

The "OUT Channel Switch" control is a left over from code copied from
thr rt5640 codec driver.

With the rt5640 codec driver the output volume controls have 2 pairs of
mute bits:
bit 7, 15: Mute Control for Spk/Headphone/Line Output Port
bit 6, 14: Mute Control for Spk/Headphone/Line Volume Channel

Bits 7 and 15 are normal mute bits on the rt5670/5672 which are
controlled by 2 dapm widgets:
	SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0,
			    &lout_l_enable_control),
	SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0,
			    &lout_r_enable_control),

But on the 5670/5672 bit 6 is always reserved, where as bit 14 is
"LOUT Differential Mode" on the 5670 and also reserved on the 5672.

So the "OUT Channel Switch" control which is controlling bits 6+14
of the "LINE Output Control" register is bogus -> remove it.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there do not
use the "OUT Channel Switch" control.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 2 --
 sound/soc/codecs/rt5670.h | 4 ----
 2 files changed, 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index c29317ea5df2..2e799e21dbda 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -637,8 +637,6 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
 		39, 1, out_vol_tlv),
 	/* OUTPUT Control */
-	SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
-		RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1,
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* DAC Digital Volume */
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index 56b13fe6bd3c..f9c4db156c80 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -212,12 +212,8 @@
 /* global definition */
 #define RT5670_L_MUTE				(0x1 << 15)
 #define RT5670_L_MUTE_SFT			15
-#define RT5670_VOL_L_MUTE			(0x1 << 14)
-#define RT5670_VOL_L_SFT			14
 #define RT5670_R_MUTE				(0x1 << 7)
 #define RT5670_R_MUTE_SFT			7
-#define RT5670_VOL_R_MUTE			(0x1 << 6)
-#define RT5670_VOL_R_SFT			6
 #define RT5670_L_VOL_MASK			(0x3f << 8)
 #define RT5670_L_VOL_SFT			8
 #define RT5670_R_VOL_MASK			(0x3f)
-- 
cgit 


From caba8d764770b6824391c5bf3c3eba6e51b69330 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Mon, 15 Feb 2021 15:21:16 +0100
Subject: ASoC: rt5670: Remove 'HP Playback Switch' control

The RT5670_L_MUTE_SFT and RT5670_R_MUTE_SFT bits (bits 15 and 7) of the
RT5670_HP_VOL register are set / unset by the headphones deplop code
run by rt5670_hp_event() on SND_SOC_DAPM_POST_PMU / SND_SOC_DAPM_PRE_PMD.

So we should not also export a control to userspace which toggles these
same bits.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there do not
use the "HP Playback Switch" control.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 2e799e21dbda..932e4cd1e9a6 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -631,8 +631,6 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA,
 
 static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL,
-		RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
 		39, 1, out_vol_tlv),
-- 
cgit 


From 02aa946ef3762aa456d87cc55606667942b3f354 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Mon, 15 Feb 2021 15:21:17 +0100
Subject: ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC
 mixer settings

The SND_SOC_DAPM_MIXER declaration for "Sto1 ADC MIXL" and "Sto1 ADC MIXR"
was using the mute bits from the RT5670_STO1_ADC_DIG_VOL control as mixer
master mute bits.

But these bits are already exposed to userspace as controls as part of the
"ADC Capture Volume" / "ADC Capture Switch" control pair:

        SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL,
                RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
        SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL,
                        RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
                        127, 0, adc_vol_tlv),

Both the fact that the mute bits belong to the same reg as the vol-ctrl
and the "Digital Mixer Path" diagram in the datasheet clearly shows that
these mute bits are not part of the mixer and having 2 separate controls
poking at the same bits is a bad idea.

Remove the master-mute bits settings from the  "Sto1 ADC MIXL" and
"Sto1 ADC MIXR" DAPM widget declarations, avoiding these bits getting
poked from 2 different places.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there already
set the "ADC Capture Switch" as needed.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 932e4cd1e9a6..2f015c24c637 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -1652,12 +1652,10 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
 			    RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2,
 			    RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL,
-			   RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix,
-			   ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
-	SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL,
-			   RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix,
-			   ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
+	SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0,
+			   rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
+	SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0,
+			   rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
 	SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0,
 			   rt5670_sto2_adc_l_mix,
 			   ARRAY_SIZE(rt5670_sto2_adc_l_mix)),
-- 
cgit 


From 42121c2645d229d348399ad278b6c3fd224bd6a2 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Mon, 15 Feb 2021 15:21:18 +0100
Subject: ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control

For reliable output-mute LED control we need a "DAC1 Playback Switch"
control. The "DAC Playback volume" control is the only control in the
path from the DAC1 data input to the speaker output, so the UCM profile
for the speaker output will have its PlaybackMixerElem set to "DAC1".

But userspace (pulseaudio) will set the "DAC1 Playback Volume" control to
its softest setting (which is not fully muted) while still showing the
speaker as being enabled at a low volume in the UI.

If we were to set the SNDRV_CTL_ELEM_ACCESS_SPK_LED on the "DAC1 Playback
Volume" control, this would mean then what pressing KEY_VOLUMEDOWN the
speaker-mute LED (embedded in the volume-mute toggle key) would light
while the UI is still showing the speaker as being enabled at a low
volume, meaning that the UI and the LED are out of sync.

Only after an _extra_ KEY_VOLUMEDOWN press would the UI show the
speaker as being muted.

The path from DAC1 data input to the speaker output does have
a digital mixer with DAC1's data as one of its inputs direclty after
the "DAC1 Playback Volume" control.

This commit adds an emulated "DAC1 Playback Switch" control by:

1. Declaring the enable flag for that mixers DAC1 input as well as the
"DAC1 Playback Switch" control both as SND_SOC_NOPM controls.

2. Storing the settings of both controls as driver-private data

3. Only clearing the mute flag for the DAC1 input of that mixer if the
stored values indicate both controls are enabled.

This is a preparation patch for adding "audio-mute" LED trigger support.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 96 +++++++++++++++++++++++++++++++++++++++++++++--
 sound/soc/codecs/rt5670.h |  5 +++
 2 files changed, 97 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 2f015c24c637..4063aac2a443 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -629,6 +629,56 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA,
 static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA,
 				RT5670_IF2_ADC_SEL_SFT, rt5670_data_select);
 
+/*
+ * For reliable output-mute LED control we need a "DAC1 Playback Switch" control.
+ * We emulate this by only clearing the RT5670_M_DAC1_L/_R AD_DA_MIXER register
+ * bits when both our emulated DAC1 Playback Switch control and the DAC1 MIXL/R
+ * DAPM-mixer DAC1 input are enabled.
+ */
+static void rt5670_update_ad_da_mixer_dac1_m_bits(struct rt5670_priv *rt5670)
+{
+	int val = RT5670_M_DAC1_L | RT5670_M_DAC1_R;
+
+	if (rt5670->dac1_mixl_dac1_switch && rt5670->dac1_playback_switch_l)
+		val &= ~RT5670_M_DAC1_L;
+
+	if (rt5670->dac1_mixr_dac1_switch && rt5670->dac1_playback_switch_r)
+		val &= ~RT5670_M_DAC1_R;
+
+	regmap_update_bits(rt5670->regmap, RT5670_AD_DA_MIXER,
+			   RT5670_M_DAC1_L | RT5670_M_DAC1_R, val);
+}
+
+static int rt5670_dac1_playback_switch_get(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	ucontrol->value.integer.value[0] = rt5670->dac1_playback_switch_l;
+	ucontrol->value.integer.value[1] = rt5670->dac1_playback_switch_r;
+
+	return 0;
+}
+
+static int rt5670_dac1_playback_switch_put(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	if (rt5670->dac1_playback_switch_l == ucontrol->value.integer.value[0] &&
+	    rt5670->dac1_playback_switch_r == ucontrol->value.integer.value[1])
+		return 0;
+
+	rt5670->dac1_playback_switch_l = ucontrol->value.integer.value[0];
+	rt5670->dac1_playback_switch_r = ucontrol->value.integer.value[1];
+
+	rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+	return 1;
+}
+
 static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 	/* Headphone Output Volume */
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
@@ -640,6 +690,8 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 	/* DAC Digital Volume */
 	SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL,
 		RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1),
+	SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0,
+			rt5670_dac1_playback_switch_get, rt5670_dac1_playback_switch_put),
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL,
 			RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
 			175, 0, dac_vol_tlv),
@@ -909,18 +961,44 @@ static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = {
 			RT5670_M_MONO_ADC_R2_SFT, 1, 1),
 };
 
+/* See comment above rt5670_update_ad_da_mixer_dac1_m_bits() */
+static int rt5670_put_dac1_mix_dac1_switch(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+	int ret;
+
+	if (mc->shift == 0)
+		rt5670->dac1_mixl_dac1_switch = ucontrol->value.integer.value[0];
+	else
+		rt5670->dac1_mixr_dac1_switch = ucontrol->value.integer.value[0];
+
+	/* Apply the update (if any) */
+	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+	if (ret == 0)
+		return 0;
+
+	rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+	return 1;
+}
+
+#define SOC_DAPM_SINGLE_RT5670_DAC1_SW(name, shift) \
+	SOC_SINGLE_EXT(name, SND_SOC_NOPM, shift, 1, 0, \
+		       snd_soc_dapm_get_volsw, rt5670_put_dac1_mix_dac1_switch)
+
 static const struct snd_kcontrol_new rt5670_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
 			RT5670_M_ADCMIX_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
-			RT5670_M_DAC1_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 0),
 };
 
 static const struct snd_kcontrol_new rt5670_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
 			RT5670_M_ADCMIX_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
-			RT5670_M_DAC1_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 1),
 };
 
 static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = {
@@ -2993,6 +3071,16 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 		dev_info(&i2c->dev, "quirk JD mode 3\n");
 	}
 
+	/*
+	 * Enable the emulated "DAC1 Playback Switch" by default to avoid
+	 * muting the output with older UCM profiles.
+	 */
+	rt5670->dac1_playback_switch_l = true;
+	rt5670->dac1_playback_switch_r = true;
+	/* The Power-On-Reset values for the DAC1 mixer have the DAC1 input enabled. */
+	rt5670->dac1_mixl_dac1_switch = true;
+	rt5670->dac1_mixr_dac1_switch = true;
+
 	rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
 	if (IS_ERR(rt5670->regmap)) {
 		ret = PTR_ERR(rt5670->regmap);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index f9c4db156c80..6fb3c369ee98 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -2013,6 +2013,11 @@ struct rt5670_priv {
 	int dsp_rate;
 	int jack_type;
 	int jack_type_saved;
+
+	bool dac1_mixl_dac1_switch;
+	bool dac1_mixr_dac1_switch;
+	bool dac1_playback_switch_l;
+	bool dac1_playback_switch_r;
 };
 
 void rt5670_jack_suspend(struct snd_soc_component *component);
-- 
cgit 


From 4ec5b96775a88dd9b1c3ba1d23c43c478cab95a2 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Wed, 24 Feb 2021 14:57:51 +0800
Subject: ASoC: ak4458: Add MODULE_DEVICE_TABLE

Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.

Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ak4458.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 472caad17012..85a1d00894a9 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -812,6 +812,7 @@ static const struct of_device_id ak4458_of_match[] = {
 	{ .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata},
 	{ },
 };
+MODULE_DEVICE_TABLE(of, ak4458_of_match);
 
 static struct i2c_driver ak4458_i2c_driver = {
 	.driver = {
-- 
cgit 


From 80cffd2468ddb850e678f17841fc356930b2304a Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Wed, 24 Feb 2021 14:57:52 +0800
Subject: ASoC: ak5558: Add MODULE_DEVICE_TABLE

Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.

Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ak5558.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index 8a32b0139cb0..85bdd0534180 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -419,6 +419,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = {
 	{ .compatible = "asahi-kasei,ak5558"},
 	{ }
 };
+MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids);
 
 static struct i2c_driver ak5558_i2c_driver = {
 	.driver = {
-- 
cgit 


From ca08ddfd961d2a17208d9182e0ee5791b39bd8bf Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Wed, 24 Feb 2021 11:50:52 +0100
Subject: ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current
 threshold

When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the
byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX
models since these use almost the same settings.

While doing this I accidentally also copied and kept the non-standard
OVCD_TH_1500UA setting used on those models. This too low threshold is
causing headsets to often be seen as headphones (without a headset-mic)
and when correctly identified it is causing ghost play/pause
button-presses to get detected.

Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA
setting, fixing these problems.

Fixes: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bytcr_rt5640.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 782f2b4d72ad..5d48cc359c3d 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
 					BYT_RT5640_JD_SRC_JD1_IN4P |
-					BYT_RT5640_OVCD_TH_1500UA |
+					BYT_RT5640_OVCD_TH_2000UA |
 					BYT_RT5640_OVCD_SF_0P75 |
 					BYT_RT5640_MCLK_EN),
 	},
-- 
cgit 


From cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Fri, 26 Feb 2021 15:38:13 +0100
Subject: ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor
 of 10

The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.

E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.

Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.

Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5640.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 1414ad15d01c..a5674c227b3a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg)
 }
 
 static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
-- 
cgit 


From eee51df776bd6cac10a76b2779a9fdee3f622b2b Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Fri, 26 Feb 2021 15:38:14 +0100
Subject: ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor
 of 10

The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.

E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.

Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.

Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5651.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index d198e191fb0c..e59fdc81dbd4 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg)
 }
 
 static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
-- 
cgit 


From f86f58e3594fb0ab1993d833d3b9a2496f3c928c Mon Sep 17 00:00:00 2001
From: Benjamin Rood <benjaminjrood@gmail.com>
Date: Fri, 19 Feb 2021 13:33:08 -0500
Subject: ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on
 probe

According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has
the following bit field definitions:

| BITS  | FIELD       | RW | RESET | DEFINITION                        |
| 15    | RSVD        | RO | 0x0   | Reserved                          |
| 14    | RSVD        | RW | 0x1   | Reserved                          |
| 13:12 | MAX_GAIN    | RW | 0x1   | Max Gain of AVC in expander mode  |
| 11:10 | RSVD        | RO | 0x0   | Reserved                          |
| 9:8   | LBI_RESP    | RW | 0x1   | Integrator Response               |
| 7:6   | RSVD        | RO | 0x0   | Reserved                          |
| 5     | HARD_LMT_EN | RW | 0x0   | Enable hard limiter mode          |
| 4:1   | RSVD        | RO | 0x0   | Reserved                          |
| 0     | EN          | RW | 0x0   | Enable/Disable AVC                |

The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510.  This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED.  It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0.  While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.

The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'.  Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.

Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.

[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf

Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 73551e36695e..6d9bb256a2cf 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_DAP_EQ_BASS_BAND4,		0x002f },
 	{ SGTL5000_DAP_MAIN_CHAN,		0x8000 },
 	{ SGTL5000_DAP_MIX_CHAN,		0x0000 },
-	{ SGTL5000_DAP_AVC_CTRL,		0x0510 },
+	{ SGTL5000_DAP_AVC_CTRL,		0x5100 },
 	{ SGTL5000_DAP_AVC_THRESHOLD,		0x1473 },
 	{ SGTL5000_DAP_AVC_ATTACK,		0x0028 },
 	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
-- 
cgit 


From bb18c678754ce1514100fb4c0bf6113b5af36c48 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Sun, 28 Feb 2021 17:04:41 +0100
Subject: ASoC: es8316: Simplify adc_pga_gain_tlv table

Most steps in this table are steps of 3dB (300 centi-dB), so we can
simplify the table.

This not only reduces the amount of space it takes inside the kernel,
this also makes alsa-lib's mixer code actually accept the table, where
as before this change alsa-lib saw the "ADC PGA Gain" control as a
control without a dB scale.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 9 ++-------
 1 file changed, 2 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index d632055370e0..067757d1d70a 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
 	1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
 	2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
 	3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
-	4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
-	5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
-	6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
-	7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
-	8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
-	9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
-	10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
+	4, 7, TLV_DB_SCALE_ITEM(700, 300, 0),
+	8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0),
 );
 
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
-- 
cgit 


From 5bb0ecddb2a7f638d65e457f3da9fa334c967b14 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 1 Mar 2021 18:34:10 -0600
Subject: ASoC: SOF: Intel: unregister DMIC device on probe error

We only unregister the platform device during the .remove operation,
but if the probe fails we will never reach this sequence.

Suggested-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Fixes: dd96daca6c83e ("ASoC: SOF: Intel: Add APL/CNL HW DSP support")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 1d29b1fd6a94..0c096db07322 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -897,6 +897,7 @@ free_streams:
 /* dsp_unmap: not currently used */
 	iounmap(sdev->bar[HDA_DSP_BAR]);
 hdac_bus_unmap:
+	platform_device_unregister(hdev->dmic_dev);
 	iounmap(bus->remap_addr);
 	hda_codec_i915_exit(sdev);
 err:
-- 
cgit 


From 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd Mon Sep 17 00:00:00 2001
From: Jon Hunter <jonathanh@nvidia.com>
Date: Wed, 3 Mar 2021 11:55:26 +0000
Subject: ASoC: soc-core: Prevent warning if no DMI table is present

Many systems do not use ACPI and hence do not provide a DMI table. On
non-ACPI systems a warning, such as the following, is printed on boot.

 WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name!

The variable 'dmi_available' is not exported and so currently cannot be
used by kernel modules without adding an accessor. However, it is
possible to use the function is_acpi_device_node() to determine if the
sound card is an ACPI device and hence indicate if we expect a DMI table
to be present. Therefore, call is_acpi_device_node() to see if we are
using ACPI and only parse the DMI table if we are booting with ACPI.

Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f6d4e99b590c..0cffc9527e28 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -31,6 +31,7 @@
 #include <linux/of.h>
 #include <linux/of_graph.h>
 #include <linux/dmi.h>
+#include <linux/acpi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 	if (card->long_name)
 		return 0; /* long name already set by driver or from DMI */
 
+	if (!is_acpi_device_node(card->dev->fwnode))
+		return 0;
+
 	/* make up dmi long name as: vendor-product-version-board */
 	vendor = dmi_get_system_info(DMI_BOARD_VENDOR);
 	if (!vendor || !is_dmi_valid(vendor)) {
-- 
cgit 


From e793c965519b8b7f2fea51a48398405e2a501729 Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Fri, 5 Mar 2021 17:34:28 +0000
Subject: ASoC: cs42l42: Fix Bitclock polarity inversion

The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 20 ++++++++------------
 sound/soc/codecs/cs42l42.h | 11 ++++++-----
 2 files changed, 14 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 210fcbedf241..df0d5fec0287 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* Bitclock/frame inversion */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	}
 
-	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG,
-				CS42L42_ASP_MODE_MASK |
-				CS42L42_ASP_SCPOL_IN_DAC_MASK |
-				CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val);
+	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK |
+								      CS42L42_ASP_SCPOL_MASK |
+								      CS42L42_ASP_LCPOL_MASK,
+								      asp_cfg_val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9e3cc528dcff..1f0d67c95a9a 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -258,11 +258,12 @@
 #define CS42L42_ASP_SLAVE_MODE		0x00
 #define CS42L42_ASP_MODE_SHIFT		4
 #define CS42L42_ASP_MODE_MASK		(1 << CS42L42_ASP_MODE_SHIFT)
-#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT	2
-#define CS42L42_ASP_SCPOL_IN_DAC_MASK	(1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT)
-#define CS42L42_ASP_LCPOL_IN_SHIFT	0
-#define CS42L42_ASP_LCPOL_IN_MASK	(1 << CS42L42_ASP_LCPOL_IN_SHIFT)
-#define CS42L42_ASP_POL_INV		1
+#define CS42L42_ASP_SCPOL_SHIFT		2
+#define CS42L42_ASP_SCPOL_MASK		(3 << CS42L42_ASP_SCPOL_SHIFT)
+#define CS42L42_ASP_SCPOL_NOR		3
+#define CS42L42_ASP_LCPOL_SHIFT		0
+#define CS42L42_ASP_LCPOL_MASK		(3 << CS42L42_ASP_LCPOL_SHIFT)
+#define CS42L42_ASP_LCPOL_INV		3
 
 #define CS42L42_ASP_FRM_CFG		(CS42L42_PAGE_12 + 0x08)
 #define CS42L42_ASP_STP_SHIFT		4
-- 
cgit 


From 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Fri, 5 Mar 2021 17:34:29 +0000
Subject: ASoC: cs42l42: Fix channel width support

Remove the hard coded 32 bits width and replace with the correct width
calculated by params_width.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++++++------------------------
 sound/soc/codecs/cs42l42.h |  1 -
 2 files changed, 22 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index df0d5fec0287..4f9ad9547929 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					CS42L42_CLK_OASRC_SEL_MASK,
 					CS42L42_CLK_OASRC_SEL_12 <<
 					CS42L42_CLK_OASRC_SEL_SHIFT);
-			/* channel 1 on low LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH1_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_LOW <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
-			/* Channel 2 on high LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH2_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_HI <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
 			if (pll_ratio_table[i].mclk_src_sel == 0) {
 				/* Pass the clock straight through */
 				snd_soc_component_update_bits(component,
@@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int retval;
+	unsigned int width = (params_width(params) / 8) - 1;
+	unsigned int val = 0;
 
 	cs42l42->srate = params_rate(params);
-	cs42l42->swidth = params_width(params);
 
-	retval = cs42l42_pll_config(component);
+	switch(substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
+		/* channel 1 on low LRCLK */
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		/* Channel 2 on high LRCLK */
+		val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT;
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		break;
+	default:
+		break;
+	}
 
-	return retval;
+	return cs42l42_pll_config(component);
 }
 
 static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
@@ -896,9 +893,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction)
 	return 0;
 }
 
-#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
-			SNDRV_PCM_FMTBIT_S32_LE)
+#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE |\
+			 SNDRV_PCM_FMTBIT_S32_LE )
 
 
 static const struct snd_soc_dai_ops cs42l42_ops = {
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 1f0d67c95a9a..9b017b76828a 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -757,7 +757,6 @@ struct  cs42l42_private {
 	struct completion pdn_done;
 	u32 sclk;
 	u32 srate;
-	u32 swidth;
 	u8 plug_state;
 	u8 hs_type;
 	u8 ts_inv;
-- 
cgit 


From 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Fri, 5 Mar 2021 17:34:30 +0000
Subject: ASoC: cs42l42: Fix mixer volume control

The minimum value is 0x3f (-63dB), which also is mute

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 4f9ad9547929..d5078ce79fad 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = {
 };
 
 static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
-static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false);
+static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
 
 static const char * const cs42l42_hpf_freq_text[] = {
 	"1.86Hz", "120Hz", "235Hz", "466Hz"
@@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				CS42L42_DAC_HPF_EN_SHIFT, true, false),
 	SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL,
 			 CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT,
-				0x3e, 1, mixer_tlv)
+				0x3f, 1, mixer_tlv)
 };
 
 static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
-- 
cgit 


From 15013240fcf791691f99c884802099db34c099b9 Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Fri, 5 Mar 2021 17:34:31 +0000
Subject: ASoC: cs42l42: Don't enable/disable regulator at Bias Level

dev_pm_ops already enable/disable the codec if not in use

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-5-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 38 --------------------------------------
 1 file changed, 38 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index d5078ce79fad..eee3fc320030 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -511,43 +511,6 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
 	{"HP", NULL, "HPDRV"}
 };
 
-static int cs42l42_set_bias_level(struct snd_soc_component *component,
-					enum snd_soc_bias_level level)
-{
-	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int ret;
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
-			regcache_cache_only(cs42l42->regmap, false);
-			regcache_sync(cs42l42->regmap);
-			ret = regulator_bulk_enable(
-						ARRAY_SIZE(cs42l42->supplies),
-						cs42l42->supplies);
-			if (ret != 0) {
-				dev_err(component->dev,
-					"Failed to enable regulators: %d\n",
-					ret);
-				return ret;
-			}
-		}
-		break;
-	case SND_SOC_BIAS_OFF:
-
-		regcache_cache_only(cs42l42->regmap, true);
-		regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies),
-						    cs42l42->supplies);
-		break;
-	}
-
-	return 0;
-}
-
 static int cs42l42_component_probe(struct snd_soc_component *component)
 {
 	struct cs42l42_private *cs42l42 =
@@ -560,7 +523,6 @@ static int cs42l42_component_probe(struct snd_soc_component *component)
 
 static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
 	.probe			= cs42l42_component_probe,
-	.set_bias_level		= cs42l42_set_bias_level,
 	.dapm_widgets		= cs42l42_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(cs42l42_dapm_widgets),
 	.dapm_routes		= cs42l42_audio_map,
-- 
cgit 


From 19325cfea04446bc79b36bffd4978af15f46a00e Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Fri, 5 Mar 2021 17:34:32 +0000
Subject: ASoC: cs42l42: Always wait at least 3ms after reset

This delay is part of the power-up sequence defined in the datasheet.
A runtime_resume is a power-up so must also include the delay.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 3 ++-
 sound/soc/codecs/cs42l42.h | 1 +
 2 files changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eee3fc320030..811b7b1c9732 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1756,7 +1756,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 		dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
 		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 	}
-	mdelay(3);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	/* Request IRQ */
 	ret = devm_request_threaded_irq(&i2c_client->dev,
@@ -1881,6 +1881,7 @@ static int cs42l42_runtime_resume(struct device *dev)
 	}
 
 	gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	regcache_cache_only(cs42l42->regmap, false);
 	regcache_sync(cs42l42->regmap);
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9b017b76828a..866d7c873e3c 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -740,6 +740,7 @@
 #define CS42L42_FRAC2_VAL(val)	(((val) & 0xff0000) >> 16)
 
 #define CS42L42_NUM_SUPPLIES	5
+#define CS42L42_BOOT_TIME_US	3000
 
 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
 	"VA",
-- 
cgit 


From 5346f0e80b7160c91fb599d4545fd12560c286ed Mon Sep 17 00:00:00 2001
From: Jonathan Marek <jonathan@marek.ca>
Date: Thu, 4 Mar 2021 16:56:46 -0500
Subject: ASoC: codecs: lpass-va-macro: mute/unmute all active decimators

An interface can have multiple decimators enabled, so loop over all active
decimators. Otherwise only one channel will be unmuted, and other channels
will be zero. This fixes recording from dual DMIC as a single two channel
stream.

Also remove the now unused "active_decimator" field.

Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lpass-va-macro.c | 28 +++++++++++++---------------
 1 file changed, 13 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c
index 91e6890d6efc..3d6976a3d9e4 100644
--- a/sound/soc/codecs/lpass-va-macro.c
+++ b/sound/soc/codecs/lpass-va-macro.c
@@ -189,7 +189,6 @@ struct va_macro {
 	struct device *dev;
 	unsigned long active_ch_mask[VA_MACRO_MAX_DAIS];
 	unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS];
-	unsigned long active_decimator[VA_MACRO_MAX_DAIS];
 	u16 dmic_clk_div;
 
 	int dec_mode[VA_MACRO_NUM_DECIMATORS];
@@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol,
 	if (enable) {
 		set_bit(dec_id, &va->active_ch_mask[dai_id]);
 		va->active_ch_cnt[dai_id]++;
-		va->active_decimator[dai_id] = dec_id;
 	} else {
 		clear_bit(dec_id, &va->active_ch_mask[dai_id]);
 		va->active_ch_cnt[dai_id]--;
-		va->active_decimator[dai_id] = -1;
 	}
 
 	snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update);
@@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 	struct va_macro *va = snd_soc_component_get_drvdata(component);
 	u16 tx_vol_ctl_reg, decimator;
 
-	decimator = va->active_decimator[dai->id];
-
-	tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
-				VA_MACRO_TX_PATH_OFFSET * decimator;
-	if (mute)
-		snd_soc_component_update_bits(component, tx_vol_ctl_reg,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN);
-	else
-		snd_soc_component_update_bits(component, tx_vol_ctl_reg,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
-					      CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+	for_each_set_bit(decimator, &va->active_ch_mask[dai->id],
+			 VA_MACRO_DEC_MAX) {
+		tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
+					VA_MACRO_TX_PATH_OFFSET * decimator;
+		if (mute)
+			snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+					CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+					CDC_VA_TX_PATH_PGA_MUTE_EN);
+		else
+			snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+					CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+					CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+	}
 
 	return 0;
 }
-- 
cgit 


From e4b8b7c916038c1ffcba2c4ce92d5523c4cc2f46 Mon Sep 17 00:00:00 2001
From: Jonathan Marek <jonathan@marek.ca>
Date: Thu, 4 Mar 2021 19:50:48 -0500
Subject: ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls

Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these
controls are incorrectly toggling the first bit of the register, which
is part of the FS_RATE field.

Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX"
control, which is to use SND_SOC_NOPM as the register and use an enum in
the shift field instead.

Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lpass-wsa-macro.c | 20 +++++++++++---------
 1 file changed, 11 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index 5ebcd935ba89..9ca49a165f69 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
 				     struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	u16 gain_reg;
+	u16 path_reg, gain_reg;
 	int val;
 
-	switch (w->reg) {
-	case CDC_WSA_RX0_RX_PATH_MIX_CTL:
+	switch (w->shift) {
+	case WSA_MACRO_RX_MIX0:
+		path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL;
 		gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL;
 		break;
-	case CDC_WSA_RX1_RX_PATH_MIX_CTL:
+	case WSA_MACRO_RX_MIX1:
+		path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL;
 		gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL;
 		break;
 	default:
@@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
 		snd_soc_component_write(component, gain_reg, val);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
-		snd_soc_component_update_bits(component, w->reg,
+		snd_soc_component_update_bits(component, path_reg,
 					      CDC_WSA_RX_PATH_MIX_CLK_EN_MASK,
 					      CDC_WSA_RX_PATH_MIX_CLK_DISABLE);
 		break;
@@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux),
 	SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux),
 	SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux),
-	SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL,
-			   0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path,
+	SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0,
+			   0, &rx0_mix_mux, wsa_macro_enable_mix_path,
 			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux),
-	SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL,
-			   0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path,
+	SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1,
+			   0, &rx1_mix_mux, wsa_macro_enable_mix_path,
 			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
 	SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0,
-- 
cgit 


From ac101985cad3912e484295bd0ec22d117fee9f17 Mon Sep 17 00:00:00 2001
From: Peter Robinson <pbrobinson@gmail.com>
Date: Sun, 7 Mar 2021 16:23:37 +0000
Subject: ASoC: remove remnants of sirf prima/atlas audio codec

In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans
up a stray header and some Kconfig entries for the codec that
were missed in the process.

Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers)
Signed-off-by: Peter Robinson <pbrobinson@gmail.com>
Cc: Arnd Bergmann <arnd@arndb.de>
Cc: Mark Brown <broonie@kernel.org>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig            |   5 --
 sound/soc/codecs/sirf-audio-codec.h | 124 ------------------------------------
 2 files changed, 129 deletions(-)
 delete mode 100644 sound/soc/codecs/sirf-audio-codec.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e4cf14e66a51..1c87b42606c9 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -186,7 +186,6 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_SI476X
 	imply SND_SOC_SIMPLE_AMPLIFIER
 	imply SND_SOC_SIMPLE_MUX
-	imply SND_SOC_SIRF_AUDIO_CODEC
 	imply SND_SOC_SPDIF
 	imply SND_SOC_SSM2305
 	imply SND_SOC_SSM2518
@@ -1279,10 +1278,6 @@ config SND_SOC_SIMPLE_MUX
 	tristate "Simple Audio Mux"
 	select GPIOLIB
 
-config SND_SOC_SIRF_AUDIO_CODEC
-	tristate "SiRF SoC internal audio codec"
-	select REGMAP_MMIO
-
 config SND_SOC_SPDIF
 	tristate "S/PDIF CODEC"
 
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
deleted file mode 100644
index a7fe2680f4c7..000000000000
--- a/sound/soc/codecs/sirf-audio-codec.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-or-later */
-/*
- * SiRF inner codec controllers define
- *
- * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
- */
-
-#ifndef _SIRF_AUDIO_CODEC_H
-#define _SIRF_AUDIO_CODEC_H
-
-
-#define AUDIO_IC_CODEC_PWR			(0x00E0)
-#define AUDIO_IC_CODEC_CTRL0			(0x00E4)
-#define AUDIO_IC_CODEC_CTRL1			(0x00E8)
-#define AUDIO_IC_CODEC_CTRL2			(0x00EC)
-#define AUDIO_IC_CODEC_CTRL3			(0x00F0)
-
-#define MICBIASEN		(1 << 3)
-
-#define IC_RDACEN		(1 << 0)
-#define IC_LDACEN		(1 << 1)
-#define IC_HSREN		(1 << 2)
-#define IC_HSLEN		(1 << 3)
-#define IC_SPEN			(1 << 4)
-#define IC_CPEN			(1 << 5)
-
-#define IC_HPRSELR		(1 << 6)
-#define IC_HPLSELR		(1 << 7)
-#define IC_HPRSELL		(1 << 8)
-#define IC_HPLSELL		(1 << 9)
-#define IC_SPSELR		(1 << 10)
-#define IC_SPSELL		(1 << 11)
-
-#define IC_MONOR		(1 << 12)
-#define IC_MONOL		(1 << 13)
-
-#define IC_RXOSRSEL		(1 << 28)
-#define IC_CPFREQ		(1 << 29)
-#define IC_HSINVEN		(1 << 30)
-
-#define IC_MICINREN		(1 << 0)
-#define IC_MICINLEN		(1 << 1)
-#define IC_MICIN1SEL		(1 << 2)
-#define IC_MICIN2SEL		(1 << 3)
-#define IC_MICDIFSEL		(1 << 4)
-#define	IC_LINEIN1SEL		(1 << 5)
-#define	IC_LINEIN2SEL		(1 << 6)
-#define	IC_RADCEN		(1 << 7)
-#define	IC_LADCEN		(1 << 8)
-#define	IC_ALM			(1 << 9)
-
-#define IC_DIGMICEN             (1 << 22)
-#define IC_DIGMICFREQ           (1 << 23)
-#define IC_ADC14B_12            (1 << 24)
-#define IC_FIRDAC_HSL_EN        (1 << 25)
-#define IC_FIRDAC_HSR_EN        (1 << 26)
-#define IC_FIRDAC_LOUT_EN       (1 << 27)
-#define IC_POR                  (1 << 28)
-#define IC_CODEC_CLK_EN         (1 << 29)
-#define IC_HP_3DB_BOOST         (1 << 30)
-
-#define IC_ADC_LEFT_GAIN_SHIFT	16
-#define IC_ADC_RIGHT_GAIN_SHIFT 10
-#define IC_ADC_GAIN_MASK	0x3F
-#define IC_MIC_MAX_GAIN		0x39
-
-#define IC_RXPGAR_MASK		0x3F
-#define IC_RXPGAR_SHIFT		14
-#define IC_RXPGAL_MASK		0x3F
-#define IC_RXPGAL_SHIFT		21
-#define IC_RXPGAR		0x7B
-#define IC_RXPGAL		0x7B
-
-#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK     0x3F
-#define AUDIO_PORT_TX_FIFO_SC_OFFSET    0
-#define AUDIO_PORT_TX_FIFO_LC_OFFSET    10
-#define AUDIO_PORT_TX_FIFO_HC_OFFSET    20
-
-#define TX_FIFO_SC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_SC_OFFSET)
-#define TX_FIFO_LC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_LC_OFFSET)
-#define TX_FIFO_HC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_HC_OFFSET)
-
-#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK     0x0F
-#define AUDIO_PORT_RX_FIFO_SC_OFFSET    0
-#define AUDIO_PORT_RX_FIFO_LC_OFFSET    10
-#define AUDIO_PORT_RX_FIFO_HC_OFFSET    20
-
-#define RX_FIFO_SC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_SC_OFFSET)
-#define RX_FIFO_LC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_LC_OFFSET)
-#define RX_FIFO_HC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_HC_OFFSET)
-#define AUDIO_PORT_IC_CODEC_TX_CTRL		(0x00F4)
-#define AUDIO_PORT_IC_CODEC_RX_CTRL		(0x00F8)
-
-#define AUDIO_PORT_IC_TXFIFO_OP			(0x00FC)
-#define AUDIO_PORT_IC_TXFIFO_LEV_CHK		(0x0100)
-#define AUDIO_PORT_IC_TXFIFO_STS		(0x0104)
-#define AUDIO_PORT_IC_TXFIFO_INT		(0x0108)
-#define AUDIO_PORT_IC_TXFIFO_INT_MSK		(0x010C)
-
-#define AUDIO_PORT_IC_RXFIFO_OP			(0x0110)
-#define AUDIO_PORT_IC_RXFIFO_LEV_CHK		(0x0114)
-#define AUDIO_PORT_IC_RXFIFO_STS		(0x0118)
-#define AUDIO_PORT_IC_RXFIFO_INT		(0x011C)
-#define AUDIO_PORT_IC_RXFIFO_INT_MSK		(0x0120)
-
-#define AUDIO_FIFO_START		(1 << 0)
-#define AUDIO_FIFO_RESET		(1 << 1)
-
-#define AUDIO_FIFO_FULL			(1 << 0)
-#define AUDIO_FIFO_EMPTY		(1 << 1)
-#define AUDIO_FIFO_OFLOW		(1 << 2)
-#define AUDIO_FIFO_UFLOW		(1 << 3)
-
-#define IC_TX_ENABLE		(0x03)
-#define IC_RX_ENABLE_MONO	(0x01)
-#define IC_RX_ENABLE_STEREO	(0x03)
-
-#endif /*__SIRF_AUDIO_CODEC_H*/
-- 
cgit 


From 1c668e1c0a0f74472469cd514f40c9012b324c31 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Tue, 9 Mar 2021 14:21:27 +0000
Subject: ASoC: qcom: sdm845: Fix array out of bounds access

Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.

Fix this by properly using AFE_PORT_MAX macro.

Fixes: 1b93a8843147 ("ASoC: qcom: sdm845: handle soundwire stream")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/sdm845.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 6c2760e27ea6..1e2c2d0902ea 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -33,12 +33,12 @@
 struct sdm845_snd_data {
 	struct snd_soc_jack jack;
 	bool jack_setup;
-	bool stream_prepared[SLIM_MAX_RX_PORTS];
+	bool stream_prepared[AFE_PORT_MAX];
 	struct snd_soc_card *card;
 	uint32_t pri_mi2s_clk_count;
 	uint32_t sec_mi2s_clk_count;
 	uint32_t quat_tdm_clk_count;
-	struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
+	struct sdw_stream_runtime *sruntime[AFE_PORT_MAX];
 };
 
 static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
-- 
cgit 


From 4800fe6ea1022eb240215b1743d2541adad8efc7 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Tue, 9 Mar 2021 14:21:28 +0000
Subject: ASoC: qcom: sdm845: Fix array out of range on rx slim channels

WCD934x has only 13 RX SLIM ports however we are setting it as 16
in set_channel_map, this will lead to array out of bounds error!

Orignally caught by enabling USBAN array out of bounds check:

Fixes: 5caf64c633a3 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/sdm845.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 1e2c2d0902ea..153e9b2de0b5 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -27,7 +27,7 @@
 #define SPK_TDM_RX_MASK         0x03
 #define NUM_TDM_SLOTS           8
 #define SLIM_MAX_TX_PORTS 16
-#define SLIM_MAX_RX_PORTS 16
+#define SLIM_MAX_RX_PORTS 13
 #define WCD934X_DEFAULT_MCLK_RATE	9600000
 
 struct sdm845_snd_data {
-- 
cgit 


From 3bb4852d598f0275ed5996a059df55be7318ac2f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Tue, 9 Mar 2021 14:21:29 +0000
Subject: ASoC: codecs: wcd934x: add a sanity check in set channel map

set channel map can be passed with a channel maps, however if
the number of channels that are passed are more than the actual
supported channels then we would be accessing array out of bounds.

So add a sanity check to validate these numbers!

Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd934x.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 40f682f5dab8..d18ae5e3ee80 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -1873,6 +1873,12 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai,
 
 	wcd = snd_soc_component_get_drvdata(dai->component);
 
+	if (tx_num > WCD934X_TX_MAX || rx_num > WCD934X_RX_MAX) {
+		dev_err(wcd->dev, "Invalid tx %d or rx %d channel count\n",
+			tx_num, rx_num);
+		return -EINVAL;
+	}
+
 	if (!tx_slot || !rx_slot) {
 		dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n",
 			tx_slot, rx_slot);
-- 
cgit 


From fd8299181995093948ec6ca75432e797b4a39143 Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Mon, 8 Mar 2021 18:41:27 -0600
Subject: ASoC: SOF: intel: fix wrong poll bits in dsp power down

The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered
down, we need to check ADSPCS_CPA, the Current Power Active bit.

Fixes: 747503b1813a3 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations")
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dsp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 5788fe356960..c3b757cf01a0 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -207,7 +207,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask)
 
 	ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
 				HDA_DSP_REG_ADSPCS, adspcs,
-				!(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)),
+				!(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)),
 				HDA_DSP_REG_POLL_INTERVAL_US,
 				HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC);
 	if (ret < 0)
-- 
cgit 


From 9922f50f7178496e709d3d064920b5031f0d9061 Mon Sep 17 00:00:00 2001
From: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Date: Thu, 11 Mar 2021 21:15:57 +0530
Subject: ASoC: qcom: lpass-cpu: Fix lpass dai ids parse

The max boundary check while parsing dai ids makes
sound card registration fail after common up dai ids.

Fixes: cd3484f7f138 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY")

Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/lpass-cpu.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index c642e5f8f28c..be360a402b67 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev,
 
 	for_each_child_of_node(dev->of_node, node) {
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id < 0 || id >= data->variant->num_dai) {
+		if (ret || id < 0) {
 			dev_err(dev, "valid dai id not found: %d\n", ret);
 			continue;
 		}
-- 
cgit 


From 2bf44e0ee95f39cc54ea1b942f0a027e0181ca4e Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 12 Mar 2021 12:14:08 +0800
Subject: ALSA: hda: generic: Fix the micmute led init state

Recently we found the micmute led init state is not correct after
freshly installing the ubuntu linux on a Lenovo AIO machine. The
internal mic is not muted, but the micmute led is on and led mode is
'follow mute'. If we mute internal mic, the led is keeping on, then
unmute the internal mic, the led is off. And from then on, the
micmute led will work correctly.

So the micmute led init state is not correct. The led is controlled
by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the
patch_realtek, the gpio data is set to 0x4 initially and the led is
on with this data. In the hda_generic, the led_value is set to
0 initially, suppose users set the 'capture switch' to on from
user space and the micmute led should change to be off with this
operation, but the check "if (val == spec->micmute_led.led_value)" in
the call_micmute_led_update() will skip the led setting.

To guarantee the led state will be set by the 1st time of changing
"Capture Switch", set -1 to the init led_value.

Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210312041408.3776-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8b7c5508f368..f5cba7afd1c6 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -4065,7 +4065,7 @@ static int add_micmute_led_hook(struct hda_codec *codec)
 
 	spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE;
 	spec->micmute_led.capture = 0;
-	spec->micmute_led.led_value = 0;
+	spec->micmute_led.led_value = -1;
 	spec->micmute_led.old_hook = spec->cap_sync_hook;
 	spec->cap_sync_hook = update_micmute_led;
 	if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl))
-- 
cgit 


From dd7b836d6bc935df95c826f69ff4d051f5561604 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 12 Mar 2021 18:34:07 +0900
Subject: ALSA: dice: fix null pointer dereference when node is disconnected

When node is removed from IEEE 1394 bus, any transaction fails to the node.
In the case, ALSA dice driver doesn't stop isochronous contexts even if
they are running. As a result, null pointer dereference occurs in callback
from the running context.

This commit fixes the bug to release isochronous contexts always.

Cc: <stable@vger.kernel.org> # v5.4 or later
Fixes: e9f21129b8d8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210312093407.23437-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-stream.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 8e0c0380b4c4..1a14c083e8ce 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -493,11 +493,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice)
 	struct reg_params tx_params, rx_params;
 
 	if (dice->substreams_counter == 0) {
-		if (get_register_params(dice, &tx_params, &rx_params) >= 0) {
-			amdtp_domain_stop(&dice->domain);
+		if (get_register_params(dice, &tx_params, &rx_params) >= 0)
 			finish_session(dice, &tx_params, &rx_params);
-		}
 
+		amdtp_domain_stop(&dice->domain);
 		release_resources(dice);
 	}
 }
-- 
cgit 


From 8d06b9633a66f41fed520f6eebd163189518ba79 Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Fri, 12 Mar 2021 10:26:45 +0800
Subject: ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge

This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3].

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1615516005-781-1-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +++-
 sound/soc/mediatek/mt8192/mt8192-reg.h     | 8 +++++---
 2 files changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
index f5de1d769679..f3bebed2428a 100644
--- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
@@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 
 	/* set tdm */
 	if (tdm_priv->bck_invert)
-		tdm_con |= 1 << BCK_INVERSE_SFT;
+		regmap_update_bits(afe->regmap, AUDIO_TOP_CON3,
+				   BCK_INVERSE_MASK_SFT,
+				   0x1 << BCK_INVERSE_SFT);
 
 	if (tdm_priv->lck_invert)
 		tdm_con |= 1 << LRCK_INVERSE_SFT;
diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h
index 562f25c79c34..b9fb80d4afec 100644
--- a/sound/soc/mediatek/mt8192/mt8192-reg.h
+++ b/sound/soc/mediatek/mt8192/mt8192-reg.h
@@ -21,6 +21,11 @@ enum {
 /*****************************************************************************
  *                  R E G I S T E R       D E F I N I T I O N
  *****************************************************************************/
+/* AUDIO_TOP_CON3 */
+#define BCK_INVERSE_SFT                              3
+#define BCK_INVERSE_MASK                             0x1
+#define BCK_INVERSE_MASK_SFT                         (0x1 << 3)
+
 /* AFE_DAC_CON0 */
 #define VUL12_ON_SFT                                   31
 #define VUL12_ON_MASK                                  0x1
@@ -2079,9 +2084,6 @@ enum {
 #define TDM_EN_SFT                                     0
 #define TDM_EN_MASK                                    0x1
 #define TDM_EN_MASK_SFT                                (0x1 << 0)
-#define BCK_INVERSE_SFT                                1
-#define BCK_INVERSE_MASK                               0x1
-#define BCK_INVERSE_MASK_SFT                           (0x1 << 1)
 #define LRCK_INVERSE_SFT                               2
 #define LRCK_INVERSE_MASK                              0x1
 #define LRCK_INVERSE_MASK_SFT                          (0x1 << 2)
-- 
cgit 


From e1c86210fe27428399643861b81b080eccd79f87 Mon Sep 17 00:00:00 2001
From: Xiaoliang Yu <yxl_22@outlook.com>
Date: Sat, 13 Mar 2021 07:54:53 +0800
Subject: ALSA: hda/realtek: Apply headset-mic quirks for Xiaomi Redmibook Air

There is another fix for headset-mic problem on Redmibook (1d72:1602),
it also works on Redmibook Air (1d72:1947), which has the same issue.

Signed-off-by: Xiaoliang Yu <yxl_22@outlook.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/TYBP286MB02856DC016849DEA0F9B6A37EE6F9@TYBP286MB0285.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b47504fa8dfd..c6d916177d5c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8243,6 +8243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
 	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
-- 
cgit 


From b95bc12e0412d14d5fc764f0b82631c7bcaf1959 Mon Sep 17 00:00:00 2001
From: Xiaoliang Yu <yxl_22@outlook.com>
Date: Tue, 16 Mar 2021 00:49:00 +0800
Subject: ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro

Built-in microphone and combojack on Xiaomi Notebook Pro (1d72:1701) needs
to be fixed, the existing quirk for Dell works well on that machine.

Signed-off-by: Xiaoliang Yu <yxl_22@outlook.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/OS0P286MB02749B9E13920E6899902CD8EE6C9@OS0P286MB0274.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c6d916177d5c..345a447aba81 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8242,6 +8242,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
-- 
cgit 


From ca6883393f0fa7f13ec8b860dbcef423a759c4a2 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Tue, 16 Mar 2021 14:54:50 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8

The HP EliteBook 840 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316065452.75659-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 345a447aba81..f06bcb1ac4f9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8048,6 +8048,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+	SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-- 
cgit 


From e7d66cf799390166e90f9a5715f2eede4fe06d51 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Tue, 16 Mar 2021 15:46:24 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8

The HP EliteBook 840 G8 Notebook PC is using ALC236 codec which is
using 0x02 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316074626.79895-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f06bcb1ac4f9..145d6cf4939e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4225,6 +4225,12 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
 	}
 }
 
+static void alc236_fixup_hp_gpio_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc_fixup_hp_gpio_led(codec, action, 0x02, 0x01);
+}
+
 static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -6381,6 +6387,7 @@ enum {
 	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
+	ALC236_FIXUP_HP_GPIO_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
 	ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
@@ -7616,6 +7623,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_mute_led,
 	},
+	[ALC236_FIXUP_HP_GPIO_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc236_fixup_hp_gpio_led,
+	},
 	[ALC236_FIXUP_HP_MUTE_LED] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc236_fixup_hp_mute_led,
@@ -8045,6 +8056,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation",
 		      ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
-- 
cgit 


From 8ca88d53351cc58d535b2bfc7386835378fb0db2 Mon Sep 17 00:00:00 2001
From: Sameer Pujar <spujar@nvidia.com>
Date: Mon, 15 Mar 2021 23:01:31 +0530
Subject: ASoC: simple-card-utils: Do not handle device clock

This reverts commit 1e30f642cf29 ("ASoC: simple-card-utils: Fix device
module clock"). The original patch ended up breaking following platform,
which depends on set_sysclk() to configure internal PLL on wm8904 codec
and expects simple-card-utils to not update the MCLK rate.
 - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"

It would be best if codec takes care of setting MCLK clock via DAI
set_sysclk() callback.

Reported-by: Michael Walle <michael@walle.cc>
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Fixes: 1e30f642cf29 ("ASoC: simple-card-utils: Fix device module clock")
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card-utils.c | 13 +++++++------
 1 file changed, 7 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ab31045cfc95..6cada4c1e283 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -172,15 +172,16 @@ int asoc_simple_parse_clk(struct device *dev,
 	 *  or device's module clock.
 	 */
 	clk = devm_get_clk_from_child(dev, node, NULL);
-	if (IS_ERR(clk))
-		clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
-
 	if (!IS_ERR(clk)) {
-		simple_dai->clk = clk;
 		simple_dai->sysclk = clk_get_rate(clk);
-	} else if (!of_property_read_u32(node, "system-clock-frequency",
-					 &val)) {
+
+		simple_dai->clk = clk;
+	} else if (!of_property_read_u32(node, "system-clock-frequency", &val)) {
 		simple_dai->sysclk = val;
+	} else {
+		clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
+		if (!IS_ERR(clk))
+			simple_dai->sysclk = clk_get_rate(clk);
 	}
 
 	if (of_property_read_bool(node, "system-clock-direction-out"))
-- 
cgit 


From dbf54a9534350d6aebbb34f5c1c606b81a4f35dd Mon Sep 17 00:00:00 2001
From: Sameer Pujar <spujar@nvidia.com>
Date: Mon, 15 Mar 2021 23:01:32 +0530
Subject: ASoC: rt5659: Update MCLK rate in set_sysclk()

Simple-card/audio-graph-card drivers do not handle MCLK clock when it
is specified in the codec device node. The expectation here is that,
the codec should actually own up the MCLK clock and do necessary setup
in the driver.

Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5659.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 41e5917b16a5..91a4ef7f620c 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int
 {
 	struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
 	unsigned int reg_val = 0;
+	int ret;
 
 	if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src)
 		return 0;
 
 	switch (clk_id) {
 	case RT5659_SCLK_S_MCLK:
+		ret = clk_set_rate(rt5659->mclk, freq);
+		if (ret)
+			return ret;
+
 		reg_val |= RT5659_SCLK_SRC_MCLK;
 		break;
 	case RT5659_SCLK_S_PLL1:
-- 
cgit 


From 899b12542b0897f92de9ba30944937c39ebb246d Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 16 Mar 2021 08:52:54 +0800
Subject: ASoC: rt711: add snd_soc_component remove callback

We do some IO operations in the snd_soc_component_set_jack callback
function and snd_soc_component_set_jack() will be called when soc
component is removed. However, we should not access SoundWire registers
when the bus is suspended.
So set regcache_cache_only(regmap, true) to avoid accessing in the
soc component removal process.

Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt711.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 85f744184a60..047f4e677d78 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component)
 	return 0;
 }
 
+static void rt711_remove(struct snd_soc_component *component)
+{
+	struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt711->regmap, true);
+}
+
 static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
 	.probe = rt711_probe,
 	.set_bias_level = rt711_set_bias_level,
@@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
 	.dapm_routes = rt711_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(rt711_audio_map),
 	.set_jack = rt711_set_jack_detect,
+	.remove = rt711_remove,
 };
 
 static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
-- 
cgit 


From 53b861bec737c189cc14ec3b5785d0f13445ac0f Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Tue, 16 Mar 2021 17:42:35 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8

The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316094236.89028-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 145d6cf4939e..316b9b4ccb32 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8060,6 +8060,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+	SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
-- 
cgit 


From 6417f03132a6952cd17ddd8eaddbac92b61b17e0 Mon Sep 17 00:00:00 2001
From: Leon Romanovsky <leonro@nvidia.com>
Date: Wed, 17 Mar 2021 12:45:47 +0200
Subject: module: remove never implemented MODULE_SUPPORTED_DEVICE

MODULE_SUPPORTED_DEVICE was added in pre-git era and never was
implemented. We can safely remove it, because the kernel has grown
to have many more reliable mechanisms to determine if device is
supported or not.

Signed-off-by: Leon Romanovsky <leonro@nvidia.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
---
 sound/drivers/aloop.c               |  1 -
 sound/drivers/dummy.c               |  1 -
 sound/drivers/mtpav.c               |  1 -
 sound/drivers/mts64.c               |  1 -
 sound/drivers/pcsp/pcsp.c           |  1 -
 sound/drivers/portman2x4.c          |  1 -
 sound/drivers/serial-u16550.c       |  1 -
 sound/drivers/virmidi.c             |  1 -
 sound/isa/ad1816a/ad1816a.c         |  7 -------
 sound/isa/ad1848/ad1848.c           |  3 ---
 sound/isa/als100.c                  | 11 -----------
 sound/isa/azt2320.c                 |  5 -----
 sound/isa/cmi8330.c                 |  1 -
 sound/isa/cs423x/cs4231.c           |  1 -
 sound/isa/cs423x/cs4236.c           | 34 ----------------------------------
 sound/isa/es1688/es1688.c           |  5 -----
 sound/isa/es18xx.c                  | 10 +---------
 sound/isa/gus/gusclassic.c          |  1 -
 sound/isa/gus/gusextreme.c          |  1 -
 sound/isa/gus/gusmax.c              |  1 -
 sound/isa/gus/interwave.c           |  6 ------
 sound/isa/opl3sa2.c                 |  5 -----
 sound/isa/opti9xx/miro.c            |  3 ---
 sound/isa/opti9xx/opti92x-ad1848.c  |  6 ------
 sound/isa/sb/jazz16.c               |  3 ---
 sound/isa/sb/sb16.c                 |  8 --------
 sound/isa/sb/sb8.c                  |  1 -
 sound/isa/sc6000.c                  |  3 ---
 sound/isa/wavefront/wavefront.c     |  1 -
 sound/mips/sgio2audio.c             |  1 -
 sound/pci/ad1889.c                  |  1 -
 sound/pci/ali5451/ali5451.c         |  1 -
 sound/pci/als300.c                  |  1 -
 sound/pci/als4000.c                 |  1 -
 sound/pci/atiixp.c                  |  1 -
 sound/pci/atiixp_modem.c            |  1 -
 sound/pci/au88x0/au88x0.c           |  2 --
 sound/pci/azt3328.c                 |  1 -
 sound/pci/bt87x.c                   |  2 --
 sound/pci/ca0106/ca0106_main.c      |  1 -
 sound/pci/cmipci.c                  |  4 ----
 sound/pci/cs4281.c                  |  1 -
 sound/pci/cs46xx/cs46xx.c           |  7 -------
 sound/pci/cs5535audio/cs5535audio.c |  1 -
 sound/pci/ctxfi/xfi.c               |  1 -
 sound/pci/echoaudio/echoaudio.c     |  1 -
 sound/pci/emu10k1/emu10k1.c         |  2 --
 sound/pci/emu10k1/emu10k1x.c        |  1 -
 sound/pci/ens1370.c                 |  8 --------
 sound/pci/es1938.c                  |  4 ----
 sound/pci/es1968.c                  |  4 ----
 sound/pci/fm801.c                   |  2 --
 sound/pci/hda/hda_intel.c           | 34 ----------------------------------
 sound/pci/ice1712/ice1712.c         |  6 ------
 sound/pci/ice1712/ice1724.c         | 19 -------------------
 sound/pci/intel8x0.c                | 23 -----------------------
 sound/pci/intel8x0m.c               | 15 ---------------
 sound/pci/korg1212/korg1212.c       |  1 -
 sound/pci/lola/lola.c               |  1 -
 sound/pci/lx6464es/lx6464es.c       |  2 --
 sound/pci/maestro3.c                |  5 -----
 sound/pci/mixart/mixart.c           |  1 -
 sound/pci/nm256/nm256.c             |  2 --
 sound/pci/oxygen/oxygen.c           |  3 ---
 sound/pci/oxygen/se6x.c             |  1 -
 sound/pci/oxygen/virtuoso.c         |  1 -
 sound/pci/pcxhr/pcxhr.c             |  1 -
 sound/pci/riptide/riptide.c         |  1 -
 sound/pci/rme32.c                   |  1 -
 sound/pci/rme96.c                   |  5 -----
 sound/pci/rme9652/hdsp.c            |  3 ---
 sound/pci/rme9652/hdspm.c           |  1 -
 sound/pci/rme9652/rme9652.c         |  2 --
 sound/pci/sis7019.c                 |  1 -
 sound/pci/sonicvibes.c              |  1 -
 sound/pci/trident/trident.c         | 12 ------------
 sound/pci/via82xx.c                 |  1 -
 sound/pci/via82xx_modem.c           |  1 -
 sound/pci/vx222/vx222.c             |  1 -
 sound/pci/ymfpci/ymfpci.c           |  6 ------
 sound/pcmcia/pdaudiocf/pdaudiocf.c  |  1 -
 sound/pcmcia/vx/vxpocket.c          |  4 ----
 sound/ppc/powermac.c                |  1 -
 sound/sh/aica.c                     |  1 -
 sound/sh/sh_dac_audio.c             |  1 -
 sound/sparc/amd7930.c               |  1 -
 sound/sparc/cs4231.c                |  1 -
 sound/sparc/dbri.c                  |  1 -
 sound/usb/6fire/chip.c              |  1 -
 sound/usb/caiaq/device.c            | 14 --------------
 sound/usb/card.c                    |  2 --
 sound/usb/hiface/chip.c             | 17 -----------------
 sound/usb/misc/ua101.c              |  1 -
 sound/usb/usx2y/usbusx2y.c          |  1 -
 sound/x86/intel_hdmi_audio.c        |  1 -
 sound/xen/xen_snd_front.c           |  1 -
 96 files changed, 1 insertion(+), 372 deletions(-)

(limited to 'sound')

diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 8a24e5ae7cef..52637180af33 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -33,7 +33,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("A loopback soundcard");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ALSA,Loopback soundcard}}");
 
 #define MAX_PCM_SUBSTREAMS	8
 
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 316c9afadefe..01a3eab50d7b 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -25,7 +25,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Dummy soundcard (/dev/null)");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}");
 
 #define MAX_PCM_DEVICES		4
 #define MAX_PCM_SUBSTREAMS	128
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index ce5fd17bd720..df4b7f9cd50f 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -53,7 +53,6 @@
 MODULE_AUTHOR("Michael T. Mayers");
 MODULE_DESCRIPTION("MOTU MidiTimePiece AV multiport MIDI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{MOTU,MidiTimePiece AV multiport MIDI}}");
 
 // io resources
 #define MTPAV_IOBASE		0x378
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index 9c708b693cb3..322d530ab07b 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -37,7 +37,6 @@ MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
 MODULE_AUTHOR("Matthias Koenig <mk@phasorlab.de>");
 MODULE_DESCRIPTION("ESI Miditerminal 4140");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESI,Miditerminal 4140}}");
 
 /*********************************************************************
  * Chip specific
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index fd79e57c85ca..7689fa2f9531 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -22,7 +22,6 @@
 MODULE_AUTHOR("Stas Sergeev <stsp@users.sourceforge.net>");
 MODULE_DESCRIPTION("PC-Speaker driver");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{PC-Speaker, pcsp}}");
 MODULE_ALIAS("platform:pcspkr");
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index c876cf9b5005..2f4514ed47c5 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -57,7 +57,6 @@ MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
 MODULE_AUTHOR("Levent Guendogdu, Tobias Gehrig, Matthias Koenig");
 MODULE_DESCRIPTION("Midiman Portman2x4");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Midiman,Portman2x4}}");
 
 /*********************************************************************
  * Chip specific
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index 3947f084dd6b..6d5d1ca59ecf 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -34,7 +34,6 @@
 
 MODULE_DESCRIPTION("MIDI serial u16550");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ALSA, MIDI serial u16550}}");
 
 #define SNDRV_SERIAL_SOUNDCANVAS 0 /* Roland Soundcanvas; F5 NN selects part */
 #define SNDRV_SERIAL_MS124T 1      /* Midiator MS-124T */
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
index f1fb68b15498..4206d93ab47e 100644
--- a/sound/drivers/virmidi.c
+++ b/sound/drivers/virmidi.c
@@ -43,7 +43,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("Dummy soundcard for virtual rawmidi devices");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}");
 
 #define MAX_MIDI_DEVICES	4
 
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
index ca18fe3ff8a5..f11af983b3b6 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -22,13 +22,6 @@
 MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
 MODULE_DESCRIPTION("AD1816A, AD1815");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Highscreen,Sound-Boostar 16 3D},"
-		"{Analog Devices,AD1815},"
-		"{Analog Devices,AD1816A},"
-		"{TerraTec,Base 64},"
-		"{TerraTec,AudioSystem EWS64S},"
-		"{Aztech/Newcom SC-16 3D},"
-		"{Shark Predator ISA}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 1-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
index 6f221eed44e2..edafb49797e7 100644
--- a/sound/isa/ad1848/ad1848.c
+++ b/sound/isa/ad1848/ad1848.c
@@ -22,9 +22,6 @@
 MODULE_DESCRIPTION(CRD_NAME);
 MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848},"
-	        "{Analog Devices,AD1847},"
-		"{Crystal Semiconductors,CS4248}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 1085f5b01318..bacb7a1b930c 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -26,17 +26,6 @@
 #define PFX "als100: "
 
 MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0");
-MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X},"
-		"{Avance Logic ALS-007}}"
-		"{{Avance Logic,ALS100 - PRO16PNP},"
-	        "{Avance Logic,ALS110},"
-	        "{Avance Logic,ALS120},"
-	        "{Avance Logic,ALS200},"
-	        "{3D Melody,MF1000},"
-	        "{Digimate,3D Sound},"
-	        "{Avance Logic,ALS120},"
-	        "{RTL,RTL3000}}");
-
 MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
 MODULE_LICENSE("GPL");
 
diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c
index 4ed52094fc8d..867e9ae8f65a 100644
--- a/sound/isa/azt2320.c
+++ b/sound/isa/azt2320.c
@@ -35,11 +35,6 @@
 MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
 MODULE_DESCRIPTION("Aztech Systems AZT2320");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Aztech Systems,PRO16V},"
-		"{Aztech Systems,AZT2320},"
-		"{Aztech Systems,AZT3300},"
-		"{Aztech Systems,AZT2320},"
-		"{Aztech Systems,AZT3000}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index 19e258527d69..bc112df10fc5 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -51,7 +51,6 @@
 MODULE_AUTHOR("George Talusan <gstalusan@uwaterloo.ca>");
 MODULE_DESCRIPTION("C-Media CMI8330/CMI8329");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index c56cbc072918..ec054b929214 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -23,7 +23,6 @@
 MODULE_DESCRIPTION(CRD_NAME);
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 63fb0cb754d0..186d7d4db45e 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -18,40 +18,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Cirrus Logic CS4232-9");
-MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000},"
-		"{Turtle Beach,Tropez Plus},"
-		"{SIC CrystalWave 32},"
-		"{Hewlett Packard,Omnibook 5500},"
-		"{TerraTec,Maestro 32/96},"
-		"{Philips,PCA70PS}},"
-		"{{Crystal Semiconductors,CS4235},"
-		"{Crystal Semiconductors,CS4236},"
-		"{Crystal Semiconductors,CS4237},"
-		"{Crystal Semiconductors,CS4238},"
-		"{Crystal Semiconductors,CS4239},"
-		"{Acer,AW37},"
-		"{Acer,AW35/Pro},"
-		"{Crystal,3D},"
-		"{Crystal Computer,TidalWave128},"
-		"{Dell,Optiplex GX1},"
-		"{Dell,Workstation 400 sound},"
-		"{EliteGroup,P5TX-LA sound},"
-		"{Gallant,SC-70P},"
-		"{Gateway,E1000 Onboard CS4236B},"
-		"{Genius,Sound Maker 3DJ},"
-		"{Hewlett Packard,HP6330 sound},"
-		"{IBM,PC 300PL sound},"
-		"{IBM,Aptiva 2137 E24},"
-		"{IBM,IntelliStation M Pro},"
-		"{Intel,Marlin Spike Mobo CS4235},"
-		"{Intel PR440FX Onboard},"
-		"{Guillemot,MaxiSound 16 PnP},"
-		"{NewClear,3D},"
-		"{TerraTec,AudioSystem EWS64L/XL},"
-		"{Typhoon Soundsystem,CS4236B},"
-		"{Turtle Beach,Malibu},"
-		"{Unknown,Digital PC 5000 Onboard}}");
-
 MODULE_ALIAS("snd_cs4232");
 
 #define IDENT "CS4232+"
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index 4a1f61f1a331..750d4995634f 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -26,11 +26,6 @@
 MODULE_DESCRIPTION(CRD_NAME);
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100},"
-	        "{ESS,ES1688 PnP AudioDrive,pnp:ESS0102},"
-	        "{ESS,ES688 AudioDrive,pnp:ESS6881},"
-	        "{ESS,ES1688 AudioDrive,pnp:ESS1681}}");
-
 MODULE_ALIAS("snd_es968");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index 9beef8079177..375a4a6a4717 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1929,17 +1929,9 @@ static int snd_es18xx_mixer(struct snd_card *card)
 
 /* Card level */
 
-MODULE_AUTHOR("Christian Fischbach <fishbach@pool.informatik.rwth-aachen.de>, Abramo Bagnara <abramo@alsa-project.org>");  
+MODULE_AUTHOR("Christian Fischbach <fishbach@pool.informatik.rwth-aachen.de>, Abramo Bagnara <abramo@alsa-project.org>");
 MODULE_DESCRIPTION("ESS ES18xx AudioDrive");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESS,ES1868 PnP AudioDrive},"
-		"{ESS,ES1869 PnP AudioDrive},"
-		"{ESS,ES1878 PnP AudioDrive},"
-		"{ESS,ES1879 PnP AudioDrive},"
-		"{ESS,ES1887 PnP AudioDrive},"
-		"{ESS,ES1888 PnP AudioDrive},"
-		"{ESS,ES1887 AudioDrive},"
-		"{ESS,ES1888 AudioDrive}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
index 015f88a11352..0fba5d8fe84f 100644
--- a/sound/isa/gus/gusclassic.c
+++ b/sound/isa/gus/gusclassic.c
@@ -23,7 +23,6 @@
 MODULE_DESCRIPTION(CRD_NAME);
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index c9f31b4fb887..da2b2ca6b721 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -27,7 +27,6 @@
 MODULE_DESCRIPTION(CRD_NAME);
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index dc09fbd6f88d..24b945f1768d 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -21,7 +21,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Gravis UltraSound MAX");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index e4d412e72b75..99581fba4ca8 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -28,14 +28,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
 #ifndef SNDRV_STB
 MODULE_DESCRIPTION("AMD InterWave");
-MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Plug & Play},"
-		"{STB,SoundRage32},"
-		"{MED,MED3210},"
-		"{Dynasonix,Dynasonix Pro},"
-		"{Panasonic,PCA761AW}}");
 #else
 MODULE_DESCRIPTION("AMD InterWave STB with TEA6330T");
-MODULE_SUPPORTED_DEVICE("{{AMD,InterWave STB with TEA6330T}}");
 #endif
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 7649a8a4128d..9bde11d1cfe8 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -22,11 +22,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Yamaha OPL3SA2+");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S},"
-		"{Genius,Sound Maker 3DX},"
-		"{Yamaha,OPL3SA3},"
-		"{Intel,AL440LX sound},"
-	        "{NeoMagic,MagicWave 3DX}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 20933342f5eb..a510b201143c 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -33,9 +33,6 @@
 MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Miro miroSOUND PCM1 pro, PCM12, PCM20 Radio");
-MODULE_SUPPORTED_DEVICE("{{Miro,miroSOUND PCM1 pro}, "
-			"{Miro,miroSOUND PCM12}, "
-			"{Miro,miroSOUND PCM20 Radio}}");
 
 static int index = SNDRV_DEFAULT_IDX1;		/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;		/* ID for this card */
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 758f5b579138..08e61d90057b 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -36,17 +36,11 @@ MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
 MODULE_LICENSE("GPL");
 #ifdef OPTi93X
 MODULE_DESCRIPTION("OPTi93X");
-MODULE_SUPPORTED_DEVICE("{{OPTi,82C931/3}}");
 #else	/* OPTi93X */
 #ifdef CS4231
 MODULE_DESCRIPTION("OPTi92X - CS4231");
-MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (CS4231)},"
-		"{OPTi,82C925 (CS4231)}}");
 #else	/* CS4231 */
 MODULE_DESCRIPTION("OPTi92X - AD1848");
-MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)},"
-		"{OPTi,82C925 (AD1848)},"
-	        "{OAK,Mozart}}");
 #endif	/* CS4231 */
 #endif	/* OPTi93X */
 
diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c
index 0e2e0ab3b9e4..7ba5dd1ec810 100644
--- a/sound/isa/sb/jazz16.c
+++ b/sound/isa/sb/jazz16.c
@@ -28,9 +28,6 @@
 #define PFX "jazz16: "
 
 MODULE_DESCRIPTION("Media Vision Jazz16");
-MODULE_SUPPORTED_DEVICE("{{Media Vision ??? },"
-		"{RTL,RTL3000}}");
-
 MODULE_AUTHOR("Krzysztof Helt <krzysztof.h1@wp.pl>");
 MODULE_LICENSE("GPL");
 
diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
index db284b7b88a7..63ef960abd25 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -31,16 +31,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
 #ifndef SNDRV_SBAWE
 MODULE_DESCRIPTION("Sound Blaster 16");
-MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 16},"
-		"{Creative Labs,SB Vibra16S},"
-		"{Creative Labs,SB Vibra16C},"
-		"{Creative Labs,SB Vibra16CL},"
-		"{Creative Labs,SB Vibra16X}}");
 #else
 MODULE_DESCRIPTION("Sound Blaster AWE");
-MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB AWE 32},"
-		"{Creative Labs,SB AWE 64},"
-		"{Creative Labs,SB AWE 64 Gold}}");
 #endif
 
 #if 0
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 8e3e67b9a341..6c9d534ce8b6 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -17,7 +17,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index def137579717..3462663050bb 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -29,9 +29,6 @@
 MODULE_AUTHOR("Krzysztof Helt");
 MODULE_DESCRIPTION("Gallant SC-6000");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000},"
-			"{AudioExcel, Audio Excel DSP 16},"
-			"{Zoltrix, AV302}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
index b750a4fd40de..a4437971df2f 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -21,7 +21,6 @@
 MODULE_AUTHOR("Paul Barton-Davis <pbd@op.net>");
 MODULE_DESCRIPTION("Turtle Beach Wavefront");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Turtle Beach,Maui/Tropez/Tropez+}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	    /* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	    /* ID for this card */
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 5bf1ea150f26..989f656e2de7 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -32,7 +32,6 @@
 MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
 MODULE_DESCRIPTION("SGI O2 Audio");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
 
 static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 5d835d2af054..4520022801d9 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -43,7 +43,6 @@
 MODULE_AUTHOR("Kyle McMartin <kyle@parisc-linux.org>, Thibaut Varene <t-bone@parisc-linux.org>");
 MODULE_DESCRIPTION("Analog Devices AD1889 ALSA sound driver");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1889}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 module_param_array(index, int, NULL, 0444);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 51f24796f03f..0d66b92466d5 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -29,7 +29,6 @@
 MODULE_AUTHOR("Matt Wu <Matt_Wu@acersoftech.com.cn>");
 MODULE_DESCRIPTION("ALI M5451");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}");
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 1dc8c4ed0592..bd4fd09e982b 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -86,7 +86,6 @@ enum {DEVICE_ALS300, DEVICE_ALS300_PLUS};
 MODULE_AUTHOR("Ash Willis <ashwillis@programmer.net>");
 MODULE_DESCRIPTION("Avance Logic ALS300");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS300},{Avance Logic,ALS300+}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 2edc7455285a..139ac2a3a0ef 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -68,7 +68,6 @@
 MODULE_AUTHOR("Bart Hartgers <bart@etpmod.phys.tue.nl>, Andreas Mohr");
 MODULE_DESCRIPTION("Avance Logic ALS4000");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index a25d75455802..579425ccbb6a 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -23,7 +23,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("ATI IXP AC97 controller");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250/300/400/600}}");
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index ae88217d685a..45e75afec7a0 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -23,7 +23,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("ATI IXP MC97 controller");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250}}");
 
 static int index = -2; /* Exclude the first card */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 5dd98e6ff34b..1b37b7225b1d 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -41,8 +41,6 @@ MODULE_PARM_DESC(pcifix, "Enable VIA-workaround for " CARD_NAME " soundcard.");
 
 MODULE_DESCRIPTION("Aureal vortex");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Aureal Semiconductor Inc., Aureal Vortex Sound Processor}}");
-
 MODULE_DEVICE_TABLE(pci, snd_vortex_ids);
 
 static void vortex_fix_latency(struct pci_dev *vortex)
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 2ac594dcf21c..51dcf1bc4c0c 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -196,7 +196,6 @@
 MODULE_AUTHOR("Andreas Mohr <andi AT lisas.de>");
 MODULE_DESCRIPTION("Aztech AZF3328 (PCI168)");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_GAMEPORT 1
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index cf9f8d80a0b6..91512b345d19 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -23,8 +23,6 @@
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("Brooktree Bt87x audio driver");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878},"
-		"{Brooktree,Bt879}}");
 
 static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index ee20f9a1aae9..bee4710916c4 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -137,7 +137,6 @@
 MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>");
 MODULE_DESCRIPTION("CA0106");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}");
 
 // module parameters (see "Module Parameters")
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 7363d61eaec2..598446348da6 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -30,10 +30,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("C-Media CMI8x38 PCI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738},"
-		"{C-Media,CMI8738B},"
-		"{C-Media,CMI8338A},"
-		"{C-Media,CMI8338B}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 94d2a6a466a8..bf3bb70ffaf9 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -25,7 +25,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Cirrus Logic CS4281");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index a6e0a4439332..1db7b4112840 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -21,13 +21,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)},"
-		"{Cirrus Logic,Sound Fusion (CS4610)},"
-		"{Cirrus Logic,Sound Fusion (CS4612)},"
-		"{Cirrus Logic,Sound Fusion (CS4615)},"
-		"{Cirrus Logic,Sound Fusion (CS4622)},"
-		"{Cirrus Logic,Sound Fusion (CS4624)},"
-		"{Cirrus Logic,Sound Fusion (CS4630)}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 359bc6af8670..9b716b56d739 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -393,4 +393,3 @@ module_pci_driver(cs5535audio_driver);
 MODULE_AUTHOR("Jaya Kumar");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("CS5535 Audio");
-MODULE_SUPPORTED_DEVICE("CS5535 Audio");
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 8c07c6463c24..713d36ea40cb 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -18,7 +18,6 @@
 MODULE_AUTHOR("Creative Technology Ltd");
 MODULE_DESCRIPTION("X-Fi driver version 1.03");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}");
 
 static unsigned int reference_rate = 48000;
 static unsigned int multiple = 2;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index a20b2bb5c898..9bd67ac33657 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -10,7 +10,6 @@
 MODULE_AUTHOR("Giuliano Pochini <pochini@shiny.it>");
 MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver");
-MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}");
 MODULE_DEVICE_TABLE(pci, snd_echo_ids);
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 353934c88cbd..45833bc2a7e7 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -18,8 +18,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("EMU10K1");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS},"
-	       "{Creative Labs,SB Audigy}}");
 
 #if IS_ENABLED(CONFIG_SND_SEQUENCER)
 #define ENABLE_SYNTH
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 785ec0cf3933..d9a12cd01647 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -31,7 +31,6 @@
 MODULE_AUTHOR("Francisco Moraes <fmoraes@nc.rr.com>");
 MODULE_DESCRIPTION("EMU10K1X");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Dell Creative Labs,SB Live!}");
 
 // module parameters (see "Module Parameters")
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 93c4fd313311..3ccccdbc0029 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -52,17 +52,9 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Thomas Sailer <sailer@ife.ee.et
 MODULE_LICENSE("GPL");
 #ifdef CHIP1370
 MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370");
-MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI-97 ES1370},"
-	        "{Creative Labs,SB PCI64/128 (ES1370)}}");
 #endif
 #ifdef CHIP1371
 MODULE_DESCRIPTION("Ensoniq/Creative AudioPCI ES1371+");
-MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI ES1371/73},"
-		"{Ensoniq,AudioPCI ES1373},"
-		"{Creative Labs,Ectiva EV1938},"
-		"{Creative Labs,SB PCI64/128 (ES1371/73)},"
-		"{Creative Labs,Vibra PCI128},"
-		"{Ectiva,EV1938}}");
 #endif
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 3b5d68ce9dd5..afc66347d162 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -52,10 +52,6 @@
 MODULE_AUTHOR("Jaromir Koutek <miri@punknet.cz>");
 MODULE_DESCRIPTION("ESS Solo-1");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESS,ES1938},"
-                "{ESS,ES1946},"
-                "{ESS,ES1969},"
-		"{TerraTec,128i PCI}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 747fa69bb1c9..5fa1861236f5 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -107,10 +107,6 @@
 
 MODULE_DESCRIPTION("ESS Maestro");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESS,Maestro 2e},"
-		"{ESS,Maestro 2},"
-		"{ESS,Maestro 1},"
-		"{TerraTec,DMX}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index c6ad6235a669..6279eb156e36 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -26,8 +26,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("ForteMedia FM801");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801},"
-		"{Genius,SoundMaker Live 5.1}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 5eea130dcf0a..b59b0f323d4e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -208,40 +208,6 @@ MODULE_PARM_DESC(snoop, "Enable/disable snooping");
 
 
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
-			 "{Intel, ICH6M},"
-			 "{Intel, ICH7},"
-			 "{Intel, ESB2},"
-			 "{Intel, ICH8},"
-			 "{Intel, ICH9},"
-			 "{Intel, ICH10},"
-			 "{Intel, PCH},"
-			 "{Intel, CPT},"
-			 "{Intel, PPT},"
-			 "{Intel, LPT},"
-			 "{Intel, LPT_LP},"
-			 "{Intel, WPT_LP},"
-			 "{Intel, SPT},"
-			 "{Intel, SPT_LP},"
-			 "{Intel, HPT},"
-			 "{Intel, PBG},"
-			 "{Intel, SCH},"
-			 "{ATI, SB450},"
-			 "{ATI, SB600},"
-			 "{ATI, RS600},"
-			 "{ATI, RS690},"
-			 "{ATI, RS780},"
-			 "{ATI, R600},"
-			 "{ATI, RV630},"
-			 "{ATI, RV610},"
-			 "{ATI, RV670},"
-			 "{ATI, RV635},"
-			 "{ATI, RV620},"
-			 "{ATI, RV770},"
-			 "{VIA, VT8251},"
-			 "{VIA, VT8237A},"
-			 "{SiS, SIS966},"
-			 "{ULI, M5461}}");
 MODULE_DESCRIPTION("Intel HDA driver");
 
 #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO)
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index f814dbbec2a4..d54cd5143e9f 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -60,12 +60,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("ICEnsemble ICE1712 (Envy24)");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{"
-	       HOONTECH_DEVICE_DESC
-	       DELTA_DEVICE_DESC
-	       EWS_DEVICE_DESC
-	       "{ICEnsemble,Generic ICE1712},"
-	       "{ICEnsemble,Generic Envy24}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index c0fca94c1dd2..ef2367d86148 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -44,25 +44,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{"
-	       REVO_DEVICE_DESC
-	       AMP_AUDIO2000_DEVICE_DESC
-	       AUREON_DEVICE_DESC
-	       VT1720_MOBO_DEVICE_DESC
-	       PONTIS_DEVICE_DESC
-	       PRODIGY192_DEVICE_DESC
-	       PRODIGY_HIFI_DEVICE_DESC
-	       JULI_DEVICE_DESC
-	       MAYA44_DEVICE_DESC
-	       PHASE_DEVICE_DESC
-	       WTM_DEVICE_DESC
-	       SE_DEVICE_DESC
-	       QTET_DEVICE_DESC
-		"{VIA,VT1720},"
-		"{VIA,VT1724},"
-		"{ICEnsemble,Generic ICE1724},"
-		"{ICEnsemble,Generic Envy24HT}"
-		"{ICEnsemble,Generic Envy24PT}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 3349e455a871..35903d1a1cbd 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -27,29 +27,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
-		"{Intel,82901AB-ICH0},"
-		"{Intel,82801BA-ICH2},"
-		"{Intel,82801CA-ICH3},"
-		"{Intel,82801DB-ICH4},"
-		"{Intel,ICH5},"
-		"{Intel,ICH6},"
-		"{Intel,ICH7},"
-		"{Intel,6300ESB},"
-		"{Intel,ESB2},"
-		"{Intel,MX440},"
-		"{SiS,SI7012},"
-		"{NVidia,nForce Audio},"
-		"{NVidia,nForce2 Audio},"
-		"{NVidia,nForce3 Audio},"
-		"{NVidia,MCP04},"
-		"{NVidia,MCP501},"
-		"{NVidia,CK804},"
-		"{NVidia,CK8},"
-		"{NVidia,CK8S},"
-		"{AMD,AMD768},"
-		"{AMD,AMD8111},"
-	        "{ALI,M5455}}");
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 19872cecc9d2..13ef838b26c1 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -25,21 +25,6 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; "
 		   "SiS 7013; NVidia MCP/2/2S/3 modems");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
-		"{Intel,82901AB-ICH0},"
-		"{Intel,82801BA-ICH2},"
-		"{Intel,82801CA-ICH3},"
-		"{Intel,82801DB-ICH4},"
-		"{Intel,ICH5},"
-		"{Intel,ICH6},"
-		"{Intel,ICH7},"
-	        "{Intel,MX440},"
-		"{SiS,7013},"
-		"{NVidia,NForce Modem},"
-		"{NVidia,NForce2 Modem},"
-		"{NVidia,NForce2s Modem},"
-		"{NVidia,NForce3 Modem},"
-		"{AMD,AMD768}}");
 
 static int index = -2; /* Exclude the first card */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 2eddd9de9e6d..80ac3c6152ad 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -388,7 +388,6 @@ struct snd_korg1212 {
 
 MODULE_DESCRIPTION("korg1212");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{KORG,korg1212}}");
 MODULE_FIRMWARE("korg/k1212.dsp");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;     /* Index 0-MAX */
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 491c90f83fbc..03b4be44bb26 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -54,7 +54,6 @@ MODULE_PARM_DESC(sample_rate_min, "Minimal sample rate");
  */
 
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Digigram, Lola}}");
 MODULE_DESCRIPTION("Digigram Lola driver");
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index b92ea074ff2a..1be97c38bc71 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -21,8 +21,6 @@
 MODULE_AUTHOR("Tim Blechmann");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("digigram lx6464es");
-MODULE_SUPPORTED_DEVICE("{digigram lx6464es{}}");
-
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index d2c2cd6006f0..cdc4b6106252 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -39,11 +39,6 @@
 MODULE_AUTHOR("Zach Brown <zab@zabbo.net>, Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("ESS Maestro3 PCI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{ESS,Maestro3 PCI},"
-		"{ESS,ES1988},"
-		"{ESS,Allegro PCI},"
-		"{ESS,Allegro-1 PCI},"
-	        "{ESS,Canyon3D-2/LE PCI}}");
 MODULE_FIRMWARE("ess/maestro3_assp_kernel.fw");
 MODULE_FIRMWARE("ess/maestro3_assp_minisrc.fw");
 
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index efff220b26ea..a0bbb386dc25 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -32,7 +32,6 @@
 MODULE_AUTHOR("Digigram <alsa@digigram.com>");
 MODULE_DESCRIPTION("Digigram " CARD_NAME);
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;             /* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;              /* ID for this card */
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 975994623c2c..6cb689aa28c2 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -32,8 +32,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("NeoMagic NM256AV/ZX");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{NeoMagic,NM256AV},"
-		"{NeoMagic,NM256ZX}}");
 
 /*
  * some compile conditions.
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index a751fcce7c8e..e335c4b5b381 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -56,9 +56,6 @@
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8786}"
-			",{C-Media,CMI8787}"
-			",{C-Media,CMI8788}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/pci/oxygen/se6x.c b/sound/pci/oxygen/se6x.c
index 78c35a0a5477..434f885f6f91 100644
--- a/sound/pci/oxygen/se6x.c
+++ b/sound/pci/oxygen/se6x.c
@@ -29,7 +29,6 @@
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("Studio Evolution SE6X driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{Studio Evolution,SE6X}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 98ab16329827..baa3244d4dab 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -16,7 +16,6 @@
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("Asus Virtuoso driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{Asus,AV66},{Asus,AV100},{Asus,AV200}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index c2e4831c3a13..751f9744b089 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -35,7 +35,6 @@ MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>, "
 	      "Marc Titinger <titinger@digigram.com>");
 MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING);
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index fcc2073c5025..56827db97239 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -103,7 +103,6 @@
 MODULE_AUTHOR("Peter Gruber <nokos@gmx.net>");
 MODULE_DESCRIPTION("riptide");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Conexant,Riptide}}");
 MODULE_FIRMWARE("riptide.hex");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 4eabece4dcba..54f3e39f97f5 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -88,7 +88,6 @@ MODULE_PARM_DESC(fullduplex, "Support full-duplex mode.");
 MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>, Pilo Chambert <pilo.c@wanadoo.fr>");
 MODULE_DESCRIPTION("RME Digi32, Digi32/8, Digi32 PRO");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}");
 
 /* Defines for RME Digi32 series */
 #define RME32_SPDIF_NCHANNELS 2
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 84eef6a3739f..66082e9f526d 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -31,11 +31,6 @@ MODULE_AUTHOR("Anders Torger <torger@ludd.luth.se>");
 MODULE_DESCRIPTION("RME Digi96, Digi96/8, Digi96/8 PRO, Digi96/8 PST, "
 		   "Digi96/8 PAD");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{RME,Digi96},"
-		"{RME,Digi96/8},"
-		"{RME,Digi96/8 PRO},"
-		"{RME,Digi96/8 PST},"
-		"{RME,Digi96/8 PAD}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 6d9029333a12..4cf879c42dc4 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -44,9 +44,6 @@ MODULE_PARM_DESC(enable, "Enable/disable specific Hammerfall DSP soundcards.");
 MODULE_AUTHOR("Paul Davis <paul@linuxaudiosystems.com>, Marcus Andersson, Thomas Charbonnel <thomas@undata.org>");
 MODULE_DESCRIPTION("RME Hammerfall DSP");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP},"
-	        "{RME HDSP-9652},"
-		"{RME HDSP-9632}}");
 MODULE_FIRMWARE("rpm_firmware.bin");
 MODULE_FIRMWARE("multiface_firmware.bin");
 MODULE_FIRMWARE("multiface_firmware_rev11.bin");
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b66711574b1a..8d900c132f0f 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -165,7 +165,6 @@ MODULE_AUTHOR
 );
 MODULE_DESCRIPTION("RME HDSPM");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 /* --- Write registers. ---
   These are defined as byte-offsets from the iobase value.  */
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 012fbec5e6a7..4df992e846f2 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -39,8 +39,6 @@ MODULE_PARM_DESC(precise_ptr, "Enable precise pointer (doesn't work reliably).")
 MODULE_AUTHOR("Paul Davis <pbd@op.net>, Winfried Ritsch");
 MODULE_DESCRIPTION("RME Digi9652/Digi9636");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall},"
-		"{RME,Hammerfall-Light}}");
 
 /* The Hammerfall has two sets of 24 ADAT + 2 S/PDIF channels, one for
    capture, one for playback. Both the ADAT and S/PDIF channels appear
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 8ffa2f53c0b5..00ab51c889da 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -24,7 +24,6 @@
 MODULE_AUTHOR("David Dillow <dave@thedillows.org>");
 MODULE_DESCRIPTION("SiS7019");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 26fd1d08c179..7de10997775f 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -29,7 +29,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("S3 SonicVibes PCI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 5bc79da6e35e..a51041205f7c 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -17,18 +17,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>");
 MODULE_DESCRIPTION("Trident 4D-WaveDX/NX & SiS SI7018");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Trident,4DWave DX},"
-		"{Trident,4DWave NX},"
-		"{SiS,SI7018 PCI Audio},"
-		"{Best Union,Miss Melody 4DWave PCI},"
-		"{HIS,4DWave PCI},"
-		"{Warpspeed,ONSpeed 4DWave PCI},"
-		"{Aztech Systems,PCI 64-Q3D},"
-		"{Addonics,SV 750},"
-		"{CHIC,True Sound 4Dwave},"
-		"{Shark,Predator4D-PCI},"
-		"{Jaton,SonicWave 4D},"
-		"{Hoontech,SoundTrack Digital 4DWave NX}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 154d88ce8813..fd1f2f9cfbc3 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -56,7 +56,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("VIA VT82xx audio");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C,pci},{VIA,VT8233A/C,8235}}");
 
 #if IS_REACHABLE(CONFIG_GAMEPORT)
 #define SUPPORT_JOYSTICK 1
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index addfa196df21..30253306f67c 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -38,7 +38,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("VIA VT82xx modem");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C modem,pci}}");
 
 static int index = -2; /* Exclude the first card */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index f7800ed1b67e..2a9e1a77a81a 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -20,7 +20,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("Digigram VX222 V2/Mic");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 9b0d18a7bf35..99be1490ef0e 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -17,12 +17,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Yamaha DS-1 PCI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724},"
-		"{Yamaha,YMF724F},"
-		"{Yamaha,YMF740},"
-		"{Yamaha,YMF740C},"
-		"{Yamaha,YMF744},"
-		"{Yamaha,YMF754}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 27d9da6d61e8..144582350a05 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -22,7 +22,6 @@
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Sound Core " CARD_NAME);
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Sound Core," CARD_NAME "}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index afd30a90c807..636320489805 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -17,13 +17,9 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 
-/*
- */
-
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("Digigram VXPocket");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Digigram,VXPocket},{Digigram,VXPocket440}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index 96ef55082bf9..9fb51ebafde1 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -18,7 +18,6 @@
 #define CHIP_NAME "PMac"
 
 MODULE_DESCRIPTION("PowerMac");
-MODULE_SUPPORTED_DEVICE("{{Apple,PowerMac}}");
 MODULE_LICENSE("GPL");
 
 static int index = SNDRV_DEFAULT_IDX1;		/* Index 0-MAX */
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 8fa68432d3c1..6e9d6bd67369 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -32,7 +32,6 @@
 MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
 MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}");
 MODULE_FIRMWARE("aica_firmware.bin");
 
 /* module parameters */
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index feb28502940f..8ebd972846ac 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -25,7 +25,6 @@
 MODULE_AUTHOR("Rafael Ignacio Zurita <rizurita@yahoo.com>");
 MODULE_DESCRIPTION("SuperH DAC audio driver");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}");
 
 /* Module Parameters */
 static int index = SNDRV_DEFAULT_IDX1;
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index 9d0da5fa1c70..d24ae00878f5 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -62,7 +62,6 @@ MODULE_PARM_DESC(enable, "Enable Sun AMD7930 soundcard.");
 MODULE_AUTHOR("Thomas K. Dyas and David S. Miller");
 MODULE_DESCRIPTION("Sun AMD7930");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Sun,AMD7930}}");
 
 /* Device register layout.  */
 
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index 0eed5f79a2bf..35c17803a430 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -52,7 +52,6 @@ MODULE_PARM_DESC(enable, "Enable Sun CS4231 soundcard.");
 MODULE_AUTHOR("Jaroslav Kysela, Derrick J. Brashear and David S. Miller");
 MODULE_DESCRIPTION("Sun CS4231");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Sun,CS4231}}");
 
 #ifdef SBUS_SUPPORT
 struct sbus_dma_info {
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index 5a6fb66dd118..b055f5839578 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -76,7 +76,6 @@
 MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets");
 MODULE_DESCRIPTION("Sun DBRI");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Sun,DBRI}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c
index 08c6e6a52eb9..33e962178c93 100644
--- a/sound/usb/6fire/chip.c
+++ b/sound/usb/6fire/chip.c
@@ -26,7 +26,6 @@
 MODULE_AUTHOR("Torsten Schenk <torsten.schenk@zoho.com>");
 MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{TerraTec,DMX 6Fire USB}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index e03481caf7f6..49f63f878e6f 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -26,20 +26,6 @@
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
 MODULE_DESCRIPTION("caiaq USB audio");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Native Instruments,RigKontrol2},"
-			 "{Native Instruments,RigKontrol3},"
-			 "{Native Instruments,Kore Controller},"
-			 "{Native Instruments,Kore Controller 2},"
-			 "{Native Instruments,Audio Kontrol 1},"
-			 "{Native Instruments,Audio 2 DJ},"
-			 "{Native Instruments,Audio 4 DJ},"
-			 "{Native Instruments,Audio 8 DJ},"
-			 "{Native Instruments,Traktor Audio 2},"
-			 "{Native Instruments,Session I/O},"
-			 "{Native Instruments,GuitarRig mobile},"
-			 "{Native Instruments,Traktor Kontrol X1},"
-			 "{Native Instruments,Traktor Kontrol S4},"
-			 "{Native Instruments,Maschine Controller}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
diff --git a/sound/usb/card.c b/sound/usb/card.c
index b6f4c0848e66..0826a437f8fc 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -58,8 +58,6 @@
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("USB Audio");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}");
-
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c
index c2824188d142..95385e90882c 100644
--- a/sound/usb/hiface/chip.c
+++ b/sound/usb/hiface/chip.c
@@ -21,23 +21,6 @@ MODULE_AUTHOR("Michael Trimarchi <michael@amarulasolutions.com>");
 MODULE_AUTHOR("Antonio Ospite <ao2@amarulasolutions.com>");
 MODULE_DESCRIPTION("M2Tech hiFace USB-SPDIF audio driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{M2Tech,Young},"
-			 "{M2Tech,hiFace},"
-			 "{M2Tech,North Star},"
-			 "{M2Tech,W4S Young},"
-			 "{M2Tech,Corrson},"
-			 "{M2Tech,AUDIA},"
-			 "{M2Tech,SL Audio},"
-			 "{M2Tech,Empirical},"
-			 "{M2Tech,Rockna},"
-			 "{M2Tech,Pathos},"
-			 "{M2Tech,Metronome},"
-			 "{M2Tech,CAD},"
-			 "{M2Tech,Audio Esclusive},"
-			 "{M2Tech,Rotel},"
-			 "{M2Tech,Eeaudio},"
-			 "{The Chord Company,CHORD},"
-			 "{AVA Group A/S,Vitus}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 6b30155964ec..5834d1dc317e 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -19,7 +19,6 @@
 MODULE_DESCRIPTION("Edirol UA-101/1000 driver");
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101},{Edirol,UA-1000}}");
 
 /*
  * Should not be lower than the minimum scheduling delay of the host
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index c54158146917..3cd28d24f0a7 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -137,7 +137,6 @@
 MODULE_AUTHOR("Karsten Wiese <annabellesgarden@yahoo.de>");
 MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2");
 MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604),"NAME_ALLCAPS"(0x8001)(0x8005)(0x8007)}}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index 1d66c3a4fb10..33b12aa67cf5 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -1887,4 +1887,3 @@ MODULE_AUTHOR("Vaibhav Agarwal <vaibhav.agarwal@intel.com>");
 MODULE_AUTHOR("Jerome Anand <jerome.anand@intel.com>");
 MODULE_DESCRIPTION("Intel HDMI Audio driver");
 MODULE_LICENSE("GPL v2");
-MODULE_SUPPORTED_DEVICE("{Intel,Intel_HAD}");
diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c
index 228d82031297..2cb0a19be2b8 100644
--- a/sound/xen/xen_snd_front.c
+++ b/sound/xen/xen_snd_front.c
@@ -391,4 +391,3 @@ module_exit(xen_drv_fini);
 MODULE_DESCRIPTION("Xen virtual sound device frontend");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("xen:" XENSND_DRIVER_NAME);
-MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual soundcard}}");
-- 
cgit 


From a908a716696eee75bf85199cde2b0989290536d1 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Sun, 7 Mar 2021 16:18:00 +0100
Subject: ASoC/extcon: arizona: Move arizona jack code to
 sound/soc/codecs/arizona-jack.c

The jack handling for arizona codecs is being refactored so that it is
done directly by the codec drivers, instead of having an extcon-driver
bind to a separate "arizona-extcon" child-device for this.

drivers/mfd/arizona-core.c has already been updated to no longer
instantiate an "arizona-extcon" child-device for the arizona codecs.

This means that the "arizona-extcon" driver is no longer useful
(there are no longer any devices for it to bind to).

This commit drops the extcon Kconfig / Makefile bits and moves
drivers/extcon/extcon-arizona.c to sound/soc/codecs/arizona-jack.c .

This is a preparation patch for converting the arizona extcon-driver into
a helper library for letting the arizona codec-drivers directly report jack
state through the standard sound/soc/soc-jack.c functions.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
---
 sound/soc/codecs/arizona-jack.c | 1819 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 1819 insertions(+)
 create mode 100644 sound/soc/codecs/arizona-jack.c

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona-jack.c b/sound/soc/codecs/arizona-jack.c
new file mode 100644
index 000000000000..56d2ce05de50
--- /dev/null
+++ b/sound/soc/codecs/arizona-jack.c
@@ -0,0 +1,1819 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/*
+ * extcon-arizona.c - Extcon driver Wolfson Arizona devices
+ *
+ *  Copyright (C) 2012-2014 Wolfson Microelectronics plc
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/gpio/consumer.h>
+#include <linux/gpio.h>
+#include <linux/input.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/property.h>
+#include <linux/regulator/consumer.h>
+#include <linux/extcon-provider.h>
+
+#include <sound/soc.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/pdata.h>
+#include <linux/mfd/arizona/registers.h>
+#include <dt-bindings/mfd/arizona.h>
+
+#define ARIZONA_MAX_MICD_RANGE 8
+
+#define ARIZONA_MICD_CLAMP_MODE_JDL      0x4
+#define ARIZONA_MICD_CLAMP_MODE_JDH      0x5
+#define ARIZONA_MICD_CLAMP_MODE_JDL_GP5H 0x9
+#define ARIZONA_MICD_CLAMP_MODE_JDH_GP5H 0xb
+
+#define ARIZONA_TST_CAP_DEFAULT 0x3
+#define ARIZONA_TST_CAP_CLAMP   0x1
+
+#define ARIZONA_HPDET_MAX 10000
+
+#define HPDET_DEBOUNCE 500
+#define DEFAULT_MICD_TIMEOUT 2000
+
+#define ARIZONA_HPDET_WAIT_COUNT 15
+#define ARIZONA_HPDET_WAIT_DELAY_MS 20
+
+#define QUICK_HEADPHONE_MAX_OHM 3
+#define MICROPHONE_MIN_OHM      1257
+#define MICROPHONE_MAX_OHM      30000
+
+#define MICD_DBTIME_TWO_READINGS 2
+#define MICD_DBTIME_FOUR_READINGS 4
+
+#define MICD_LVL_1_TO_7 (ARIZONA_MICD_LVL_1 | ARIZONA_MICD_LVL_2 | \
+			 ARIZONA_MICD_LVL_3 | ARIZONA_MICD_LVL_4 | \
+			 ARIZONA_MICD_LVL_5 | ARIZONA_MICD_LVL_6 | \
+			 ARIZONA_MICD_LVL_7)
+
+#define MICD_LVL_0_TO_7 (ARIZONA_MICD_LVL_0 | MICD_LVL_1_TO_7)
+
+#define MICD_LVL_0_TO_8 (MICD_LVL_0_TO_7 | ARIZONA_MICD_LVL_8)
+
+struct arizona_extcon_info {
+	struct device *dev;
+	struct arizona *arizona;
+	struct mutex lock;
+	struct regulator *micvdd;
+	struct input_dev *input;
+
+	u16 last_jackdet;
+
+	int micd_mode;
+	const struct arizona_micd_config *micd_modes;
+	int micd_num_modes;
+
+	const struct arizona_micd_range *micd_ranges;
+	int num_micd_ranges;
+
+	bool micd_reva;
+	bool micd_clamp;
+
+	struct delayed_work hpdet_work;
+	struct delayed_work micd_detect_work;
+	struct delayed_work micd_timeout_work;
+
+	bool hpdet_active;
+	bool hpdet_done;
+	bool hpdet_retried;
+
+	int num_hpdet_res;
+	unsigned int hpdet_res[3];
+
+	bool mic;
+	bool detecting;
+	int jack_flips;
+
+	int hpdet_ip_version;
+
+	struct extcon_dev *edev;
+
+	struct gpio_desc *micd_pol_gpio;
+};
+
+static const struct arizona_micd_config micd_default_modes[] = {
+	{ ARIZONA_ACCDET_SRC, 1, 0 },
+	{ 0,                  2, 1 },
+};
+
+static const struct arizona_micd_range micd_default_ranges[] = {
+	{ .max =  11, .key = BTN_0 },
+	{ .max =  28, .key = BTN_1 },
+	{ .max =  54, .key = BTN_2 },
+	{ .max = 100, .key = BTN_3 },
+	{ .max = 186, .key = BTN_4 },
+	{ .max = 430, .key = BTN_5 },
+};
+
+/* The number of levels in arizona_micd_levels valid for button thresholds */
+#define ARIZONA_NUM_MICD_BUTTON_LEVELS 64
+
+static const int arizona_micd_levels[] = {
+	3, 6, 8, 11, 13, 16, 18, 21, 23, 26, 28, 31, 34, 36, 39, 41, 44, 46,
+	49, 52, 54, 57, 60, 62, 65, 67, 70, 73, 75, 78, 81, 83, 89, 94, 100,
+	105, 111, 116, 122, 127, 139, 150, 161, 173, 186, 196, 209, 220, 245,
+	270, 295, 321, 348, 375, 402, 430, 489, 550, 614, 681, 752, 903, 1071,
+	1257, 30000,
+};
+
+static const unsigned int arizona_cable[] = {
+	EXTCON_MECHANICAL,
+	EXTCON_JACK_MICROPHONE,
+	EXTCON_JACK_HEADPHONE,
+	EXTCON_JACK_LINE_OUT,
+	EXTCON_NONE,
+};
+
+static void arizona_start_hpdet_acc_id(struct arizona_extcon_info *info);
+
+static void arizona_extcon_hp_clamp(struct arizona_extcon_info *info,
+				    bool clamp)
+{
+	struct arizona *arizona = info->arizona;
+	unsigned int mask = 0, val = 0;
+	unsigned int cap_sel = 0;
+	int ret;
+
+	switch (arizona->type) {
+	case WM8998:
+	case WM1814:
+		mask = 0;
+		break;
+	case WM5110:
+	case WM8280:
+		mask = ARIZONA_HP1L_SHRTO | ARIZONA_HP1L_FLWR |
+		       ARIZONA_HP1L_SHRTI;
+		if (clamp) {
+			val = ARIZONA_HP1L_SHRTO;
+			cap_sel = ARIZONA_TST_CAP_CLAMP;
+		} else {
+			val = ARIZONA_HP1L_FLWR | ARIZONA_HP1L_SHRTI;
+			cap_sel = ARIZONA_TST_CAP_DEFAULT;
+		}
+
+		ret = regmap_update_bits(arizona->regmap,
+					 ARIZONA_HP_TEST_CTRL_1,
+					 ARIZONA_HP1_TST_CAP_SEL_MASK,
+					 cap_sel);
+		if (ret != 0)
+			dev_warn(arizona->dev,
+				 "Failed to set TST_CAP_SEL: %d\n", ret);
+		break;
+	default:
+		mask = ARIZONA_RMV_SHRT_HP1L;
+		if (clamp)
+			val = ARIZONA_RMV_SHRT_HP1L;
+		break;
+	}
+
+	snd_soc_dapm_mutex_lock(arizona->dapm);
+
+	arizona->hpdet_clamp = clamp;
+
+	/* Keep the HP output stages disabled while doing the clamp */
+	if (clamp) {
+		ret = regmap_update_bits(arizona->regmap,
+					 ARIZONA_OUTPUT_ENABLES_1,
+					 ARIZONA_OUT1L_ENA |
+					 ARIZONA_OUT1R_ENA, 0);
+		if (ret != 0)
+			dev_warn(arizona->dev,
+				"Failed to disable headphone outputs: %d\n",
+				 ret);
+	}
+
+	if (mask) {
+		ret = regmap_update_bits(arizona->regmap, ARIZONA_HP_CTRL_1L,
+					 mask, val);
+		if (ret != 0)
+			dev_warn(arizona->dev, "Failed to do clamp: %d\n",
+				 ret);
+
+		ret = regmap_update_bits(arizona->regmap, ARIZONA_HP_CTRL_1R,
+					 mask, val);
+		if (ret != 0)
+			dev_warn(arizona->dev, "Failed to do clamp: %d\n",
+				 ret);
+	}
+
+	/* Restore the desired state while not doing the clamp */
+	if (!clamp) {
+		ret = regmap_update_bits(arizona->regmap,
+					 ARIZONA_OUTPUT_ENABLES_1,
+					 ARIZONA_OUT1L_ENA |
+					 ARIZONA_OUT1R_ENA, arizona->hp_ena);
+		if (ret != 0)
+			dev_warn(arizona->dev,
+				 "Failed to restore headphone outputs: %d\n",
+				 ret);
+	}
+
+	snd_soc_dapm_mutex_unlock(arizona->dapm);
+}
+
+static void arizona_extcon_set_mode(struct arizona_extcon_info *info, int mode)
+{
+	struct arizona *arizona = info->arizona;
+
+	mode %= info->micd_num_modes;
+
+	gpiod_set_value_cansleep(info->micd_pol_gpio,
+				 info->micd_modes[mode].gpio);
+
+	regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+			   ARIZONA_MICD_BIAS_SRC_MASK,
+			   info->micd_modes[mode].bias <<
+			   ARIZONA_MICD_BIAS_SRC_SHIFT);
+	regmap_update_bits(arizona->regmap, ARIZONA_ACCESSORY_DETECT_MODE_1,
+			   ARIZONA_ACCDET_SRC, info->micd_modes[mode].src);
+
+	info->micd_mode = mode;
+
+	dev_dbg(arizona->dev, "Set jack polarity to %d\n", mode);
+}
+
+static const char *arizona_extcon_get_micbias(struct arizona_extcon_info *info)
+{
+	switch (info->micd_modes[0].bias) {
+	case 1:
+		return "MICBIAS1";
+	case 2:
+		return "MICBIAS2";
+	case 3:
+		return "MICBIAS3";
+	default:
+		return "MICVDD";
+	}
+}
+
+static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	const char *widget = arizona_extcon_get_micbias(info);
+	struct snd_soc_dapm_context *dapm = arizona->dapm;
+	struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
+	int ret;
+
+	ret = snd_soc_component_force_enable_pin(component, widget);
+	if (ret != 0)
+		dev_warn(arizona->dev, "Failed to enable %s: %d\n",
+			 widget, ret);
+
+	snd_soc_dapm_sync(dapm);
+
+	if (!arizona->pdata.micd_force_micbias) {
+		ret = snd_soc_component_disable_pin(component, widget);
+		if (ret != 0)
+			dev_warn(arizona->dev, "Failed to disable %s: %d\n",
+				 widget, ret);
+
+		snd_soc_dapm_sync(dapm);
+	}
+}
+
+static void arizona_start_mic(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	bool change;
+	int ret;
+	unsigned int mode;
+
+	/* Microphone detection can't use idle mode */
+	pm_runtime_get_sync(info->dev);
+
+	if (info->detecting) {
+		ret = regulator_allow_bypass(info->micvdd, false);
+		if (ret != 0) {
+			dev_err(arizona->dev,
+				"Failed to regulate MICVDD: %d\n",
+				ret);
+		}
+	}
+
+	ret = regulator_enable(info->micvdd);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to enable MICVDD: %d\n",
+			ret);
+	}
+
+	if (info->micd_reva) {
+		const struct reg_sequence reva[] = {
+			{ 0x80,  0x3 },
+			{ 0x294, 0x0 },
+			{ 0x80,  0x0 },
+		};
+
+		regmap_multi_reg_write(arizona->regmap, reva, ARRAY_SIZE(reva));
+	}
+
+	if (info->detecting && arizona->pdata.micd_software_compare)
+		mode = ARIZONA_ACCDET_MODE_ADC;
+	else
+		mode = ARIZONA_ACCDET_MODE_MIC;
+
+	regmap_update_bits(arizona->regmap,
+			   ARIZONA_ACCESSORY_DETECT_MODE_1,
+			   ARIZONA_ACCDET_MODE_MASK, mode);
+
+	arizona_extcon_pulse_micbias(info);
+
+	ret = regmap_update_bits_check(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				       ARIZONA_MICD_ENA, ARIZONA_MICD_ENA,
+				       &change);
+	if (ret < 0) {
+		dev_err(arizona->dev, "Failed to enable micd: %d\n", ret);
+	} else if (!change) {
+		regulator_disable(info->micvdd);
+		pm_runtime_put_autosuspend(info->dev);
+	}
+}
+
+static void arizona_stop_mic(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	const char *widget = arizona_extcon_get_micbias(info);
+	struct snd_soc_dapm_context *dapm = arizona->dapm;
+	struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
+	bool change = false;
+	int ret;
+
+	ret = regmap_update_bits_check(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				       ARIZONA_MICD_ENA, 0,
+				       &change);
+	if (ret < 0)
+		dev_err(arizona->dev, "Failed to disable micd: %d\n", ret);
+
+	ret = snd_soc_component_disable_pin(component, widget);
+	if (ret != 0)
+		dev_warn(arizona->dev,
+			 "Failed to disable %s: %d\n",
+			 widget, ret);
+
+	snd_soc_dapm_sync(dapm);
+
+	if (info->micd_reva) {
+		const struct reg_sequence reva[] = {
+			{ 0x80,  0x3 },
+			{ 0x294, 0x2 },
+			{ 0x80,  0x0 },
+		};
+
+		regmap_multi_reg_write(arizona->regmap, reva, ARRAY_SIZE(reva));
+	}
+
+	ret = regulator_allow_bypass(info->micvdd, true);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to bypass MICVDD: %d\n",
+			ret);
+	}
+
+	if (change) {
+		regulator_disable(info->micvdd);
+		pm_runtime_mark_last_busy(info->dev);
+		pm_runtime_put_autosuspend(info->dev);
+	}
+}
+
+static struct {
+	unsigned int threshold;
+	unsigned int factor_a;
+	unsigned int factor_b;
+} arizona_hpdet_b_ranges[] = {
+	{ 100,  5528,   362464 },
+	{ 169, 11084,  6186851 },
+	{ 169, 11065, 65460395 },
+};
+
+#define ARIZONA_HPDET_B_RANGE_MAX 0x3fb
+
+static struct {
+	int min;
+	int max;
+} arizona_hpdet_c_ranges[] = {
+	{ 0,       30 },
+	{ 8,      100 },
+	{ 100,   1000 },
+	{ 1000, 10000 },
+};
+
+static int arizona_hpdet_read(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	unsigned int val, range;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, ARIZONA_HEADPHONE_DETECT_2, &val);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to read HPDET status: %d\n",
+			ret);
+		return ret;
+	}
+
+	switch (info->hpdet_ip_version) {
+	case 0:
+		if (!(val & ARIZONA_HP_DONE)) {
+			dev_err(arizona->dev, "HPDET did not complete: %x\n",
+				val);
+			return -EAGAIN;
+		}
+
+		val &= ARIZONA_HP_LVL_MASK;
+		break;
+
+	case 1:
+		if (!(val & ARIZONA_HP_DONE_B)) {
+			dev_err(arizona->dev, "HPDET did not complete: %x\n",
+				val);
+			return -EAGAIN;
+		}
+
+		ret = regmap_read(arizona->regmap, ARIZONA_HP_DACVAL, &val);
+		if (ret != 0) {
+			dev_err(arizona->dev, "Failed to read HP value: %d\n",
+				ret);
+			return -EAGAIN;
+		}
+
+		regmap_read(arizona->regmap, ARIZONA_HEADPHONE_DETECT_1,
+			    &range);
+		range = (range & ARIZONA_HP_IMPEDANCE_RANGE_MASK)
+			   >> ARIZONA_HP_IMPEDANCE_RANGE_SHIFT;
+
+		if (range < ARRAY_SIZE(arizona_hpdet_b_ranges) - 1 &&
+		    (val < arizona_hpdet_b_ranges[range].threshold ||
+		     val >= ARIZONA_HPDET_B_RANGE_MAX)) {
+			range++;
+			dev_dbg(arizona->dev, "Moving to HPDET range %d\n",
+				range);
+			regmap_update_bits(arizona->regmap,
+					   ARIZONA_HEADPHONE_DETECT_1,
+					   ARIZONA_HP_IMPEDANCE_RANGE_MASK,
+					   range <<
+					   ARIZONA_HP_IMPEDANCE_RANGE_SHIFT);
+			return -EAGAIN;
+		}
+
+		/* If we go out of range report top of range */
+		if (val < arizona_hpdet_b_ranges[range].threshold ||
+		    val >= ARIZONA_HPDET_B_RANGE_MAX) {
+			dev_dbg(arizona->dev, "Measurement out of range\n");
+			return ARIZONA_HPDET_MAX;
+		}
+
+		dev_dbg(arizona->dev, "HPDET read %d in range %d\n",
+			val, range);
+
+		val = arizona_hpdet_b_ranges[range].factor_b
+			/ ((val * 100) -
+			   arizona_hpdet_b_ranges[range].factor_a);
+		break;
+
+	case 2:
+		if (!(val & ARIZONA_HP_DONE_B)) {
+			dev_err(arizona->dev, "HPDET did not complete: %x\n",
+				val);
+			return -EAGAIN;
+		}
+
+		val &= ARIZONA_HP_LVL_B_MASK;
+		/* Convert to ohms, the value is in 0.5 ohm increments */
+		val /= 2;
+
+		regmap_read(arizona->regmap, ARIZONA_HEADPHONE_DETECT_1,
+			    &range);
+		range = (range & ARIZONA_HP_IMPEDANCE_RANGE_MASK)
+			   >> ARIZONA_HP_IMPEDANCE_RANGE_SHIFT;
+
+		/* Skip up a range, or report? */
+		if (range < ARRAY_SIZE(arizona_hpdet_c_ranges) - 1 &&
+		    (val >= arizona_hpdet_c_ranges[range].max)) {
+			range++;
+			dev_dbg(arizona->dev, "Moving to HPDET range %d-%d\n",
+				arizona_hpdet_c_ranges[range].min,
+				arizona_hpdet_c_ranges[range].max);
+			regmap_update_bits(arizona->regmap,
+					   ARIZONA_HEADPHONE_DETECT_1,
+					   ARIZONA_HP_IMPEDANCE_RANGE_MASK,
+					   range <<
+					   ARIZONA_HP_IMPEDANCE_RANGE_SHIFT);
+			return -EAGAIN;
+		}
+
+		if (range && (val < arizona_hpdet_c_ranges[range].min)) {
+			dev_dbg(arizona->dev, "Reporting range boundary %d\n",
+				arizona_hpdet_c_ranges[range].min);
+			val = arizona_hpdet_c_ranges[range].min;
+		}
+		break;
+
+	default:
+		dev_warn(arizona->dev, "Unknown HPDET IP revision %d\n",
+			 info->hpdet_ip_version);
+		return -EINVAL;
+	}
+
+	dev_dbg(arizona->dev, "HP impedance %d ohms\n", val);
+	return val;
+}
+
+static int arizona_hpdet_do_id(struct arizona_extcon_info *info, int *reading,
+			       bool *mic)
+{
+	struct arizona *arizona = info->arizona;
+	int id_gpio = arizona->pdata.hpdet_id_gpio;
+
+	if (!arizona->pdata.hpdet_acc_id)
+		return 0;
+
+	/*
+	 * If we're using HPDET for accessory identification we need
+	 * to take multiple measurements, step through them in sequence.
+	 */
+	info->hpdet_res[info->num_hpdet_res++] = *reading;
+
+	/* Only check the mic directly if we didn't already ID it */
+	if (id_gpio && info->num_hpdet_res == 1) {
+		dev_dbg(arizona->dev, "Measuring mic\n");
+
+		regmap_update_bits(arizona->regmap,
+				   ARIZONA_ACCESSORY_DETECT_MODE_1,
+				   ARIZONA_ACCDET_MODE_MASK |
+				   ARIZONA_ACCDET_SRC,
+				   ARIZONA_ACCDET_MODE_HPR |
+				   info->micd_modes[0].src);
+
+		gpio_set_value_cansleep(id_gpio, 1);
+
+		regmap_update_bits(arizona->regmap, ARIZONA_HEADPHONE_DETECT_1,
+				   ARIZONA_HP_POLL, ARIZONA_HP_POLL);
+		return -EAGAIN;
+	}
+
+	/* OK, got both.  Now, compare... */
+	dev_dbg(arizona->dev, "HPDET measured %d %d\n",
+		info->hpdet_res[0], info->hpdet_res[1]);
+
+	/* Take the headphone impedance for the main report */
+	*reading = info->hpdet_res[0];
+
+	/* Sometimes we get false readings due to slow insert */
+	if (*reading >= ARIZONA_HPDET_MAX && !info->hpdet_retried) {
+		dev_dbg(arizona->dev, "Retrying high impedance\n");
+		info->num_hpdet_res = 0;
+		info->hpdet_retried = true;
+		arizona_start_hpdet_acc_id(info);
+		pm_runtime_put(info->dev);
+		return -EAGAIN;
+	}
+
+	/*
+	 * If we measure the mic as high impedance
+	 */
+	if (!id_gpio || info->hpdet_res[1] > 50) {
+		dev_dbg(arizona->dev, "Detected mic\n");
+		*mic = true;
+		info->detecting = true;
+	} else {
+		dev_dbg(arizona->dev, "Detected headphone\n");
+	}
+
+	/* Make sure everything is reset back to the real polarity */
+	regmap_update_bits(arizona->regmap, ARIZONA_ACCESSORY_DETECT_MODE_1,
+			   ARIZONA_ACCDET_SRC, info->micd_modes[0].src);
+
+	return 0;
+}
+
+static irqreturn_t arizona_hpdet_irq(int irq, void *data)
+{
+	struct arizona_extcon_info *info = data;
+	struct arizona *arizona = info->arizona;
+	int id_gpio = arizona->pdata.hpdet_id_gpio;
+	unsigned int report = EXTCON_JACK_HEADPHONE;
+	int ret, reading, state;
+	bool mic = false;
+
+	mutex_lock(&info->lock);
+
+	/* If we got a spurious IRQ for some reason then ignore it */
+	if (!info->hpdet_active) {
+		dev_warn(arizona->dev, "Spurious HPDET IRQ\n");
+		mutex_unlock(&info->lock);
+		return IRQ_NONE;
+	}
+
+	/* If the cable was removed while measuring ignore the result */
+	state = extcon_get_state(info->edev, EXTCON_MECHANICAL);
+	if (state < 0) {
+		dev_err(arizona->dev, "Failed to check cable state: %d\n", state);
+		goto out;
+	} else if (!state) {
+		dev_dbg(arizona->dev, "Ignoring HPDET for removed cable\n");
+		goto done;
+	}
+
+	ret = arizona_hpdet_read(info);
+	if (ret == -EAGAIN)
+		goto out;
+	else if (ret < 0)
+		goto done;
+	reading = ret;
+
+	/* Reset back to starting range */
+	regmap_update_bits(arizona->regmap,
+			   ARIZONA_HEADPHONE_DETECT_1,
+			   ARIZONA_HP_IMPEDANCE_RANGE_MASK | ARIZONA_HP_POLL,
+			   0);
+
+	ret = arizona_hpdet_do_id(info, &reading, &mic);
+	if (ret == -EAGAIN)
+		goto out;
+	else if (ret < 0)
+		goto done;
+
+	/* Report high impedence cables as line outputs */
+	if (reading >= 5000)
+		report = EXTCON_JACK_LINE_OUT;
+	else
+		report = EXTCON_JACK_HEADPHONE;
+
+	ret = extcon_set_state_sync(info->edev, report, true);
+	if (ret != 0)
+		dev_err(arizona->dev, "Failed to report HP/line: %d\n",
+			ret);
+
+done:
+	/* Reset back to starting range */
+	regmap_update_bits(arizona->regmap,
+			   ARIZONA_HEADPHONE_DETECT_1,
+			   ARIZONA_HP_IMPEDANCE_RANGE_MASK | ARIZONA_HP_POLL,
+			   0);
+
+	arizona_extcon_hp_clamp(info, false);
+
+	if (id_gpio)
+		gpio_set_value_cansleep(id_gpio, 0);
+
+	/* If we have a mic then reenable MICDET */
+	if (state && (mic || info->mic))
+		arizona_start_mic(info);
+
+	if (info->hpdet_active) {
+		pm_runtime_put_autosuspend(info->dev);
+		info->hpdet_active = false;
+	}
+
+	/* Do not set hp_det done when the cable has been unplugged */
+	if (state)
+		info->hpdet_done = true;
+
+out:
+	mutex_unlock(&info->lock);
+
+	return IRQ_HANDLED;
+}
+
+static void arizona_identify_headphone(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	int ret;
+
+	if (info->hpdet_done)
+		return;
+
+	dev_dbg(arizona->dev, "Starting HPDET\n");
+
+	/* Make sure we keep the device enabled during the measurement */
+	pm_runtime_get_sync(info->dev);
+
+	info->hpdet_active = true;
+
+	arizona_stop_mic(info);
+
+	arizona_extcon_hp_clamp(info, true);
+
+	ret = regmap_update_bits(arizona->regmap,
+				 ARIZONA_ACCESSORY_DETECT_MODE_1,
+				 ARIZONA_ACCDET_MODE_MASK,
+				 arizona->pdata.hpdet_channel);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to set HPDET mode: %d\n", ret);
+		goto err;
+	}
+
+	ret = regmap_update_bits(arizona->regmap, ARIZONA_HEADPHONE_DETECT_1,
+				 ARIZONA_HP_POLL, ARIZONA_HP_POLL);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Can't start HPDETL measurement: %d\n",
+			ret);
+		goto err;
+	}
+
+	return;
+
+err:
+	arizona_extcon_hp_clamp(info, false);
+	pm_runtime_put_autosuspend(info->dev);
+
+	/* Just report headphone */
+	ret = extcon_set_state_sync(info->edev, EXTCON_JACK_HEADPHONE, true);
+	if (ret != 0)
+		dev_err(arizona->dev, "Failed to report headphone: %d\n", ret);
+
+	if (info->mic)
+		arizona_start_mic(info);
+
+	info->hpdet_active = false;
+}
+
+static void arizona_start_hpdet_acc_id(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	int hp_reading = 32;
+	bool mic;
+	int ret;
+
+	dev_dbg(arizona->dev, "Starting identification via HPDET\n");
+
+	/* Make sure we keep the device enabled during the measurement */
+	pm_runtime_get_sync(info->dev);
+
+	info->hpdet_active = true;
+
+	arizona_extcon_hp_clamp(info, true);
+
+	ret = regmap_update_bits(arizona->regmap,
+				 ARIZONA_ACCESSORY_DETECT_MODE_1,
+				 ARIZONA_ACCDET_SRC | ARIZONA_ACCDET_MODE_MASK,
+				 info->micd_modes[0].src |
+				 arizona->pdata.hpdet_channel);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to set HPDET mode: %d\n", ret);
+		goto err;
+	}
+
+	if (arizona->pdata.hpdet_acc_id_line) {
+		ret = regmap_update_bits(arizona->regmap,
+					 ARIZONA_HEADPHONE_DETECT_1,
+					 ARIZONA_HP_POLL, ARIZONA_HP_POLL);
+		if (ret != 0) {
+			dev_err(arizona->dev,
+				"Can't start HPDETL measurement: %d\n",
+				ret);
+			goto err;
+		}
+	} else {
+		arizona_hpdet_do_id(info, &hp_reading, &mic);
+	}
+
+	return;
+
+err:
+	/* Just report headphone */
+	ret = extcon_set_state_sync(info->edev, EXTCON_JACK_HEADPHONE, true);
+	if (ret != 0)
+		dev_err(arizona->dev, "Failed to report headphone: %d\n", ret);
+
+	info->hpdet_active = false;
+}
+
+static void arizona_micd_timeout_work(struct work_struct *work)
+{
+	struct arizona_extcon_info *info = container_of(work,
+						struct arizona_extcon_info,
+						micd_timeout_work.work);
+
+	mutex_lock(&info->lock);
+
+	dev_dbg(info->arizona->dev, "MICD timed out, reporting HP\n");
+
+	info->detecting = false;
+
+	arizona_identify_headphone(info);
+
+	mutex_unlock(&info->lock);
+}
+
+static int arizona_micd_adc_read(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	unsigned int val;
+	int ret;
+
+	/* Must disable MICD before we read the ADCVAL */
+	regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+			   ARIZONA_MICD_ENA, 0);
+
+	ret = regmap_read(arizona->regmap, ARIZONA_MIC_DETECT_4, &val);
+	if (ret != 0) {
+		dev_err(arizona->dev,
+			"Failed to read MICDET_ADCVAL: %d\n", ret);
+		return ret;
+	}
+
+	dev_dbg(arizona->dev, "MICDET_ADCVAL: %x\n", val);
+
+	val &= ARIZONA_MICDET_ADCVAL_MASK;
+	if (val < ARRAY_SIZE(arizona_micd_levels))
+		val = arizona_micd_levels[val];
+	else
+		val = INT_MAX;
+
+	if (val <= QUICK_HEADPHONE_MAX_OHM)
+		val = ARIZONA_MICD_STS | ARIZONA_MICD_LVL_0;
+	else if (val <= MICROPHONE_MIN_OHM)
+		val = ARIZONA_MICD_STS | ARIZONA_MICD_LVL_1;
+	else if (val <= MICROPHONE_MAX_OHM)
+		val = ARIZONA_MICD_STS | ARIZONA_MICD_LVL_8;
+	else
+		val = ARIZONA_MICD_LVL_8;
+
+	return val;
+}
+
+static int arizona_micd_read(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	unsigned int val = 0;
+	int ret, i;
+
+	for (i = 0; i < 10 && !(val & MICD_LVL_0_TO_8); i++) {
+		ret = regmap_read(arizona->regmap, ARIZONA_MIC_DETECT_3, &val);
+		if (ret != 0) {
+			dev_err(arizona->dev,
+				"Failed to read MICDET: %d\n", ret);
+			return ret;
+		}
+
+		dev_dbg(arizona->dev, "MICDET: %x\n", val);
+
+		if (!(val & ARIZONA_MICD_VALID)) {
+			dev_warn(arizona->dev,
+				 "Microphone detection state invalid\n");
+			return -EINVAL;
+		}
+	}
+
+	if (i == 10 && !(val & MICD_LVL_0_TO_8)) {
+		dev_err(arizona->dev, "Failed to get valid MICDET value\n");
+		return -EINVAL;
+	}
+
+	return val;
+}
+
+static int arizona_micdet_reading(void *priv)
+{
+	struct arizona_extcon_info *info = priv;
+	struct arizona *arizona = info->arizona;
+	int ret, val;
+
+	if (info->detecting && arizona->pdata.micd_software_compare)
+		ret = arizona_micd_adc_read(info);
+	else
+		ret = arizona_micd_read(info);
+	if (ret < 0)
+		return ret;
+
+	val = ret;
+
+	/* Due to jack detect this should never happen */
+	if (!(val & ARIZONA_MICD_STS)) {
+		dev_warn(arizona->dev, "Detected open circuit\n");
+		info->mic = false;
+		info->detecting = false;
+		arizona_identify_headphone(info);
+		return 0;
+	}
+
+	/* If we got a high impedence we should have a headset, report it. */
+	if (val & ARIZONA_MICD_LVL_8) {
+		info->mic = true;
+		info->detecting = false;
+
+		arizona_identify_headphone(info);
+
+		ret = extcon_set_state_sync(info->edev,
+					      EXTCON_JACK_MICROPHONE, true);
+		if (ret != 0)
+			dev_err(arizona->dev, "Headset report failed: %d\n",
+				ret);
+
+		/* Don't need to regulate for button detection */
+		ret = regulator_allow_bypass(info->micvdd, true);
+		if (ret != 0) {
+			dev_err(arizona->dev, "Failed to bypass MICVDD: %d\n",
+				ret);
+		}
+
+		return 0;
+	}
+
+	/* If we detected a lower impedence during initial startup
+	 * then we probably have the wrong polarity, flip it.  Don't
+	 * do this for the lowest impedences to speed up detection of
+	 * plain headphones.  If both polarities report a low
+	 * impedence then give up and report headphones.
+	 */
+	if (val & MICD_LVL_1_TO_7) {
+		if (info->jack_flips >= info->micd_num_modes * 10) {
+			dev_dbg(arizona->dev, "Detected HP/line\n");
+
+			info->detecting = false;
+
+			arizona_identify_headphone(info);
+		} else {
+			info->micd_mode++;
+			if (info->micd_mode == info->micd_num_modes)
+				info->micd_mode = 0;
+			arizona_extcon_set_mode(info, info->micd_mode);
+
+			info->jack_flips++;
+
+			if (arizona->pdata.micd_software_compare)
+				regmap_update_bits(arizona->regmap,
+						   ARIZONA_MIC_DETECT_1,
+						   ARIZONA_MICD_ENA,
+						   ARIZONA_MICD_ENA);
+
+			queue_delayed_work(system_power_efficient_wq,
+					   &info->micd_timeout_work,
+					   msecs_to_jiffies(arizona->pdata.micd_timeout));
+		}
+
+		return 0;
+	}
+
+	/*
+	 * If we're still detecting and we detect a short then we've
+	 * got a headphone.
+	 */
+	dev_dbg(arizona->dev, "Headphone detected\n");
+	info->detecting = false;
+
+	arizona_identify_headphone(info);
+
+	return 0;
+}
+
+static int arizona_button_reading(void *priv)
+{
+	struct arizona_extcon_info *info = priv;
+	struct arizona *arizona = info->arizona;
+	int val, key, lvl, i;
+
+	val = arizona_micd_read(info);
+	if (val < 0)
+		return val;
+
+	/*
+	 * If we're still detecting and we detect a short then we've
+	 * got a headphone.  Otherwise it's a button press.
+	 */
+	if (val & MICD_LVL_0_TO_7) {
+		if (info->mic) {
+			dev_dbg(arizona->dev, "Mic button detected\n");
+
+			lvl = val & ARIZONA_MICD_LVL_MASK;
+			lvl >>= ARIZONA_MICD_LVL_SHIFT;
+
+			for (i = 0; i < info->num_micd_ranges; i++)
+				input_report_key(info->input,
+						 info->micd_ranges[i].key, 0);
+
+			if (lvl && ffs(lvl) - 1 < info->num_micd_ranges) {
+				key = info->micd_ranges[ffs(lvl) - 1].key;
+				input_report_key(info->input, key, 1);
+				input_sync(info->input);
+			} else {
+				dev_err(arizona->dev, "Button out of range\n");
+			}
+		} else {
+			dev_warn(arizona->dev, "Button with no mic: %x\n",
+				 val);
+		}
+	} else {
+		dev_dbg(arizona->dev, "Mic button released\n");
+		for (i = 0; i < info->num_micd_ranges; i++)
+			input_report_key(info->input,
+					 info->micd_ranges[i].key, 0);
+		input_sync(info->input);
+		arizona_extcon_pulse_micbias(info);
+	}
+
+	return 0;
+}
+
+static void arizona_micd_detect(struct work_struct *work)
+{
+	struct arizona_extcon_info *info = container_of(work,
+						struct arizona_extcon_info,
+						micd_detect_work.work);
+	struct arizona *arizona = info->arizona;
+	int ret;
+
+	cancel_delayed_work_sync(&info->micd_timeout_work);
+
+	mutex_lock(&info->lock);
+
+	/* If the cable was removed while measuring ignore the result */
+	ret = extcon_get_state(info->edev, EXTCON_MECHANICAL);
+	if (ret < 0) {
+		dev_err(arizona->dev, "Failed to check cable state: %d\n",
+				ret);
+		mutex_unlock(&info->lock);
+		return;
+	} else if (!ret) {
+		dev_dbg(arizona->dev, "Ignoring MICDET for removed cable\n");
+		mutex_unlock(&info->lock);
+		return;
+	}
+
+	if (info->detecting)
+		arizona_micdet_reading(info);
+	else
+		arizona_button_reading(info);
+
+	pm_runtime_mark_last_busy(info->dev);
+	mutex_unlock(&info->lock);
+}
+
+static irqreturn_t arizona_micdet(int irq, void *data)
+{
+	struct arizona_extcon_info *info = data;
+	struct arizona *arizona = info->arizona;
+	int debounce = arizona->pdata.micd_detect_debounce;
+
+	cancel_delayed_work_sync(&info->micd_detect_work);
+	cancel_delayed_work_sync(&info->micd_timeout_work);
+
+	mutex_lock(&info->lock);
+	if (!info->detecting)
+		debounce = 0;
+	mutex_unlock(&info->lock);
+
+	if (debounce)
+		queue_delayed_work(system_power_efficient_wq,
+				   &info->micd_detect_work,
+				   msecs_to_jiffies(debounce));
+	else
+		arizona_micd_detect(&info->micd_detect_work.work);
+
+	return IRQ_HANDLED;
+}
+
+static void arizona_hpdet_work(struct work_struct *work)
+{
+	struct arizona_extcon_info *info = container_of(work,
+						struct arizona_extcon_info,
+						hpdet_work.work);
+
+	mutex_lock(&info->lock);
+	arizona_start_hpdet_acc_id(info);
+	mutex_unlock(&info->lock);
+}
+
+static int arizona_hpdet_wait(struct arizona_extcon_info *info)
+{
+	struct arizona *arizona = info->arizona;
+	unsigned int val;
+	int i, ret;
+
+	for (i = 0; i < ARIZONA_HPDET_WAIT_COUNT; i++) {
+		ret = regmap_read(arizona->regmap, ARIZONA_HEADPHONE_DETECT_2,
+				&val);
+		if (ret) {
+			dev_err(arizona->dev,
+				"Failed to read HPDET state: %d\n", ret);
+			return ret;
+		}
+
+		switch (info->hpdet_ip_version) {
+		case 0:
+			if (val & ARIZONA_HP_DONE)
+				return 0;
+			break;
+		default:
+			if (val & ARIZONA_HP_DONE_B)
+				return 0;
+			break;
+		}
+
+		msleep(ARIZONA_HPDET_WAIT_DELAY_MS);
+	}
+
+	dev_warn(arizona->dev, "HPDET did not appear to complete\n");
+
+	return -ETIMEDOUT;
+}
+
+static irqreturn_t arizona_jackdet(int irq, void *data)
+{
+	struct arizona_extcon_info *info = data;
+	struct arizona *arizona = info->arizona;
+	unsigned int val, present, mask;
+	bool cancelled_hp, cancelled_mic;
+	int ret, i;
+
+	cancelled_hp = cancel_delayed_work_sync(&info->hpdet_work);
+	cancelled_mic = cancel_delayed_work_sync(&info->micd_timeout_work);
+
+	pm_runtime_get_sync(info->dev);
+
+	mutex_lock(&info->lock);
+
+	if (info->micd_clamp) {
+		mask = ARIZONA_MICD_CLAMP_STS;
+		present = 0;
+	} else {
+		mask = ARIZONA_JD1_STS;
+		if (arizona->pdata.jd_invert)
+			present = 0;
+		else
+			present = ARIZONA_JD1_STS;
+	}
+
+	ret = regmap_read(arizona->regmap, ARIZONA_AOD_IRQ_RAW_STATUS, &val);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to read jackdet status: %d\n",
+			ret);
+		mutex_unlock(&info->lock);
+		pm_runtime_put_autosuspend(info->dev);
+		return IRQ_NONE;
+	}
+
+	val &= mask;
+	if (val == info->last_jackdet) {
+		dev_dbg(arizona->dev, "Suppressing duplicate JACKDET\n");
+		if (cancelled_hp)
+			queue_delayed_work(system_power_efficient_wq,
+					   &info->hpdet_work,
+					   msecs_to_jiffies(HPDET_DEBOUNCE));
+
+		if (cancelled_mic) {
+			int micd_timeout = arizona->pdata.micd_timeout;
+
+			queue_delayed_work(system_power_efficient_wq,
+					   &info->micd_timeout_work,
+					   msecs_to_jiffies(micd_timeout));
+		}
+
+		goto out;
+	}
+	info->last_jackdet = val;
+
+	if (info->last_jackdet == present) {
+		dev_dbg(arizona->dev, "Detected jack\n");
+		ret = extcon_set_state_sync(info->edev,
+					      EXTCON_MECHANICAL, true);
+
+		if (ret != 0)
+			dev_err(arizona->dev, "Mechanical report failed: %d\n",
+				ret);
+
+		info->detecting = true;
+		info->mic = false;
+		info->jack_flips = 0;
+
+		if (!arizona->pdata.hpdet_acc_id) {
+			arizona_start_mic(info);
+		} else {
+			queue_delayed_work(system_power_efficient_wq,
+					   &info->hpdet_work,
+					   msecs_to_jiffies(HPDET_DEBOUNCE));
+		}
+
+		if (info->micd_clamp || !arizona->pdata.jd_invert)
+			regmap_update_bits(arizona->regmap,
+					   ARIZONA_JACK_DETECT_DEBOUNCE,
+					   ARIZONA_MICD_CLAMP_DB |
+					   ARIZONA_JD1_DB, 0);
+	} else {
+		dev_dbg(arizona->dev, "Detected jack removal\n");
+
+		arizona_stop_mic(info);
+
+		info->num_hpdet_res = 0;
+		for (i = 0; i < ARRAY_SIZE(info->hpdet_res); i++)
+			info->hpdet_res[i] = 0;
+		info->mic = false;
+		info->hpdet_done = false;
+		info->hpdet_retried = false;
+
+		for (i = 0; i < info->num_micd_ranges; i++)
+			input_report_key(info->input,
+					 info->micd_ranges[i].key, 0);
+		input_sync(info->input);
+
+		for (i = 0; i < ARRAY_SIZE(arizona_cable) - 1; i++) {
+			ret = extcon_set_state_sync(info->edev,
+					arizona_cable[i], false);
+			if (ret != 0)
+				dev_err(arizona->dev,
+					"Removal report failed: %d\n", ret);
+		}
+
+		/*
+		 * If the jack was removed during a headphone detection we
+		 * need to wait for the headphone detection to finish, as
+		 * it can not be aborted. We don't want to be able to start
+		 * a new headphone detection from a fresh insert until this
+		 * one is finished.
+		 */
+		arizona_hpdet_wait(info);
+
+		regmap_update_bits(arizona->regmap,
+				   ARIZONA_JACK_DETECT_DEBOUNCE,
+				   ARIZONA_MICD_CLAMP_DB | ARIZONA_JD1_DB,
+				   ARIZONA_MICD_CLAMP_DB | ARIZONA_JD1_DB);
+	}
+
+out:
+	/* Clear trig_sts to make sure DCVDD is not forced up */
+	regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG,
+		     ARIZONA_MICD_CLAMP_FALL_TRIG_STS |
+		     ARIZONA_MICD_CLAMP_RISE_TRIG_STS |
+		     ARIZONA_JD1_FALL_TRIG_STS |
+		     ARIZONA_JD1_RISE_TRIG_STS);
+
+	mutex_unlock(&info->lock);
+
+	pm_runtime_mark_last_busy(info->dev);
+	pm_runtime_put_autosuspend(info->dev);
+
+	return IRQ_HANDLED;
+}
+
+/* Map a level onto a slot in the register bank */
+static void arizona_micd_set_level(struct arizona *arizona, int index,
+				   unsigned int level)
+{
+	int reg;
+	unsigned int mask;
+
+	reg = ARIZONA_MIC_DETECT_LEVEL_4 - (index / 2);
+
+	if (!(index % 2)) {
+		mask = 0x3f00;
+		level <<= 8;
+	} else {
+		mask = 0x3f;
+	}
+
+	/* Program the level itself */
+	regmap_update_bits(arizona->regmap, reg, mask, level);
+}
+
+static int arizona_extcon_get_micd_configs(struct device *dev,
+					   struct arizona *arizona)
+{
+	const char * const prop = "wlf,micd-configs";
+	const int entries_per_config = 3;
+	struct arizona_micd_config *micd_configs;
+	int nconfs, ret;
+	int i, j;
+	u32 *vals;
+
+	nconfs = device_property_count_u32(arizona->dev, prop);
+	if (nconfs <= 0)
+		return 0;
+
+	vals = kcalloc(nconfs, sizeof(u32), GFP_KERNEL);
+	if (!vals)
+		return -ENOMEM;
+
+	ret = device_property_read_u32_array(arizona->dev, prop, vals, nconfs);
+	if (ret < 0)
+		goto out;
+
+	nconfs /= entries_per_config;
+	micd_configs = devm_kcalloc(dev, nconfs, sizeof(*micd_configs),
+				    GFP_KERNEL);
+	if (!micd_configs) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	for (i = 0, j = 0; i < nconfs; ++i) {
+		micd_configs[i].src = vals[j++] ? ARIZONA_ACCDET_SRC : 0;
+		micd_configs[i].bias = vals[j++];
+		micd_configs[i].gpio = vals[j++];
+	}
+
+	arizona->pdata.micd_configs = micd_configs;
+	arizona->pdata.num_micd_configs = nconfs;
+
+out:
+	kfree(vals);
+	return ret;
+}
+
+static int arizona_extcon_device_get_pdata(struct device *dev,
+					   struct arizona *arizona)
+{
+	struct arizona_pdata *pdata = &arizona->pdata;
+	unsigned int val = ARIZONA_ACCDET_MODE_HPL;
+	int ret;
+
+	device_property_read_u32(arizona->dev, "wlf,hpdet-channel", &val);
+	switch (val) {
+	case ARIZONA_ACCDET_MODE_HPL:
+	case ARIZONA_ACCDET_MODE_HPR:
+		pdata->hpdet_channel = val;
+		break;
+	default:
+		dev_err(arizona->dev,
+			"Wrong wlf,hpdet-channel DT value %d\n", val);
+		pdata->hpdet_channel = ARIZONA_ACCDET_MODE_HPL;
+	}
+
+	device_property_read_u32(arizona->dev, "wlf,micd-detect-debounce",
+				 &pdata->micd_detect_debounce);
+
+	device_property_read_u32(arizona->dev, "wlf,micd-bias-start-time",
+				 &pdata->micd_bias_start_time);
+
+	device_property_read_u32(arizona->dev, "wlf,micd-rate",
+				 &pdata->micd_rate);
+
+	device_property_read_u32(arizona->dev, "wlf,micd-dbtime",
+				 &pdata->micd_dbtime);
+
+	device_property_read_u32(arizona->dev, "wlf,micd-timeout-ms",
+				 &pdata->micd_timeout);
+
+	pdata->micd_force_micbias = device_property_read_bool(arizona->dev,
+						"wlf,micd-force-micbias");
+
+	pdata->micd_software_compare = device_property_read_bool(arizona->dev,
+						"wlf,micd-software-compare");
+
+	pdata->jd_invert = device_property_read_bool(arizona->dev,
+						     "wlf,jd-invert");
+
+	device_property_read_u32(arizona->dev, "wlf,gpsw", &pdata->gpsw);
+
+	pdata->jd_gpio5 = device_property_read_bool(arizona->dev,
+						    "wlf,use-jd2");
+	pdata->jd_gpio5_nopull = device_property_read_bool(arizona->dev,
+						"wlf,use-jd2-nopull");
+
+	ret = arizona_extcon_get_micd_configs(dev, arizona);
+	if (ret < 0)
+		dev_err(arizona->dev, "Failed to read micd configs: %d\n", ret);
+
+	return 0;
+}
+
+static int arizona_extcon_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct arizona_pdata *pdata = &arizona->pdata;
+	struct arizona_extcon_info *info;
+	unsigned int val;
+	unsigned int clamp_mode;
+	int jack_irq_fall, jack_irq_rise;
+	int ret, mode, i, j;
+
+	if (!arizona->dapm || !arizona->dapm->card)
+		return -EPROBE_DEFER;
+
+	info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
+	if (!info)
+		return -ENOMEM;
+
+	if (!dev_get_platdata(arizona->dev))
+		arizona_extcon_device_get_pdata(&pdev->dev, arizona);
+
+	info->micvdd = devm_regulator_get(&pdev->dev, "MICVDD");
+	if (IS_ERR(info->micvdd)) {
+		ret = PTR_ERR(info->micvdd);
+		dev_err(arizona->dev, "Failed to get MICVDD: %d\n", ret);
+		return ret;
+	}
+
+	mutex_init(&info->lock);
+	info->arizona = arizona;
+	info->dev = &pdev->dev;
+	info->last_jackdet = ~(ARIZONA_MICD_CLAMP_STS | ARIZONA_JD1_STS);
+	INIT_DELAYED_WORK(&info->hpdet_work, arizona_hpdet_work);
+	INIT_DELAYED_WORK(&info->micd_detect_work, arizona_micd_detect);
+	INIT_DELAYED_WORK(&info->micd_timeout_work, arizona_micd_timeout_work);
+	platform_set_drvdata(pdev, info);
+
+	switch (arizona->type) {
+	case WM5102:
+		switch (arizona->rev) {
+		case 0:
+			info->micd_reva = true;
+			break;
+		default:
+			info->micd_clamp = true;
+			info->hpdet_ip_version = 1;
+			break;
+		}
+		break;
+	case WM5110:
+	case WM8280:
+		switch (arizona->rev) {
+		case 0 ... 2:
+			break;
+		default:
+			info->micd_clamp = true;
+			info->hpdet_ip_version = 2;
+			break;
+		}
+		break;
+	case WM8998:
+	case WM1814:
+		info->micd_clamp = true;
+		info->hpdet_ip_version = 2;
+		break;
+	default:
+		break;
+	}
+
+	info->edev = devm_extcon_dev_allocate(&pdev->dev, arizona_cable);
+	if (IS_ERR(info->edev)) {
+		dev_err(&pdev->dev, "failed to allocate extcon device\n");
+		return -ENOMEM;
+	}
+
+	ret = devm_extcon_dev_register(&pdev->dev, info->edev);
+	if (ret < 0) {
+		dev_err(arizona->dev, "extcon_dev_register() failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	info->input = devm_input_allocate_device(&pdev->dev);
+	if (!info->input) {
+		dev_err(arizona->dev, "Can't allocate input dev\n");
+		ret = -ENOMEM;
+		return ret;
+	}
+
+	info->input->name = "Headset";
+	info->input->phys = "arizona/extcon";
+
+	if (!pdata->micd_timeout)
+		pdata->micd_timeout = DEFAULT_MICD_TIMEOUT;
+
+	if (pdata->num_micd_configs) {
+		info->micd_modes = pdata->micd_configs;
+		info->micd_num_modes = pdata->num_micd_configs;
+	} else {
+		info->micd_modes = micd_default_modes;
+		info->micd_num_modes = ARRAY_SIZE(micd_default_modes);
+	}
+
+	if (arizona->pdata.gpsw > 0)
+		regmap_update_bits(arizona->regmap, ARIZONA_GP_SWITCH_1,
+				ARIZONA_SW1_MODE_MASK, arizona->pdata.gpsw);
+
+	if (pdata->micd_pol_gpio > 0) {
+		if (info->micd_modes[0].gpio)
+			mode = GPIOF_OUT_INIT_HIGH;
+		else
+			mode = GPIOF_OUT_INIT_LOW;
+
+		ret = devm_gpio_request_one(&pdev->dev, pdata->micd_pol_gpio,
+					    mode, "MICD polarity");
+		if (ret != 0) {
+			dev_err(arizona->dev, "Failed to request GPIO%d: %d\n",
+				pdata->micd_pol_gpio, ret);
+			return ret;
+		}
+
+		info->micd_pol_gpio = gpio_to_desc(pdata->micd_pol_gpio);
+	} else {
+		if (info->micd_modes[0].gpio)
+			mode = GPIOD_OUT_HIGH;
+		else
+			mode = GPIOD_OUT_LOW;
+
+		/* We can't use devm here because we need to do the get
+		 * against the MFD device, as that is where the of_node
+		 * will reside, but if we devm against that the GPIO
+		 * will not be freed if the extcon driver is unloaded.
+		 */
+		info->micd_pol_gpio = gpiod_get_optional(arizona->dev,
+							 "wlf,micd-pol",
+							 mode);
+		if (IS_ERR(info->micd_pol_gpio)) {
+			ret = PTR_ERR(info->micd_pol_gpio);
+			dev_err(arizona->dev,
+				"Failed to get microphone polarity GPIO: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	if (arizona->pdata.hpdet_id_gpio > 0) {
+		ret = devm_gpio_request_one(&pdev->dev,
+					    arizona->pdata.hpdet_id_gpio,
+					    GPIOF_OUT_INIT_LOW,
+					    "HPDET");
+		if (ret != 0) {
+			dev_err(arizona->dev, "Failed to request GPIO%d: %d\n",
+				arizona->pdata.hpdet_id_gpio, ret);
+			goto err_gpio;
+		}
+	}
+
+	if (arizona->pdata.micd_bias_start_time)
+		regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				   ARIZONA_MICD_BIAS_STARTTIME_MASK,
+				   arizona->pdata.micd_bias_start_time
+				   << ARIZONA_MICD_BIAS_STARTTIME_SHIFT);
+
+	if (arizona->pdata.micd_rate)
+		regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				   ARIZONA_MICD_RATE_MASK,
+				   arizona->pdata.micd_rate
+				   << ARIZONA_MICD_RATE_SHIFT);
+
+	switch (arizona->pdata.micd_dbtime) {
+	case MICD_DBTIME_FOUR_READINGS:
+		regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				   ARIZONA_MICD_DBTIME_MASK,
+				   ARIZONA_MICD_DBTIME);
+		break;
+	case MICD_DBTIME_TWO_READINGS:
+		regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				   ARIZONA_MICD_DBTIME_MASK, 0);
+		break;
+	default:
+		break;
+	}
+
+	BUILD_BUG_ON(ARRAY_SIZE(arizona_micd_levels) <
+		     ARIZONA_NUM_MICD_BUTTON_LEVELS);
+
+	if (arizona->pdata.num_micd_ranges) {
+		info->micd_ranges = pdata->micd_ranges;
+		info->num_micd_ranges = pdata->num_micd_ranges;
+	} else {
+		info->micd_ranges = micd_default_ranges;
+		info->num_micd_ranges = ARRAY_SIZE(micd_default_ranges);
+	}
+
+	if (arizona->pdata.num_micd_ranges > ARIZONA_MAX_MICD_RANGE) {
+		dev_err(arizona->dev, "Too many MICD ranges: %d\n",
+			arizona->pdata.num_micd_ranges);
+	}
+
+	if (info->num_micd_ranges > 1) {
+		for (i = 1; i < info->num_micd_ranges; i++) {
+			if (info->micd_ranges[i - 1].max >
+			    info->micd_ranges[i].max) {
+				dev_err(arizona->dev,
+					"MICD ranges must be sorted\n");
+				ret = -EINVAL;
+				goto err_gpio;
+			}
+		}
+	}
+
+	/* Disable all buttons by default */
+	regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_2,
+			   ARIZONA_MICD_LVL_SEL_MASK, 0x81);
+
+	/* Set up all the buttons the user specified */
+	for (i = 0; i < info->num_micd_ranges; i++) {
+		for (j = 0; j < ARIZONA_NUM_MICD_BUTTON_LEVELS; j++)
+			if (arizona_micd_levels[j] >= info->micd_ranges[i].max)
+				break;
+
+		if (j == ARIZONA_NUM_MICD_BUTTON_LEVELS) {
+			dev_err(arizona->dev, "Unsupported MICD level %d\n",
+				info->micd_ranges[i].max);
+			ret = -EINVAL;
+			goto err_gpio;
+		}
+
+		dev_dbg(arizona->dev, "%d ohms for MICD threshold %d\n",
+			arizona_micd_levels[j], i);
+
+		arizona_micd_set_level(arizona, i, j);
+		input_set_capability(info->input, EV_KEY,
+				     info->micd_ranges[i].key);
+
+		/* Enable reporting of that range */
+		regmap_update_bits(arizona->regmap, ARIZONA_MIC_DETECT_2,
+				   1 << i, 1 << i);
+	}
+
+	/* Set all the remaining keys to a maximum */
+	for (; i < ARIZONA_MAX_MICD_RANGE; i++)
+		arizona_micd_set_level(arizona, i, 0x3f);
+
+	/*
+	 * If we have a clamp use it, activating in conjunction with
+	 * GPIO5 if that is connected for jack detect operation.
+	 */
+	if (info->micd_clamp) {
+		if (arizona->pdata.jd_gpio5) {
+			/* Put the GPIO into input mode with optional pull */
+			val = 0xc101;
+			if (arizona->pdata.jd_gpio5_nopull)
+				val &= ~ARIZONA_GPN_PU;
+
+			regmap_write(arizona->regmap, ARIZONA_GPIO5_CTRL,
+				     val);
+
+			if (arizona->pdata.jd_invert)
+				clamp_mode = ARIZONA_MICD_CLAMP_MODE_JDH_GP5H;
+			else
+				clamp_mode = ARIZONA_MICD_CLAMP_MODE_JDL_GP5H;
+		} else {
+			if (arizona->pdata.jd_invert)
+				clamp_mode = ARIZONA_MICD_CLAMP_MODE_JDH;
+			else
+				clamp_mode = ARIZONA_MICD_CLAMP_MODE_JDL;
+		}
+
+		regmap_update_bits(arizona->regmap,
+				   ARIZONA_MICD_CLAMP_CONTROL,
+				   ARIZONA_MICD_CLAMP_MODE_MASK, clamp_mode);
+
+		regmap_update_bits(arizona->regmap,
+				   ARIZONA_JACK_DETECT_DEBOUNCE,
+				   ARIZONA_MICD_CLAMP_DB,
+				   ARIZONA_MICD_CLAMP_DB);
+	}
+
+	arizona_extcon_set_mode(info, 0);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+	pm_runtime_get_sync(&pdev->dev);
+
+	if (info->micd_clamp) {
+		jack_irq_rise = ARIZONA_IRQ_MICD_CLAMP_RISE;
+		jack_irq_fall = ARIZONA_IRQ_MICD_CLAMP_FALL;
+	} else {
+		jack_irq_rise = ARIZONA_IRQ_JD_RISE;
+		jack_irq_fall = ARIZONA_IRQ_JD_FALL;
+	}
+
+	ret = arizona_request_irq(arizona, jack_irq_rise,
+				  "JACKDET rise", arizona_jackdet, info);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to get JACKDET rise IRQ: %d\n",
+			ret);
+		goto err_pm;
+	}
+
+	ret = arizona_set_irq_wake(arizona, jack_irq_rise, 1);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to set JD rise IRQ wake: %d\n",
+			ret);
+		goto err_rise;
+	}
+
+	ret = arizona_request_irq(arizona, jack_irq_fall,
+				  "JACKDET fall", arizona_jackdet, info);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to get JD fall IRQ: %d\n", ret);
+		goto err_rise_wake;
+	}
+
+	ret = arizona_set_irq_wake(arizona, jack_irq_fall, 1);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to set JD fall IRQ wake: %d\n",
+			ret);
+		goto err_fall;
+	}
+
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_MICDET,
+				  "MICDET", arizona_micdet, info);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to get MICDET IRQ: %d\n", ret);
+		goto err_fall_wake;
+	}
+
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_HPDET,
+				  "HPDET", arizona_hpdet_irq, info);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to get HPDET IRQ: %d\n", ret);
+		goto err_micdet;
+	}
+
+	arizona_clk32k_enable(arizona);
+	regmap_update_bits(arizona->regmap, ARIZONA_JACK_DETECT_DEBOUNCE,
+			   ARIZONA_JD1_DB, ARIZONA_JD1_DB);
+	regmap_update_bits(arizona->regmap, ARIZONA_JACK_DETECT_ANALOGUE,
+			   ARIZONA_JD1_ENA, ARIZONA_JD1_ENA);
+
+	ret = regulator_allow_bypass(info->micvdd, true);
+	if (ret != 0)
+		dev_warn(arizona->dev, "Failed to set MICVDD to bypass: %d\n",
+			 ret);
+
+	ret = input_register_device(info->input);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't register input device: %d\n", ret);
+		goto err_hpdet;
+	}
+
+	pm_runtime_put(&pdev->dev);
+
+	return 0;
+
+err_hpdet:
+	arizona_free_irq(arizona, ARIZONA_IRQ_HPDET, info);
+err_micdet:
+	arizona_free_irq(arizona, ARIZONA_IRQ_MICDET, info);
+err_fall_wake:
+	arizona_set_irq_wake(arizona, jack_irq_fall, 0);
+err_fall:
+	arizona_free_irq(arizona, jack_irq_fall, info);
+err_rise_wake:
+	arizona_set_irq_wake(arizona, jack_irq_rise, 0);
+err_rise:
+	arizona_free_irq(arizona, jack_irq_rise, info);
+err_pm:
+	pm_runtime_put(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+err_gpio:
+	gpiod_put(info->micd_pol_gpio);
+	return ret;
+}
+
+static int arizona_extcon_remove(struct platform_device *pdev)
+{
+	struct arizona_extcon_info *info = platform_get_drvdata(pdev);
+	struct arizona *arizona = info->arizona;
+	int jack_irq_rise, jack_irq_fall;
+	bool change;
+	int ret;
+
+	if (info->micd_clamp) {
+		jack_irq_rise = ARIZONA_IRQ_MICD_CLAMP_RISE;
+		jack_irq_fall = ARIZONA_IRQ_MICD_CLAMP_FALL;
+	} else {
+		jack_irq_rise = ARIZONA_IRQ_JD_RISE;
+		jack_irq_fall = ARIZONA_IRQ_JD_FALL;
+	}
+
+	arizona_set_irq_wake(arizona, jack_irq_rise, 0);
+	arizona_set_irq_wake(arizona, jack_irq_fall, 0);
+	arizona_free_irq(arizona, ARIZONA_IRQ_HPDET, info);
+	arizona_free_irq(arizona, ARIZONA_IRQ_MICDET, info);
+	arizona_free_irq(arizona, jack_irq_rise, info);
+	arizona_free_irq(arizona, jack_irq_fall, info);
+	cancel_delayed_work_sync(&info->hpdet_work);
+	cancel_delayed_work_sync(&info->micd_detect_work);
+	cancel_delayed_work_sync(&info->micd_timeout_work);
+
+	ret = regmap_update_bits_check(arizona->regmap, ARIZONA_MIC_DETECT_1,
+				       ARIZONA_MICD_ENA, 0,
+				       &change);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to disable micd on remove: %d\n",
+			ret);
+	} else if (change) {
+		regulator_disable(info->micvdd);
+		pm_runtime_put(info->dev);
+	}
+
+	regmap_update_bits(arizona->regmap,
+			   ARIZONA_MICD_CLAMP_CONTROL,
+			   ARIZONA_MICD_CLAMP_MODE_MASK, 0);
+	regmap_update_bits(arizona->regmap, ARIZONA_JACK_DETECT_ANALOGUE,
+			   ARIZONA_JD1_ENA, 0);
+	arizona_clk32k_disable(arizona);
+
+	gpiod_put(info->micd_pol_gpio);
+
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver arizona_extcon_driver = {
+	.driver		= {
+		.name	= "arizona-extcon",
+	},
+	.probe		= arizona_extcon_probe,
+	.remove		= arizona_extcon_remove,
+};
+
+module_platform_driver(arizona_extcon_driver);
+
+MODULE_DESCRIPTION("Arizona Extcon driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:extcon-arizona");
-- 
cgit 


From 50b1affc891cbc103a2334ce909a026e25f4c84d Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Thu, 18 Mar 2021 13:20:08 +0000
Subject: ALSA: usb-audio: Fix unintentional sign extension issue

The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.

Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c7990 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 08873d2afe4d..ffd922327ae4 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2883,7 +2883,7 @@ static int snd_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v
 	u8 group = (private_value & SND_DJM_GROUP_MASK) >> SND_DJM_GROUP_SHIFT;
 	u16 value = elem->value.enumerated.item[0];
 
-	kctl->private_value = ((device << SND_DJM_DEVICE_SHIFT) |
+	kctl->private_value = (((unsigned long)device << SND_DJM_DEVICE_SHIFT) |
 			      (group << SND_DJM_GROUP_SHIFT) |
 			      value);
 
@@ -2921,7 +2921,7 @@ static int snd_djm_controls_create(struct usb_mixer_interface *mixer,
 		value = device->controls[i].default_value;
 		knew.name = device->controls[i].name;
 		knew.private_value = (
-			(device_idx << SND_DJM_DEVICE_SHIFT) |
+			((unsigned long)device_idx << SND_DJM_DEVICE_SHIFT) |
 			(i << SND_DJM_GROUP_SHIFT) |
 			value);
 		err = snd_djm_controls_update(mixer, device_idx, i, value);
-- 
cgit