From 8822702f6e4c8917c83ba79e0ebf2c8c218910d4 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 7 May 2021 10:44:52 +0800
Subject: ALSA: hda/realtek: reset eapd coeff to default value for alc287

Ubuntu users reported an audio bug on the Lenovo Yoga Slim 7 14IIL05,
he installed dual OS (Windows + Linux), if he booted to the Linux
from Windows, the Speaker can't work well, it has crackling noise,
if he poweroff the machine first after Windows, the Speaker worked
well.

Before rebooting or shutdown from Windows, the Windows changes the
codec eapd coeff value, but the BIOS doesn't re-initialize its value,
when booting into the Linux from Windows, the eapd coeff value is not
correct. To fix it, set the codec default value to that coeff register
in the alsa driver.

BugLink: http://bugs.launchpad.net/bugs/1925057
Suggested-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210507024452.8300-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d58f24c9702..a5f3e78ec04e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -395,7 +395,6 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0282:
 	case 0x10ec0283:
 	case 0x10ec0286:
-	case 0x10ec0287:
 	case 0x10ec0288:
 	case 0x10ec0285:
 	case 0x10ec0298:
@@ -406,6 +405,10 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0275:
 		alc_update_coef_idx(codec, 0xe, 0, 1<<0);
 		break;
+	case 0x10ec0287:
+		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
+		alc_write_coef_idx(codec, 0x8, 0x4ab7);
+		break;
 	case 0x10ec0293:
 		alc_update_coef_idx(codec, 0xa, 1<<13, 0);
 		break;
-- 
cgit v1.2.3


From c1b55029493879f5bd585ff79f326e71f0bc05e3 Mon Sep 17 00:00:00 2001
From: Daniel Cordova A <danesc87@gmail.com>
Date: Fri, 7 May 2021 12:31:16 -0500
Subject: ALSA: hda: fixup headset for ASUS GU502 laptop

The GU502 requires a few steps to make headset i/o works properly:
pincfg, verbs to unmute headphone out and callback to toggle output
between speakers and headphone using jack.

Signed-off-by: Daniel Cordova A <danesc87@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210507173116.12043-1-danesc87@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 62 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a5f3e78ec04e..b4b71609dff1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6254,6 +6254,35 @@ static void alc294_fixup_gx502_hp(struct hda_codec *codec,
 	}
 }
 
+static void alc294_gu502_toggle_output(struct hda_codec *codec,
+				       struct hda_jack_callback *cb)
+{
+	/* Windows sets 0x10 to 0x8420 for Node 0x20 which is
+	 * responsible from changes between speakers and headphones
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+		alc_write_coef_idx(codec, 0x10, 0x8420);
+	else
+		alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gu502_hp(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	if (!is_jack_detectable(codec, 0x21))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x21,
+				alc294_gu502_toggle_output);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		alc294_gu502_toggle_output(codec, NULL);
+		break;
+	}
+}
+
 static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
 			      const struct hda_fixup *fix, int action)
 {
@@ -6471,6 +6500,9 @@ enum {
 	ALC294_FIXUP_ASUS_GX502_HP,
 	ALC294_FIXUP_ASUS_GX502_PINS,
 	ALC294_FIXUP_ASUS_GX502_VERBS,
+	ALC294_FIXUP_ASUS_GU502_HP,
+	ALC294_FIXUP_ASUS_GU502_PINS,
+	ALC294_FIXUP_ASUS_GU502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_GPIO_LED,
@@ -7712,6 +7744,35 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc294_fixup_gx502_hp,
 	},
+	[ALC294_FIXUP_ASUS_GU502_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a11050 }, /* rear HP mic */
+			{ 0x1a, 0x01a11830 }, /* rear external mic */
+			{ 0x21, 0x012110f0 }, /* rear HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GU502_VERBS
+	},
+	[ALC294_FIXUP_ASUS_GU502_VERBS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* set 0x15 to HP-OUT ctrl */
+			{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+			/* unmute the 0x15 amp */
+			{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+			/* set 0x1b to HP-OUT */
+			{ 0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GU502_HP
+	},
+	[ALC294_FIXUP_ASUS_GU502_HP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc294_fixup_gu502_hp,
+	},
 	[ALC294_FIXUP_ASUS_COEF_1B] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -8256,6 +8317,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
+	SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
-- 
cgit v1.2.3


From e84749a78dc82bc545f12ce009e3dbcc2c5a8a91 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 May 2021 17:06:59 +0200
Subject: ALSA: usb-audio: Validate MS endpoint descriptors

snd_usbmidi_get_ms_info() may access beyond the border when a
malformed descriptor is passed.  This patch adds the sanity checks of
the given MS endpoint descriptors, and skips invalid ones.

Reported-by: syzbot+6bb23a5d5548b93c94aa@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210510150659.17710-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/midi.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index a10ac75969a8..649eb8d1ab7d 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1956,8 +1956,12 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi,
 		ms_ep = find_usb_ms_endpoint_descriptor(hostep);
 		if (!ms_ep)
 			continue;
+		if (ms_ep->bLength <= sizeof(*ms_ep))
+			continue;
 		if (ms_ep->bNumEmbMIDIJack > 0x10)
 			continue;
+		if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack)
+			continue;
 		if (usb_endpoint_dir_out(ep)) {
 			if (endpoints[epidx].out_ep) {
 				if (++epidx >= MIDI_MAX_ENDPOINTS) {
-- 
cgit v1.2.3


From 91e02557f377b6837d4f82b14229d92cae231001 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 May 2021 11:05:00 +0200
Subject: ALSA: usb-audio: Fix potential out-of-bounce access in MIDI EP parser

The recently introduced MIDI endpoint parser code has an access to the
field without the size validation, hence it might lead to
out-of-bounce access.  Add the sanity checks for the descriptor
sizes.

Fixes: eb596e0fd13c ("ALSA: usb-audio: generate midi streaming substream names from jack names")
Link: https://lore.kernel.org/r/20210511090500.2637-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/midi.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 649eb8d1ab7d..2c01649c70f6 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1750,7 +1750,7 @@ static struct usb_midi_in_jack_descriptor *find_usb_in_jack_descriptor(
 		struct usb_midi_in_jack_descriptor *injd =
 				(struct usb_midi_in_jack_descriptor *)extra;
 
-		if (injd->bLength > 4 &&
+		if (injd->bLength >= sizeof(*injd) &&
 		    injd->bDescriptorType == USB_DT_CS_INTERFACE &&
 		    injd->bDescriptorSubtype == UAC_MIDI_IN_JACK &&
 				injd->bJackID == jack_id)
@@ -1773,7 +1773,7 @@ static struct usb_midi_out_jack_descriptor *find_usb_out_jack_descriptor(
 		struct usb_midi_out_jack_descriptor *outjd =
 				(struct usb_midi_out_jack_descriptor *)extra;
 
-		if (outjd->bLength > 4 &&
+		if (outjd->bLength >= sizeof(*outjd) &&
 		    outjd->bDescriptorType == USB_DT_CS_INTERFACE &&
 		    outjd->bDescriptorSubtype == UAC_MIDI_OUT_JACK &&
 				outjd->bJackID == jack_id)
@@ -1820,7 +1820,8 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi,
 			outjd = find_usb_out_jack_descriptor(hostif, jack_id);
 			if (outjd) {
 				sz = USB_DT_MIDI_OUT_SIZE(outjd->bNrInputPins);
-				iJack = *(((uint8_t *) outjd) + sz - sizeof(uint8_t));
+				if (outjd->bLength >= sz)
+					iJack = *(((uint8_t *) outjd) + sz - sizeof(uint8_t));
 			}
 		} else {
 			/* and out jacks connect to ins */
-- 
cgit v1.2.3


From f2be77fee648ddd6d0d259d3527344ba0120e314 Mon Sep 17 00:00:00 2001
From: Elia Devito <eliadevito@gmail.com>
Date: Tue, 11 May 2021 14:46:49 +0200
Subject: ALSA: hda/realtek: Add fixup for HP Spectre x360 15-df0xxx

Fixup to enable all 4 speaker on HP Spectre x360 15-df0xxx and probably
on similar models.

0x14 pin config override is required to enable all speakers and
alc285-speaker2-to-dac1 fixup to enable volume adjustment.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=189331
Signed-off-by: Elia Devito <eliadevito@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210511124651.4802-1-eliadevito@gmail.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 b4b71609dff1..3e269de84079 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6542,6 +6542,7 @@ enum {
 	ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST,
 	ALC295_FIXUP_ASUS_DACS,
 	ALC295_FIXUP_HP_OMEN,
+	ALC285_FIXUP_HP_SPECTRE_X360,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8099,6 +8100,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED,
 	},
+	[ALC285_FIXUP_HP_SPECTRE_X360] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x90170110 }, /* enable top speaker */
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8259,6 +8269,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
@@ -8665,6 +8676,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"},
 	{.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"},
 	{.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"},
+	{.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From 4b059ce1f4b368208c2310925f49be77f15e527b Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:56:34 +0200
Subject: Revert "ALSA: sb: fix a missing check of snd_ctl_add"

This reverts commit beae77170c60aa786f3e4599c18ead2854d8694d.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It is safe to ignore this error as the
mixer element is optional, and the driver is very legacy.

Cc: Aditya Pakki <pakki001@umn.edu>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210503115736.2104747-8-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/isa/sb/sb16_main.c | 10 +++-------
 1 file changed, 3 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
index 38dc1fde25f3..aa4870531023 100644
--- a/sound/isa/sb/sb16_main.c
+++ b/sound/isa/sb/sb16_main.c
@@ -846,14 +846,10 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops);
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops);
 
-	if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) {
-		err = snd_ctl_add(card, snd_ctl_new1(
-					&snd_sb16_dma_control, chip));
-		if (err)
-			return err;
-	} else {
+	if (chip->dma16 >= 0 && chip->dma8 != chip->dma16)
+		snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip));
+	else
 		pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
-	}
 
 	snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
 				       card->dev, 64*1024, 128*1024);
-- 
cgit v1.2.3


From 1dacca7fa1ebea47d38d20cd2df37094805d2649 Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:56:59 +0200
Subject: Revert "ALSA: gus: add a check of the status of snd_ctl_add"

This reverts commit 0f25e000cb4398081748e54f62a902098aa79ec1.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It will be fixed up "correctly" in a
later kernel change.

The original commit did nothing if there was an error, except to print
out a message, which is pointless.  So remove the commit as it gives a
"false sense of doing something".

Cc: Kangjie Lu <kjlu@umn.edu>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210503115736.2104747-33-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/isa/gus/gus_main.c | 13 ++-----------
 1 file changed, 2 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index afc088f0377c..b7518122a10d 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -77,17 +77,8 @@ static const struct snd_kcontrol_new snd_gus_joystick_control = {
 
 static void snd_gus_init_control(struct snd_gus_card *gus)
 {
-	int ret;
-
-	if (!gus->ace_flag) {
-		ret =
-			snd_ctl_add(gus->card,
-					snd_ctl_new1(&snd_gus_joystick_control,
-						gus));
-		if (ret)
-			snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n",
-					ret);
-	}
+	if (!gus->ace_flag)
+		snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus));
 }
 
 /*
-- 
cgit v1.2.3


From 94f88309f201821073f57ae6005caefa61bf7b7e Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:57:01 +0200
Subject: Revert "ALSA: sb8: add a check for request_region"

This reverts commit dcd0feac9bab901d5739de51b3f69840851f8919.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It will be fixed up "correctly" in a
later kernel change.

The original commit message for this change was incorrect as the code
path can never result in a NULL dereference, alluding to the fact that
whatever tool was used to "find this" is broken.  It's just an optional
resource reservation, so removing this check is fine.

Cc: Kangjie Lu <kjlu@umn.edu>
Acked-by: Takashi Iwai <tiwai@suse.de>
Fixes: dcd0feac9bab ("ALSA: sb8: add a check for request_region")
Cc: stable <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210503115736.2104747-35-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/isa/sb/sb8.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 6c9d534ce8b6..95290ffe5c6e 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -95,10 +95,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev)
 
 	/* block the 0x388 port to avoid PnP conflicts */
 	acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
-	if (!acard->fm_res) {
-		err = -EBUSY;
-		goto _err;
-	}
 
 	if (port[dev] != SNDRV_AUTO_PORT) {
 		if ((err = snd_sbdsp_create(card, port[dev], irq[dev],
-- 
cgit v1.2.3


From a28591f61b60fac820c6de59826ffa710e5e314e Mon Sep 17 00:00:00 2001
From: Atul Gopinathan <atulgopinathan@gmail.com>
Date: Mon, 3 May 2021 13:57:02 +0200
Subject: ALSA: sb8: Add a comment note regarding an unused pointer

The field "fm_res" of "struct snd_sb8" is never used/dereferenced
throughout the sb8.c code. Therefore there is no need for any null value
check after the "request_region()".

Add a comment note to make developers know about this and prevent any
"NULL check" patches on this part of code.

Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Atul Gopinathan <atulgopinathan@gmail.com>
Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/isa/sb/sb8.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 95290ffe5c6e..ed3a87ebe3f4 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -93,7 +93,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev)
 	acard = card->private_data;
 	card->private_free = snd_sb8_free;
 
-	/* block the 0x388 port to avoid PnP conflicts */
+	/*
+	 * Block the 0x388 port to avoid PnP conflicts.
+	 * No need to check this value after request_region,
+	 * as we never do anything with it.
+	 */
 	acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
 
 	if (port[dev] != SNDRV_AUTO_PORT) {
-- 
cgit v1.2.3


From 4667a6fc1777ce071504bab570d3599107f4790f Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:57:03 +0200
Subject: Revert "ALSA: usx2y: Fix potential NULL pointer dereference"

This reverts commit a2c6433ee5a35a8de6d563f6512a26f87835ea0f.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It will be fixed up "correctly" in a
later kernel change.

The original patch was incorrect, and would leak memory if the error
path the patch added was hit.

Cc: Aditya Pakki <pakki001@umn.edu>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210503115736.2104747-37-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/usb/usx2y/usb_stream.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 091c071b270a..6bba17bf689a 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -91,12 +91,7 @@ static int init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize,
 
 	for (u = 0; u < USB_STREAM_NURBS; ++u) {
 		sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
-		if (!sk->inurb[u])
-			return -ENOMEM;
-
 		sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
-		if (!sk->outurb[u])
-			return -ENOMEM;
 	}
 
 	if (init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe) ||
-- 
cgit v1.2.3


From 1e0ce84215dbfd6065872e5d3755352da34f198b Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:57:21 +0200
Subject: Revert "ASoC: rt5645: fix a NULL pointer dereference"

This reverts commit 51dd97d1df5fb9ac58b9b358e63e67b530f6ae21.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It will be fixed up "correctly" in a
later kernel change.

Lots of things seem to be still allocated here and must be properly
cleaned up if an error happens here.

Cc: Kangjie Lu <kjlu@umn.edu>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210503115736.2104747-55-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/soc/codecs/rt5645.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9408ee63cb26..7cb90975009a 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3431,9 +3431,6 @@ static int rt5645_probe(struct snd_soc_component *component)
 		RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s),
 		GFP_KERNEL);
 
-	if (!rt5645->eq_param)
-		return -ENOMEM;
-
 	return 0;
 }
 
-- 
cgit v1.2.3


From 5e70b8e22b64eed13d5bbebcb5911dae65bf8c6b Mon Sep 17 00:00:00 2001
From: Phillip Potter <phil@philpotter.co.uk>
Date: Mon, 3 May 2021 13:57:22 +0200
Subject: ASoC: rt5645: add error checking to rt5645_probe function

Check for return value from various snd_soc_dapm_* calls, as many of
them can return errors and this should be handled. Also, reintroduce
the allocation failure check for rt5645->eq_param as well. Make all
areas where return values are checked lead to the end of the function
in the case of an error. Finally, introduce a comment explaining how
resources here are actually eventually cleaned up by the caller.

Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Phillip Potter <phil@philpotter.co.uk>
Link: https://lore.kernel.org/r/20210503115736.2104747-56-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/soc/codecs/rt5645.c | 48 ++++++++++++++++++++++++++++++++++++++---------
 1 file changed, 39 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7cb90975009a..438fa18bcb55 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3388,30 +3388,44 @@ static int rt5645_probe(struct snd_soc_component *component)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+	int ret = 0;
 
 	rt5645->component = component;
 
 	switch (rt5645->codec_type) {
 	case CODEC_TYPE_RT5645:
-		snd_soc_dapm_new_controls(dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			rt5645_specific_dapm_widgets,
 			ARRAY_SIZE(rt5645_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(dapm,
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_add_routes(dapm,
 			rt5645_specific_dapm_routes,
 			ARRAY_SIZE(rt5645_specific_dapm_routes));
+		if (ret < 0)
+			goto exit;
+
 		if (rt5645->v_id < 3) {
-			snd_soc_dapm_add_routes(dapm,
+			ret = snd_soc_dapm_add_routes(dapm,
 				rt5645_old_dapm_routes,
 				ARRAY_SIZE(rt5645_old_dapm_routes));
+			if (ret < 0)
+				goto exit;
 		}
 		break;
 	case CODEC_TYPE_RT5650:
-		snd_soc_dapm_new_controls(dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			rt5650_specific_dapm_widgets,
 			ARRAY_SIZE(rt5650_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(dapm,
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_add_routes(dapm,
 			rt5650_specific_dapm_routes,
 			ARRAY_SIZE(rt5650_specific_dapm_routes));
+		if (ret < 0)
+			goto exit;
 		break;
 	}
 
@@ -3419,9 +3433,17 @@ static int rt5645_probe(struct snd_soc_component *component)
 
 	/* for JD function */
 	if (rt5645->pdata.jd_mode) {
-		snd_soc_dapm_force_enable_pin(dapm, "JD Power");
-		snd_soc_dapm_force_enable_pin(dapm, "LDO2");
-		snd_soc_dapm_sync(dapm);
+		ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power");
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2");
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_sync(dapm);
+		if (ret < 0)
+			goto exit;
 	}
 
 	if (rt5645->pdata.long_name)
@@ -3431,7 +3453,15 @@ static int rt5645_probe(struct snd_soc_component *component)
 		RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s),
 		GFP_KERNEL);
 
-	return 0;
+	if (!rt5645->eq_param)
+		ret = -ENOMEM;
+exit:
+	/*
+	 * If there was an error above, everything will be cleaned up by the
+	 * caller if we return an error here.  This will be done with a later
+	 * call to rt5645_remove().
+	 */
+	return ret;
 }
 
 static void rt5645_remove(struct snd_soc_component *component)
-- 
cgit v1.2.3


From fdda0dd2686ecd1f2e616c9e0366ea71b40c485d Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:57:23 +0200
Subject: Revert "ASoC: cs43130: fix a NULL pointer dereference"

This reverts commit a2be42f18d409213bb7e7a736e3ef6ba005115bb.

Because of recent interactions with developers from @umn.edu, all
commits from them have been recently re-reviewed to ensure if they were
correct or not.

Upon review, this commit was found to be incorrect for the reasons
below, so it must be reverted.  It will be fixed up "correctly" in a
later kernel change.

The original patch here is not correct, sysfs files that were created
are not unwound.

Cc: Kangjie Lu <kjlu@umn.edu>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210503115736.2104747-57-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/soc/codecs/cs43130.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index 80bc7c10ed75..c2b6f0ae6d57 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -2319,8 +2319,6 @@ static int cs43130_probe(struct snd_soc_component *component)
 			return ret;
 
 		cs43130->wq = create_singlethread_workqueue("cs43130_hp");
-		if (!cs43130->wq)
-			return -ENOMEM;
 		INIT_WORK(&cs43130->work, cs43130_imp_meas);
 	}
 
-- 
cgit v1.2.3


From 2da441a6491d93eff8ffff523837fd621dc80389 Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Mon, 3 May 2021 13:57:24 +0200
Subject: ASoC: cs43130: handle errors in cs43130_probe() properly

cs43130_probe() does not do any valid error checking of things it
initializes, OR what it does, it does not unwind properly if there are
errors.

Fix this up by moving the sysfs files to an attribute group so the
driver core will correctly add/remove them all at once and handle errors
with them, and correctly check for creating a new workqueue and
unwinding if that fails.

Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210503115736.2104747-58-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/soc/codecs/cs43130.c | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index c2b6f0ae6d57..80cd3ea0c157 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL);
 static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL);
 static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL);
 
+static struct attribute *hpload_attrs[] = {
+	&dev_attr_hpload_dc_l.attr,
+	&dev_attr_hpload_dc_r.attr,
+	&dev_attr_hpload_ac_l.attr,
+	&dev_attr_hpload_ac_r.attr,
+};
+ATTRIBUTE_GROUPS(hpload);
+
 static struct reg_sequence hp_en_cal_seq[] = {
 	{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
 	{CS43130_HP_MEAS_LOAD_1, 0},
@@ -2302,23 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component)
 
 	cs43130->hpload_done = false;
 	if (cs43130->dc_meas) {
-		ret = device_create_file(component->dev, &dev_attr_hpload_dc_l);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_dc_r);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_ac_l);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_ac_r);
-		if (ret < 0)
+		ret = sysfs_create_groups(&component->dev->kobj, hpload_groups);
+		if (ret)
 			return ret;
 
 		cs43130->wq = create_singlethread_workqueue("cs43130_hp");
+		if (!cs43130->wq) {
+			sysfs_remove_groups(&component->dev->kobj, hpload_groups);
+			return -ENOMEM;
+		}
 		INIT_WORK(&cs43130->work, cs43130_imp_meas);
 	}
 
-- 
cgit v1.2.3


From 27b57bb76a897be80494ee11ee4e85326d19383d Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Thu, 13 May 2021 21:40:38 +0200
Subject: Revert "Revert "ALSA: usx2y: Fix potential NULL pointer dereference""

This reverts commit 4667a6fc1777ce071504bab570d3599107f4790f.

Takashi writes:
	I have already started working on the bigger cleanup of this driver
	code based on 5.13-rc1, so could you drop this revert?

I missed our previous discussion about this, my fault for applying it.

Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/usb/usx2y/usb_stream.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 6bba17bf689a..091c071b270a 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -91,7 +91,12 @@ static int init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize,
 
 	for (u = 0; u < USB_STREAM_NURBS; ++u) {
 		sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
+		if (!sk->inurb[u])
+			return -ENOMEM;
+
 		sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
+		if (!sk->outurb[u])
+			return -ENOMEM;
 	}
 
 	if (init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe) ||
-- 
cgit v1.2.3


From 1b6604896e78969baffc1b6cc6bc175f95929ac4 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 13 May 2021 21:56:48 +0900
Subject: ALSA: dice: fix stream format at middle sampling rate for Alesis iO
 26

Alesis iO 26 FireWire has two pairs of digital optical interface. It
delivers PCM frames from the interfaces by second isochronous packet
streaming. Although both of the interfaces are available at 44.1/48.0
kHz, first one of them is only available at 88.2/96.0 kHz. It reduces
the number of PCM samples to 4 in Multi Bit Linear Audio data channel
of data blocks on the second isochronous packet streaming.

This commit fixes hardcoded stream formats.

Cc: <stable@vger.kernel.org>
Fixes: 28b208f600a3 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-alesis.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
index 0916864511d5..27c13b9cc9ef 100644
--- a/sound/firewire/dice/dice-alesis.c
+++ b/sound/firewire/dice/dice-alesis.c
@@ -16,7 +16,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
 static const unsigned int
 alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
 	{10, 10, 4},	/* Tx0 = Analog + S/PDIF. */
-	{16, 8, 0},	/* Tx1 = ADAT1 + ADAT2. */
+	{16, 4, 0},	/* Tx1 = ADAT1 + ADAT2 (available at low rate). */
 };
 
 int snd_dice_detect_alesis_formats(struct snd_dice *dice)
-- 
cgit v1.2.3


From 0edabdfe89581669609eaac5f6a8d0ae6fe95e7f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 13 May 2021 21:56:49 +0900
Subject: ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro

Mackie d.2 has an extension card for IEEE 1394 communication, which uses
BridgeCo DM1000 ASIC. On the other hand, Mackie d.4 Pro has built-in
function for IEEE 1394 communication by Oxford Semiconductor OXFW971,
according to schematic diagram available in Mackie website. Although I
misunderstood that Mackie d.2 Pro would be also a model with OXFW971,
it's wrong. Mackie d.2 Pro is a model which includes the extension card
as factory settings.

This commit fixes entries in Kconfig and comment in ALSA OXFW driver.

Cc: <stable@vger.kernel.org>
Fixes: fd6f4b0dc167 ("ALSA: bebob: Add skelton for BeBoB based devices")
Fixes: ec4dba5053e1 ("ALSA: oxfw: Add support for Behringer/Mackie devices")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/Kconfig       | 4 ++--
 sound/firewire/bebob/bebob.c | 2 +-
 sound/firewire/oxfw/oxfw.c   | 1 -
 3 files changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 25778765cbfe..9897bd26a438 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -38,7 +38,7 @@ config SND_OXFW
 	   * Mackie(Loud) Onyx 1640i (former model)
 	   * Mackie(Loud) Onyx Satellite
 	   * Mackie(Loud) Tapco Link.Firewire
-	   * Mackie(Loud) d.2 pro/d.4 pro
+	   * Mackie(Loud) d.4 pro
 	   * Mackie(Loud) U.420/U.420d
 	   * TASCAM FireOne
 	   * Stanton Controllers & Systems 1 Deck/Mixer
@@ -84,7 +84,7 @@ config SND_BEBOB
 	  * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
 	  * BridgeCo RDAudio1/Audio5
 	  * Mackie Onyx 1220/1620/1640 (FireWire I/O Card)
-	  * Mackie d.2 (FireWire Option)
+	  * Mackie d.2 (FireWire Option) and d.2 Pro
 	  * Stanton FinalScratch 2 (ScratchAmp)
 	  * Tascam IF-FW/DM
 	  * Behringer XENIX UFX 1204/1604
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 2c8e3392a490..daeecfa8b9aa 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -387,7 +387,7 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
 	/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
 	SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
-	/* Mackie, d.2 (Firewire Option) */
+	// Mackie, d.2 (Firewire option card) and d.2 Pro (the card is built-in).
 	SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
 	/* Stanton, ScratchAmp */
 	SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 1f1e3236efb8..9eea25c46dc7 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -355,7 +355,6 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
 	 *  Onyx-i series (former models):	0x081216
 	 *  Mackie Onyx Satellite:		0x00200f
 	 *  Tapco LINK.firewire 4x6:		0x000460
-	 *  d.2 pro:				Unknown
 	 *  d.4 pro:				Unknown
 	 *  U.420:				Unknown
 	 *  U.420d:				Unknown
-- 
cgit v1.2.3


From 395f41e2cdac63e7581fb9574e5ac0f02556e34a Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 13 May 2021 21:56:50 +0900
Subject: ALSA: firewire-lib: fix check for the size of isochronous packet
 payload

The check for size of isochronous packet payload just cares of the size of
IR context payload without the size of CIP header.

Cc: <stable@vger.kernel.org>
Fixes: f11453c7cc01 ("ALSA: firewire-lib: use 16 bytes IR context header to separate CIP header")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 4e2f2bb7879f..b53971bf4b90 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -633,18 +633,24 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 			       unsigned int *syt, unsigned int index)
 {
 	const __be32 *cip_header;
+	unsigned int cip_header_size;
 	int err;
 
 	*payload_length = be32_to_cpu(ctx_header[0]) >> ISO_DATA_LENGTH_SHIFT;
-	if (*payload_length > s->ctx_data.tx.ctx_header_size +
-					s->ctx_data.tx.max_ctx_payload_length) {
+
+	if (!(s->flags & CIP_NO_HEADER))
+		cip_header_size = 8;
+	else
+		cip_header_size = 0;
+
+	if (*payload_length > cip_header_size + s->ctx_data.tx.max_ctx_payload_length) {
 		dev_err(&s->unit->device,
 			"Detect jumbo payload: %04x %04x\n",
-			*payload_length, s->ctx_data.tx.max_ctx_payload_length);
+			*payload_length, cip_header_size + s->ctx_data.tx.max_ctx_payload_length);
 		return -EIO;
 	}
 
-	if (!(s->flags & CIP_NO_HEADER)) {
+	if (cip_header_size > 0) {
 		cip_header = ctx_header + 2;
 		err = check_cip_header(s, cip_header, *payload_length,
 				       data_blocks, data_block_counter, syt);
-- 
cgit v1.2.3


From 1be4f21d9984fa9835fae5411a29465dc5aece6f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 13 May 2021 21:56:51 +0900
Subject: ALSA: firewire-lib: fix calculation for size of IR context payload

The quadlets for CIP header is handled as a part of IR context header,
thus it doesn't join in IR context payload. However current calculation
includes the quadlets in IR context payload.

Cc: <stable@vger.kernel.org>
Fixes: f11453c7cc01 ("ALSA: firewire-lib: use 16 bytes IR context header to separate CIP header")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 13 ++++++-------
 1 file changed, 6 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index b53971bf4b90..73aff017dc9a 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1071,23 +1071,22 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed,
 		s->data_block_counter = 0;
 	}
 
-	/* initialize packet buffer */
+	// initialize packet buffer.
+	max_ctx_payload_size = amdtp_stream_get_max_payload(s);
 	if (s->direction == AMDTP_IN_STREAM) {
 		dir = DMA_FROM_DEVICE;
 		type = FW_ISO_CONTEXT_RECEIVE;
-		if (!(s->flags & CIP_NO_HEADER))
+		if (!(s->flags & CIP_NO_HEADER)) {
+			max_ctx_payload_size -= 8;
 			ctx_header_size = IR_CTX_HEADER_SIZE_CIP;
-		else
+		} else {
 			ctx_header_size = IR_CTX_HEADER_SIZE_NO_CIP;
-
-		max_ctx_payload_size = amdtp_stream_get_max_payload(s) -
-				       ctx_header_size;
+		}
 	} else {
 		dir = DMA_TO_DEVICE;
 		type = FW_ISO_CONTEXT_TRANSMIT;
 		ctx_header_size = 0;	// No effect for IT context.
 
-		max_ctx_payload_size = amdtp_stream_get_max_payload(s);
 		if (!(s->flags & CIP_NO_HEADER))
 			max_ctx_payload_size -= IT_PKT_HEADER_SIZE_CIP;
 	}
-- 
cgit v1.2.3


From 814b43127f4ac69332e809152e30773941438aff Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 13 May 2021 21:56:52 +0900
Subject: ALSA: firewire-lib: fix amdtp_packet tracepoints event for
 packet_index field

The snd_firewire_lib:amdtp_packet tracepoints event includes index of
packet processed in a context handling. However in IR context, it is not
calculated as expected.

Cc: <stable@vger.kernel.org>
Fixes: 753e717986c2 ("ALSA: firewire-lib: use packet descriptor for IR context")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream-trace.h |  6 +++---
 sound/firewire/amdtp-stream.c       | 15 +++++++++------
 2 files changed, 12 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h
index 26e7cb555d3c..aa53c13b89d3 100644
--- a/sound/firewire/amdtp-stream-trace.h
+++ b/sound/firewire/amdtp-stream-trace.h
@@ -14,8 +14,8 @@
 #include <linux/tracepoint.h>
 
 TRACE_EVENT(amdtp_packet,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int data_block_counter, unsigned int index),
-	TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, data_block_counter, index),
+	TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int data_block_counter, unsigned int packet_index, unsigned int index),
+	TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, data_block_counter, packet_index, index),
 	TP_STRUCT__entry(
 		__field(unsigned int, second)
 		__field(unsigned int, cycle)
@@ -48,7 +48,7 @@ TRACE_EVENT(amdtp_packet,
 		__entry->payload_quadlets = payload_length / sizeof(__be32);
 		__entry->data_blocks = data_blocks;
 		__entry->data_block_counter = data_block_counter,
-		__entry->packet_index = s->packet_index;
+		__entry->packet_index = packet_index;
 		__entry->irq = !!in_interrupt();
 		__entry->index = index;
 	),
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 73aff017dc9a..e0faa6601966 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -526,7 +526,7 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 	}
 
 	trace_amdtp_packet(s, cycle, cip_header, payload_length, data_blocks,
-			   data_block_counter, index);
+			   data_block_counter, s->packet_index, index);
 }
 
 static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
@@ -630,7 +630,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 			       unsigned int *payload_length,
 			       unsigned int *data_blocks,
 			       unsigned int *data_block_counter,
-			       unsigned int *syt, unsigned int index)
+			       unsigned int *syt, unsigned int packet_index, unsigned int index)
 {
 	const __be32 *cip_header;
 	unsigned int cip_header_size;
@@ -668,7 +668,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 	}
 
 	trace_amdtp_packet(s, cycle, cip_header, *payload_length, *data_blocks,
-			   *data_block_counter, index);
+			   *data_block_counter, packet_index, index);
 
 	return err;
 }
@@ -707,12 +707,13 @@ static int generate_device_pkt_descs(struct amdtp_stream *s,
 				     unsigned int packets)
 {
 	unsigned int dbc = s->data_block_counter;
+	unsigned int packet_index = s->packet_index;
+	unsigned int queue_size = s->queue_size;
 	int i;
 	int err;
 
 	for (i = 0; i < packets; ++i) {
 		struct pkt_desc *desc = descs + i;
-		unsigned int index = (s->packet_index + i) % s->queue_size;
 		unsigned int cycle;
 		unsigned int payload_length;
 		unsigned int data_blocks;
@@ -721,7 +722,7 @@ static int generate_device_pkt_descs(struct amdtp_stream *s,
 		cycle = compute_cycle_count(ctx_header[1]);
 
 		err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length,
-					  &data_blocks, &dbc, &syt, i);
+					  &data_blocks, &dbc, &syt, packet_index, i);
 		if (err < 0)
 			return err;
 
@@ -729,13 +730,15 @@ static int generate_device_pkt_descs(struct amdtp_stream *s,
 		desc->syt = syt;
 		desc->data_blocks = data_blocks;
 		desc->data_block_counter = dbc;
-		desc->ctx_payload = s->buffer.packets[index].buffer;
+		desc->ctx_payload = s->buffer.packets[packet_index].buffer;
 
 		if (!(s->flags & CIP_DBC_IS_END_EVENT))
 			dbc = (dbc + desc->data_blocks) & 0xff;
 
 		ctx_header +=
 			s->ctx_data.tx.ctx_header_size / sizeof(*ctx_header);
+
+		packet_index = (packet_index + 1) % queue_size;
 	}
 
 	s->data_block_counter = dbc;
-- 
cgit v1.2.3


From 1d5cfca286178ce81fb0c8a5f5777ef123cd69e4 Mon Sep 17 00:00:00 2001
From: PeiSen Hou <pshou@realtek.com>
Date: Fri, 14 May 2021 12:50:48 +0200
Subject: ALSA: hda/realtek: Add some CLOVE SSIDs of ALC293

Fix "use as headset mic, without its own jack detect" problen.

Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d0746eaf29f248a5acc30313e3ba4f99@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3e269de84079..552e2cb73291 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8385,12 +8385,19 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x50f5, "Clevo NH55EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x50f6, "Clevo NH55DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
@@ -8408,9 +8415,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x951d, "Clevo N950T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL5XNU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xc018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS),
 	SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
-- 
cgit v1.2.3


From c1f0616124c455c5c762b6f123e40bba5df759e6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 16 May 2021 18:17:55 +0200
Subject: ALSA: intel8x0: Don't update period unless prepared

The interrupt handler of intel8x0 calls snd_intel8x0_update() whenever
the hardware sets the corresponding status bit for each stream.  This
works fine for most cases as long as the hardware behaves properly.
But when the hardware gives a wrong bit set, this leads to a zero-
division Oops, and reportedly, this seems what happened on a VM.

For fixing the crash, this patch adds a internal flag indicating that
the stream is ready to be updated, and check it (as well as the flag
being in suspended) to ignore such spurious update.

Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Sergey Senozhatsky <senozhatsky@chromium.org>
Link: https://lore.kernel.org/r/s5h5yzi7uh0.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 35903d1a1cbd..5b124c4ad572 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -331,6 +331,7 @@ struct ichdev {
 	unsigned int ali_slot;			/* ALI DMA slot */
 	struct ac97_pcm *pcm;
 	int pcm_open_flag;
+	unsigned int prepared:1;
 	unsigned int suspended: 1;
 };
 
@@ -691,6 +692,9 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich
 	int status, civ, i, step;
 	int ack = 0;
 
+	if (!ichdev->prepared || ichdev->suspended)
+		return;
+
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	status = igetbyte(chip, port + ichdev->roff_sr);
 	civ = igetbyte(chip, port + ICH_REG_OFF_CIV);
@@ -881,6 +885,7 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
 	if (ichdev->pcm_open_flag) {
 		snd_ac97_pcm_close(ichdev->pcm);
 		ichdev->pcm_open_flag = 0;
+		ichdev->prepared = 0;
 	}
 	err = snd_ac97_pcm_open(ichdev->pcm, params_rate(hw_params),
 				params_channels(hw_params),
@@ -902,6 +907,7 @@ static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
 	if (ichdev->pcm_open_flag) {
 		snd_ac97_pcm_close(ichdev->pcm);
 		ichdev->pcm_open_flag = 0;
+		ichdev->prepared = 0;
 	}
 	return 0;
 }
@@ -976,6 +982,7 @@ static int snd_intel8x0_pcm_prepare(struct snd_pcm_substream *substream)
 			ichdev->pos_shift = (runtime->sample_bits > 16) ? 2 : 1;
 	}
 	snd_intel8x0_setup_periods(chip, ichdev);
+	ichdev->prepared = 1;
 	return 0;
 }
 
-- 
cgit v1.2.3


From 9f079c1bdc9087842dc5ac9d81b1d7f2578e81ce Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 18 May 2021 10:25:10 +0900
Subject: ALSA: dice: disable double_pcm_frames mode for M-Audio Profire 610,
 2626 and Avid M-Box 3 Pro

ALSA dice driver detects jumbo payload at high sampling transfer frequency
for below models:

 * Avid M-Box 3 Pro
 * M-Audio Profire 610
 * M-Audio Profire 2626

Although many DICE-based devices have a quirk at high sampling transfer
frequency to multiplex double number of PCM frames into data block than
the number in IEC 61883-1/6, the above devices are just compliant to
IEC 61883-1/6.

This commit disables the mode of double_pcm_frames for the models.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518012510.37126-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-pcm.c    |  4 ++--
 sound/firewire/dice/dice-stream.c |  2 +-
 sound/firewire/dice/dice.c        | 24 ++++++++++++++++++++++++
 sound/firewire/dice/dice.h        |  3 ++-
 4 files changed, 29 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index af8a90ee40f3..a69ca1111b03 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -218,7 +218,7 @@ static int pcm_open(struct snd_pcm_substream *substream)
 
 		if (frames_per_period > 0) {
 			// For double_pcm_frame quirk.
-			if (rate > 96000) {
+			if (rate > 96000 && !dice->disable_double_pcm_frames) {
 				frames_per_period *= 2;
 				frames_per_buffer *= 2;
 			}
@@ -273,7 +273,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream,
 
 		mutex_lock(&dice->mutex);
 		// For double_pcm_frame quirk.
-		if (rate > 96000) {
+		if (rate > 96000 && !dice->disable_double_pcm_frames) {
 			events_per_period /= 2;
 			events_per_buffer /= 2;
 		}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 1a14c083e8ce..c4dfe76500c2 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -181,7 +181,7 @@ static int keep_resources(struct snd_dice *dice, struct amdtp_stream *stream,
 	// as 'Dual Wire'.
 	// For this quirk, blocking mode is required and PCM buffer size should
 	// be aligned to SYT_INTERVAL.
-	double_pcm_frames = rate > 96000;
+	double_pcm_frames = (rate > 96000 && !dice->disable_double_pcm_frames);
 	if (double_pcm_frames) {
 		rate /= 2;
 		pcm_chs *= 2;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 107a81691f0e..239d164b0eea 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -21,6 +21,7 @@ MODULE_LICENSE("GPL v2");
 #define OUI_SSL			0x0050c2	// Actually ID reserved by IEEE.
 #define OUI_PRESONUS		0x000a92
 #define OUI_HARMAN		0x000fd7
+#define OUI_AVID		0x00a07e
 
 #define DICE_CATEGORY_ID	0x04
 #define WEISS_CATEGORY_ID	0x00
@@ -222,6 +223,14 @@ static int dice_probe(struct fw_unit *unit,
 				(snd_dice_detect_formats_t)entry->driver_data;
 	}
 
+	// Below models are compliant to IEC 61883-1/6 and have no quirk at high sampling transfer
+	// frequency.
+	// * Avid M-Box 3 Pro
+	// * M-Audio Profire 610
+	// * M-Audio Profire 2626
+	if (entry->vendor_id == OUI_MAUDIO || entry->vendor_id == OUI_AVID)
+		dice->disable_double_pcm_frames = true;
+
 	spin_lock_init(&dice->lock);
 	mutex_init(&dice->mutex);
 	init_completion(&dice->clock_accepted);
@@ -278,7 +287,22 @@ static void dice_bus_reset(struct fw_unit *unit)
 
 #define DICE_INTERFACE	0x000001
 
+#define DICE_DEV_ENTRY_TYPICAL(vendor, model, data) \
+	{ \
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID | \
+				  IEEE1394_MATCH_MODEL_ID | \
+				  IEEE1394_MATCH_SPECIFIER_ID | \
+				  IEEE1394_MATCH_VERSION, \
+		.vendor_id	= (vendor), \
+		.model_id	= (model), \
+		.specifier_id	= (vendor), \
+		.version	= DICE_INTERFACE, \
+		.driver_data = (kernel_ulong_t)(data), \
+	}
+
 static const struct ieee1394_device_id dice_id_table[] = {
+	// Avid M-Box 3 Pro. To match in probe function.
+	DICE_DEV_ENTRY_TYPICAL(OUI_AVID, 0x000004, snd_dice_detect_extension_formats),
 	/* M-Audio Profire 2626 has a different value in version field. */
 	{
 		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index adc6f7c84460..3c967d1b3605 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -109,7 +109,8 @@ struct snd_dice {
 	struct fw_iso_resources rx_resources[MAX_STREAMS];
 	struct amdtp_stream tx_stream[MAX_STREAMS];
 	struct amdtp_stream rx_stream[MAX_STREAMS];
-	bool global_enabled;
+	bool global_enabled:1;
+	bool disable_double_pcm_frames:1;
 	struct completion clock_accepted;
 	unsigned int substreams_counter;
 
-- 
cgit v1.2.3


From 4c6fe8c547e3c9e8c15dabdd23c569ee0df3adb1 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 18 May 2021 10:26:12 +0900
Subject: ALSA: dice: fix stream format for TC Electronic Konnekt Live at high
 sampling transfer frequency

At high sampling transfer frequency, TC Electronic Konnekt Live
transfers/receives 6 audio data frames in multi bit linear audio data
channel of data block in CIP payload. Current hard-coded stream format
is wrong.

Cc: <stable@vger.kernel.org>
Fixes: f1f0f330b1d0 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518012612.37268-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-tcelectronic.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c
index a8875d24ba2a..43a3bcb15b3d 100644
--- a/sound/firewire/dice/dice-tcelectronic.c
+++ b/sound/firewire/dice/dice-tcelectronic.c
@@ -38,8 +38,8 @@ static const struct dice_tc_spec konnekt_24d = {
 };
 
 static const struct dice_tc_spec konnekt_live = {
-	.tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
-	.rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+	.tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+	.rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
 	.has_midi = true,
 };
 
-- 
cgit v1.2.3


From 05ca447630334c323c9e2b788b61133ab75d60d3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 18 May 2021 10:39:39 +0200
Subject: ALSA: line6: Fix racy initialization of LINE6 MIDI

The initialization of MIDI devices that are found on some LINE6
drivers are currently done in a racy way; namely, the MIDI buffer
instance is allocated and initialized in each private_init callback
while the communication with the interface is already started via
line6_init_cap_control() call before that point.  This may lead to
Oops in line6_data_received() when a spurious event is received, as
reported by syzkaller.

This patch moves the MIDI initialization to line6_init_cap_control()
as well instead of the too-lately-called private_init for avoiding the
race.  Also this reduces slightly more lines, so it's a win-win
change.

Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com
Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com
Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo
Cc: Hyeonggon Yoo <42.hyeyoo@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210518083939.1927-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/line6/driver.c | 4 ++++
 sound/usb/line6/pod.c    | 5 -----
 sound/usb/line6/variax.c | 6 ------
 3 files changed, 4 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index a030dd65eb28..9602929b7de9 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -699,6 +699,10 @@ static int line6_init_cap_control(struct usb_line6 *line6)
 		line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL);
 		if (!line6->buffer_message)
 			return -ENOMEM;
+
+		ret = line6_init_midi(line6);
+		if (ret < 0)
+			return ret;
 	} else {
 		ret = line6_hwdep_init(line6);
 		if (ret < 0)
diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c
index cd44cb5f1310..16e644330c4d 100644
--- a/sound/usb/line6/pod.c
+++ b/sound/usb/line6/pod.c
@@ -376,11 +376,6 @@ static int pod_init(struct usb_line6 *line6,
 	if (err < 0)
 		return err;
 
-	/* initialize MIDI subsystem: */
-	err = line6_init_midi(line6);
-	if (err < 0)
-		return err;
-
 	/* initialize PCM subsystem: */
 	err = line6_init_pcm(line6, &pod_pcm_properties);
 	if (err < 0)
diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c
index ed158f04de80..c2245aa93b08 100644
--- a/sound/usb/line6/variax.c
+++ b/sound/usb/line6/variax.c
@@ -159,7 +159,6 @@ static int variax_init(struct usb_line6 *line6,
 		       const struct usb_device_id *id)
 {
 	struct usb_line6_variax *variax = line6_to_variax(line6);
-	int err;
 
 	line6->process_message = line6_variax_process_message;
 	line6->disconnect = line6_variax_disconnect;
@@ -172,11 +171,6 @@ static int variax_init(struct usb_line6 *line6,
 	if (variax->buffer_activate == NULL)
 		return -ENOMEM;
 
-	/* initialize MIDI subsystem: */
-	err = line6_init_midi(&variax->line6);
-	if (err < 0)
-		return err;
-
 	/* initiate startup procedure: */
 	schedule_delayed_work(&line6->startup_work,
 			      msecs_to_jiffies(VARIAX_STARTUP_DELAY1));
-- 
cgit v1.2.3


From 51cb8e206afd463e66f16869e5ddc95bef107142 Mon Sep 17 00:00:00 2001
From: Johan Hovold <johan@kernel.org>
Date: Fri, 21 May 2021 15:37:42 +0200
Subject: ALSA: usb-audio: fix control-request direction

The direction of the pipe argument must match the request-type direction
bit or control requests may fail depending on the host-controller-driver
implementation.

Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe().

Fixes: 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: stable@vger.kernel.org      # 5.10
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20210521133742.18098-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/format.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/format.c b/sound/usb/format.c
index e6ff317a6785..2287f8c65315 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -436,7 +436,7 @@ static bool check_valid_altsetting_v2v3(struct snd_usb_audio *chip, int iface,
 	if (snd_BUG_ON(altsetting >= 64 - 8))
 		return false;
 
-	err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
 			      UAC2_AS_VAL_ALT_SETTINGS << 8,
 			      iface, &raw_data, sizeof(raw_data));
-- 
cgit v1.2.3


From 764fa6e686e0107c0357a988d193de04cf047583 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Fri, 21 May 2021 17:50:12 +0930
Subject: ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci

Use usb_rcvctrlpipe() not usb_sndctrlpipe() for USB control input in
the Scarlett Gen 2 mixer driver. This fixes the device hang during
initialisation when used with the ehci-pci host driver.

Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/66a3d05dac325d5b53e4930578e143cef1f50dbe.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_scarlett_gen2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 560c2ade829d..dcff3e3a49f3 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -635,7 +635,7 @@ static int scarlett2_usb(
 	/* send a second message to get the response */
 
 	err = snd_usb_ctl_msg(mixer->chip->dev,
-			usb_sndctrlpipe(mixer->chip->dev, 0),
+			usb_rcvctrlpipe(mixer->chip->dev, 0),
 			SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP,
 			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			0,
-- 
cgit v1.2.3


From 265d1a90e4fb6d3264d8122fbd10760e5e733be6 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Fri, 21 May 2021 17:50:13 +0930
Subject: ALSA: usb-audio: scarlett2: Improve driver startup messages

Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.

Use info level instead of error for notifications.

Display the VID/PID so device_setup is targeted to the right device.

Display "enabled" message to easily confirm that the driver is loaded.

Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c        |  2 +-
 sound/usb/mixer_scarlett_gen2.c | 79 ++++++++++++++++++++++++++---------------
 sound/usb/mixer_scarlett_gen2.h |  2 +-
 3 files changed, 52 insertions(+), 31 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index fda66b2dbb01..37ad77524c0b 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -3060,7 +3060,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */
 	case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */
 	case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */
-		err = snd_scarlett_gen2_controls_create(mixer);
+		err = snd_scarlett_gen2_init(mixer);
 		break;
 
 	case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index dcff3e3a49f3..3ad8f61a2095 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -1997,38 +1997,11 @@ static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer)
 	return usb_submit_urb(mixer->urb, GFP_KERNEL);
 }
 
-/* Entry point */
-int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer)
+int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer,
+				      const struct scarlett2_device_info *info)
 {
-	const struct scarlett2_device_info *info;
 	int err;
 
-	/* only use UAC_VERSION_2 */
-	if (!mixer->protocol)
-		return 0;
-
-	switch (mixer->chip->usb_id) {
-	case USB_ID(0x1235, 0x8203):
-		info = &s6i6_gen2_info;
-		break;
-	case USB_ID(0x1235, 0x8204):
-		info = &s18i8_gen2_info;
-		break;
-	case USB_ID(0x1235, 0x8201):
-		info = &s18i20_gen2_info;
-		break;
-	default: /* device not (yet) supported */
-		return -EINVAL;
-	}
-
-	if (!(mixer->chip->setup & SCARLETT2_ENABLE)) {
-		usb_audio_err(mixer->chip,
-			"Focusrite Scarlett Gen 2 Mixer Driver disabled; "
-			"use options snd_usb_audio device_setup=1 "
-			"to enable and report any issues to g@b4.vu");
-		return 0;
-	}
-
 	/* Initialise private data, routing, sequence number */
 	err = scarlett2_init_private(mixer, info);
 	if (err < 0)
@@ -2073,3 +2046,51 @@ int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer)
 
 	return 0;
 }
+
+int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	const struct scarlett2_device_info *info;
+	int err;
+
+	/* only use UAC_VERSION_2 */
+	if (!mixer->protocol)
+		return 0;
+
+	switch (chip->usb_id) {
+	case USB_ID(0x1235, 0x8203):
+		info = &s6i6_gen2_info;
+		break;
+	case USB_ID(0x1235, 0x8204):
+		info = &s18i8_gen2_info;
+		break;
+	case USB_ID(0x1235, 0x8201):
+		info = &s18i20_gen2_info;
+		break;
+	default: /* device not (yet) supported */
+		return -EINVAL;
+	}
+
+	if (!(chip->setup & SCARLETT2_ENABLE)) {
+		usb_audio_info(chip,
+			"Focusrite Scarlett Gen 2 Mixer Driver disabled; "
+			"use options snd_usb_audio vid=0x%04x pid=0x%04x "
+			"device_setup=1 to enable and report any issues "
+			"to g@b4.vu",
+			USB_ID_VENDOR(chip->usb_id),
+			USB_ID_PRODUCT(chip->usb_id));
+		return 0;
+	}
+
+	usb_audio_info(chip,
+		"Focusrite Scarlett Gen 2 Mixer Driver enabled pid=0x%04x",
+		USB_ID_PRODUCT(chip->usb_id));
+
+	err = snd_scarlett_gen2_controls_create(mixer, info);
+	if (err < 0)
+		usb_audio_err(mixer->chip,
+			      "Error initialising Scarlett Mixer Driver: %d",
+			      err);
+
+	return err;
+}
diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h
index 52e1dad77afd..668c6b0cb50a 100644
--- a/sound/usb/mixer_scarlett_gen2.h
+++ b/sound/usb/mixer_scarlett_gen2.h
@@ -2,6 +2,6 @@
 #ifndef __USB_MIXER_SCARLETT_GEN2_H
 #define __USB_MIXER_SCARLETT_GEN2_H
 
-int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer);
+int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer);
 
 #endif /* __USB_MIXER_SCARLETT_GEN2_H */
-- 
cgit v1.2.3


From 119b75c150773425a89033215eab4d15d4198f8b Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Sat, 22 May 2021 11:47:41 +0800
Subject: ALSA: hda/realtek: Headphone volume is controlled by Front mixer

On some ASUS and MSI machines, the audio codec is alc1220 and the
Headphone is connected to audio mixer 0xf and DAC 0x5, in theory
the Headphone volume is controlled by DAC 0x5 (Heapdhone Playback
Volume), but somehow it is controlled by DAC 0x2 (Front Playback
Volume), maybe this is a defect on the codec alc1220.

Because of this issue, the PA couldn't switch the headphone and
Lineout correctly, If we apply the quirk CLEVO_P950 to those machines,
the Lineout and Headphone will share the audio mixer 0xc and DAC 0x2,
and generate Headphone+LO mixer, then PA could handle them when
switching between them.

BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1206
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210522034741.13415-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++
 1 file changed, 23 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 552e2cb73291..ffaeb8d3c316 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2603,6 +2603,28 @@ static const struct hda_model_fixup alc882_fixup_models[] = {
 	{}
 };
 
+static const struct snd_hda_pin_quirk alc882_pin_fixup_tbl[] = {
+	SND_HDA_PIN_QUIRK(0x10ec1220, 0x1043, "ASUS", ALC1220_FIXUP_CLEVO_P950,
+		{0x14, 0x01014010},
+		{0x15, 0x01011012},
+		{0x16, 0x01016011},
+		{0x18, 0x01a19040},
+		{0x19, 0x02a19050},
+		{0x1a, 0x0181304f},
+		{0x1b, 0x0221401f},
+		{0x1e, 0x01456130}),
+	SND_HDA_PIN_QUIRK(0x10ec1220, 0x1462, "MS-7C35", ALC1220_FIXUP_CLEVO_P950,
+		{0x14, 0x01015010},
+		{0x15, 0x01011012},
+		{0x16, 0x01011011},
+		{0x18, 0x01a11040},
+		{0x19, 0x02a19050},
+		{0x1a, 0x0181104f},
+		{0x1b, 0x0221401f},
+		{0x1e, 0x01451130}),
+	{}
+};
+
 /*
  * BIOS auto configuration
  */
@@ -2644,6 +2666,7 @@ static int patch_alc882(struct hda_codec *codec)
 
 	snd_hda_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
 		       alc882_fixups);
+	snd_hda_pick_pin_fixup(codec, alc882_pin_fixup_tbl, alc882_fixups, true);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
-- 
cgit v1.2.3


From 9ebaef0540a981093bce5df15af32354d32391d9 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Sat, 22 May 2021 12:26:45 +0800
Subject: ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i

The Lenovo Yoga 9i has bass speaker, but the bass speaker can't work,
that is because there is an i2s amplifier on that speaker, need to
run ideapad_s740_coef() to initialize the amplifier.

And also needs to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK to rename
the speaker's mixer control name, otherwise the PA can't handle them.

BugLink: http://bugs.launchpad.net/bugs/1926165
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210522042645.14221-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 +++++++++
 1 file changed, 9 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ffaeb8d3c316..6571c3713732 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6566,6 +6566,7 @@ enum {
 	ALC295_FIXUP_ASUS_DACS,
 	ALC295_FIXUP_HP_OMEN,
 	ALC285_FIXUP_HP_SPECTRE_X360,
+	ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8132,6 +8133,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1,
 	},
+	[ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_ideapad_s740_coef,
+		.chained = true,
+		.chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8500,6 +8507,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
 	SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
+	SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
@@ -8715,6 +8723,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"},
 	{.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"},
 	{.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
+	{.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From 2b899f31f1a6db2db4608bac2ac04fe2c4ad89eb Mon Sep 17 00:00:00 2001
From: kernel test robot <lkp@intel.com>
Date: Sun, 23 May 2021 02:09:00 +0800
Subject: ALSA: usb-audio: scarlett2: snd_scarlett_gen2_controls_create() can
 be static

sound/usb/mixer_scarlett_gen2.c:2000:5: warning: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static?

Fixes: 265d1a90e4fb ("ALSA: usb-audio: scarlett2: Improve driver startup messages")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20210522180900.GA83915@f59a3af2f1d9
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_scarlett_gen2.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 3ad8f61a2095..4caf379d5b99 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -1997,8 +1997,8 @@ static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer)
 	return usb_submit_urb(mixer->urb, GFP_KERNEL);
 }
 
-int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer,
-				      const struct scarlett2_device_info *info)
+static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer,
+					     const struct scarlett2_device_info *info)
 {
 	int err;
 
-- 
cgit v1.2.3


From 29c8f40b54a45dd23971e2bc395697731bcffbe1 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Mon, 24 May 2021 23:37:26 +0300
Subject: ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340

Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces
bug plock/pop noise over line out (green jack on the back) which can be
fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine.

Convert the existing entry for the same SSID to chain to apply this fixup
as well.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210524203726.2278-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 10 +++++++++-
 1 file changed, 9 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6571c3713732..90bf0d3a830a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6567,6 +6567,7 @@ enum {
 	ALC295_FIXUP_HP_OMEN,
 	ALC285_FIXUP_HP_SPECTRE_X360,
 	ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP,
+	ALC623_FIXUP_LENOVO_THINKSTATION_P340,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8139,6 +8140,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	},
+	[ALC623_FIXUP_LENOVO_THINKSTATION_P340] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_no_shutup,
+		.chained = true,
+		.chain_id = ALC283_FIXUP_HEADSET_MIC,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8457,7 +8464,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS),
-	SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -8724,6 +8731,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"},
 	{.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
 	{.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"},
+	{.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From 0e68c4b11f1e66d211ad242007e9f1076a6b7709 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Thu, 20 May 2021 01:03:53 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8

The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs
ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the
fixup, the mute/micmute LEDs work good.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-1-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 90bf0d3a830a..7f743382d395 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8328,6 +8328,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
-- 
cgit v1.2.3


From bbe183e07817a46cf8d3d7fc88093df81d23a957 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Thu, 20 May 2021 01:03:54 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook G8

The HP ZBook Studio 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210519170357.58410-2-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 7f743382d395..f33537099ae2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8328,6 +8328,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
-- 
cgit v1.2.3


From e650c1a959da49f2b873cb56564b825882c22e7a Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Thu, 20 May 2021 01:03:55 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook
 Fury 15 G8

The HP ZBook Fury 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210519170357.58410-3-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 f33537099ae2..784fdeb8dfea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8328,6 +8328,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
-- 
cgit v1.2.3


From 50dbfae972cbe0e3c631e73c7c58cbc48bfc6a49 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Thu, 20 May 2021 01:03:56 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook
 Fury 17 G8

The HP ZBook Studio 17.3 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210519170357.58410-4-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 784fdeb8dfea..61a60c420f6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8328,6 +8328,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
-- 
cgit v1.2.3


From 6308c44ed6eeadf65c0a7ba68d609773ed860fbb Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Thu, 27 May 2021 01:06:51 +0000
Subject: ASoC: rt5659: Fix the lost powers for the HDA header

The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after
the DAPM widgets were added, and these powers were set by the JD settings
"RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function,
these powers were ignored to prevent them controlled by DAPM.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Message-Id: <15fced51977b458798ca4eebf03dafb9@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++-----
 1 file changed, 21 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 87f5709fe2cc..4a50b169fe03 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0,
 		NULL, 0),
-	SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
+		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL,
 		RT5659_PWR_MIC_DET_BIT, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
+		NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1,
 		RT5659_PWR_VREF3_BIT, 0, NULL, 0),
 
@@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
 		RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0),
 
 	/* Input Side */
-	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
-		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT,
 		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT,
@@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component,
 
 static int rt5659_probe(struct snd_soc_component *component)
 {
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
 	struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
 
 	rt5659->component = component;
 
+	switch (rt5659->pdata.jd_src) {
+	case RT5659_JD_HDA_HEADER:
+		break;
+
+	default:
+		snd_soc_dapm_new_controls(dapm,
+			rt5659_particular_dapm_widgets,
+			ARRAY_SIZE(rt5659_particular_dapm_widgets));
+		break;
+	}
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From 4ad7935df6a566225c3d51900bde8f2f0f8b6de3 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Fri, 28 May 2021 21:51:23 +0300
Subject: ALSA: hda: Add AlderLake-M PCI ID
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Add HD Audio PCI ID for Intel AlderLake-M. Add rules to
snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with
PCH-DMIC or Soundwire codecs, and legacy driver for the rest.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/intel-dsp-config.c | 4 ++++
 sound/pci/hda/hda_intel.c    | 3 +++
 2 files changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index ab5ff7867eb9..d8be146793ee 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -331,6 +331,10 @@ static const struct config_entry config_table[] = {
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x51c8,
 	},
+	{
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
+		.device = 0x51cc,
+	},
 #endif
 
 };
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 79ade335c8a0..470753b36c8a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Alderlake-P */
 	{ PCI_DEVICE(0x8086, 0x51c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Alderlake-M */
+	{ PCI_DEVICE(0x8086, 0x51cc),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
-- 
cgit v1.2.3


From 08a4b904a2a90246aadd6aa2e4f26abca9037385 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 31 May 2021 20:06:33 +0200
Subject: ALSA: hda: Fix a regression in Capture Switch mixer read

The recent commit to drop the HDA-specific mute-LED control,
e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the
audio LED trigger"), caused a regression on the mixer element read for
"Capture Switch" when it's built from bind controls.  The function
create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly
via snd_hda_gen_add_kctl() instead of add_control().  Although the
commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in
add_control() as default, this code path fell out from the radar.  As
a result, now the driver gives -EPERM error because of the lack of the
proper access bit at reading "Capture Switch" element value.

Fix the regression by setting the access bit properly.

Fixes: e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634
Link: https://lore.kernel.org/r/20210531180633.27831-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b638fc2ef6f7..1f8018f9ce57 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3520,6 +3520,7 @@ static int cap_sw_put(struct snd_kcontrol *kcontrol,
 static const struct snd_kcontrol_new cap_sw_temp = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Switch",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.info = cap_sw_info,
 	.get = cap_sw_get,
 	.put = cap_sw_put,
-- 
cgit v1.2.3


From 527ff9550682a3d08066a000435ffd8330bdd729 Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Mon, 31 May 2021 17:37:54 +0100
Subject: ALSA: hda/cirrus: Set Initial DMIC volume to -26 dB

Previously this fix was applied only to Bullseye variant laptops,
and should be applied to Cyborg and Warlock variants.

Fixes: 45b14fe200ba ("ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210531163754.136736-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 726507d0b04c..8629e84fef23 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -2206,10 +2206,9 @@ static void cs8409_cs42l42_fixups(struct hda_codec *codec,
 		break;
 	case HDA_FIXUP_ACT_PROBE:
 
-		/* Set initial volume on Bullseye to -26 dB */
-		if (codec->fixup_id == CS8409_BULLSEYE)
-			snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID,
-					HDA_INPUT, 0, 0xff, 0x19);
+		/* Set initial DMIC volume to -26 dB */
+		snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID,
+				HDA_INPUT, 0, 0xff, 0x19);
 		snd_hda_gen_add_kctl(&spec->gen,
 			NULL, &cs8409_cs42l42_hp_volume_mixer);
 		snd_hda_gen_add_kctl(&spec->gen,
-- 
cgit v1.2.3


From 901be145a46eb79879367d853194346a549e623d Mon Sep 17 00:00:00 2001
From: Carlos M <carlos.marr.pz@gmail.com>
Date: Mon, 31 May 2021 22:20:26 +0200
Subject: ALSA: hda: Fix for mute key LED for HP Pavilion 15-CK0xx

For the HP Pavilion 15-CK0xx, with audio subsystem ID 0x103c:0x841c,
adding a line in patch_realtek.c to apply the ALC269_FIXUP_HP_MUTE_LED_MIC3
fix activates the mute key LED.

Signed-off-by: Carlos M <carlos.marr.pz@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210531202026.35427-1-carlos.marr.pz@gmail.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 61a60c420f6f..43e37145eb5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8303,6 +8303,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
-- 
cgit v1.2.3


From ce1f25718b2520d0210c24f1e4145d75c5620c9f Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Tue, 1 Jun 2021 11:35:06 +0100
Subject: ASoC: topology: Fix spelling mistake "vesion" -> "version"

There are spelling mistakes in comments. Fix them.

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210601103506.9477-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-topology.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 73076d425efb..4893a56208e0 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1901,7 +1901,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest,
  * @src: older version of pcm as a source
  * @pcm: latest version of pcm created from the source
  *
- * Support from vesion 4. User should free the returned pcm manually.
+ * Support from version 4. User should free the returned pcm manually.
  */
 static int pcm_new_ver(struct soc_tplg *tplg,
 		       struct snd_soc_tplg_pcm *src,
@@ -2089,7 +2089,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link,
  * @src: old version of phyical link config as a source
  * @link: latest version of physical link config created from the source
  *
- * Support from vesion 4. User need free the returned link config manually.
+ * Support from version 4. User need free the returned link config manually.
  */
 static int link_new_ver(struct soc_tplg *tplg,
 			struct snd_soc_tplg_link_config *src,
@@ -2400,7 +2400,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
  * @src: old version of manifest as a source
  * @manifest: latest version of manifest created from the source
  *
- * Support from vesion 4. Users need free the returned manifest manually.
+ * Support from version 4. Users need free the returned manifest manually.
  */
 static int manifest_new_ver(struct soc_tplg *tplg,
 			    struct snd_soc_tplg_manifest *src,
-- 
cgit v1.2.3


From a8437f05384cb472518ec21bf4fffbe8f0a47378 Mon Sep 17 00:00:00 2001
From: Nicolas Cavallari <nicolas.cavallari@green-communications.fr>
Date: Thu, 27 May 2021 18:34:09 +0200
Subject: ASoC: fsl-asoc-card: Set .owner attribute when registering card.

Otherwise, when compiled as module, a WARN_ON is triggered:

WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd]
[...]
CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1
Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree)
Workqueue: events deferred_probe_work_func
[<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14)
[<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104)
[<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114)
[<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4)
[<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd])
[<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core])
[<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core])
[<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card])
[<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98)
[...]

Signed-off-by: Nicolas Cavallari <nicolas.cavallari@green-communications.fr>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-communications.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl-asoc-card.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index c62bfd1c3ac7..4f55b316cf0f 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	/* Initialize sound card */
 	priv->pdev = pdev;
 	priv->card.dev = &pdev->dev;
+	priv->card.owner = THIS_MODULE;
 	ret = snd_soc_of_parse_card_name(&priv->card, "model");
 	if (ret) {
 		snprintf(priv->name, sizeof(priv->name), "%s-audio",
-- 
cgit v1.2.3


From b640e8a4bd24e17ce24a064d704aba14831651a8 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Fri, 28 May 2021 17:43:30 +0300
Subject: ASoC: SOF: reset enabled_cores state at suspend

The recent changes to use common code to power up/down DSP cores also
removed the reset of the core state at suspend. It turns out this is
still needed. When the firmware state is reset to
SOF_FW_BOOT_NOT_STARTED, also enabled_cores should be reset, and
existing DSP drivers depend on this.

BugLink: https://github.com/thesofproject/linux/issues/2824
Fixes: 42077f08b3 ("ASoC: SOF: update dsp core power status in common APIs")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20210528144330.2551-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/pm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index fd265803f7bc..c83fb6255961 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -256,6 +256,7 @@ suspend:
 
 	/* reset FW state */
 	sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+	sdev->enabled_cores_mask = 0;
 
 	return ret;
 }
-- 
cgit v1.2.3


From 3ae72f6ab9c1f688bd578cdc252dabce65fdaf57 Mon Sep 17 00:00:00 2001
From: Dongliang Mu <mudongliangabcd@gmail.com>
Date: Wed, 2 Jun 2021 11:41:36 +0800
Subject: ALSA: control led: fix memory leak in snd_ctl_led_register

The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain
the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails
to decrease the refcount to zero, which causes device_release never to
be invoked. This leads to memory leak to some resources, like struct
device_private. In addition, we also free some other similar memory
leaks in snd_ctl_led_init/snd_ctl_led_exit.

Fix this by replacing device_del to device_unregister
in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit.

Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will
call kobject_release and delay the release of kobject, which will cause
use-after-free when the memory backing the kobject is freed at once.

Reported-by: syzbot+08a7d8b51ea048a74ffb@syzkaller.appspotmail.com
Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer")
Signed-off-by: Dongliang Mu <mudongliangabcd@gmail.com>
Reviewed-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210602034136.2762497-1-mudongliangabcd@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/control_led.c | 33 ++++++++++++++++++++++++++-------
 1 file changed, 26 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/core/control_led.c b/sound/core/control_led.c
index 25f57c14f294..a90e31dbde61 100644
--- a/sound/core/control_led.c
+++ b/sound/core/control_led.c
@@ -17,6 +17,9 @@ MODULE_LICENSE("GPL");
 #define MAX_LED (((SNDRV_CTL_ELEM_ACCESS_MIC_LED - SNDRV_CTL_ELEM_ACCESS_SPK_LED) \
 			>> SNDRV_CTL_ELEM_ACCESS_LED_SHIFT) + 1)
 
+#define to_led_card_dev(_dev) \
+	container_of(_dev, struct snd_ctl_led_card, dev)
+
 enum snd_ctl_led_mode {
 	 MODE_FOLLOW_MUTE = 0,
 	 MODE_FOLLOW_ROUTE,
@@ -371,6 +374,21 @@ static void snd_ctl_led_disconnect(struct snd_card *card)
 	snd_ctl_led_refresh();
 }
 
+static void snd_ctl_led_card_release(struct device *dev)
+{
+	struct snd_ctl_led_card *led_card = to_led_card_dev(dev);
+
+	kfree(led_card);
+}
+
+static void snd_ctl_led_release(struct device *dev)
+{
+}
+
+static void snd_ctl_led_dev_release(struct device *dev)
+{
+}
+
 /*
  * sysfs
  */
@@ -663,6 +681,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card)
 		led_card->number = card->number;
 		led_card->led = led;
 		device_initialize(&led_card->dev);
+		led_card->dev.release = snd_ctl_led_card_release;
 		if (dev_set_name(&led_card->dev, "card%d", card->number) < 0)
 			goto cerr;
 		led_card->dev.parent = &led->dev;
@@ -681,7 +700,6 @@ cerr:
 		put_device(&led_card->dev);
 cerr2:
 		printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number);
-		kfree(led_card);
 	}
 }
 
@@ -700,8 +718,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card)
 		snprintf(link_name, sizeof(link_name), "led-%s", led->name);
 		sysfs_remove_link(&card->ctl_dev.kobj, link_name);
 		sysfs_remove_link(&led_card->dev.kobj, "card");
-		device_del(&led_card->dev);
-		kfree(led_card);
+		device_unregister(&led_card->dev);
 		led->cards[card->number] = NULL;
 	}
 }
@@ -723,6 +740,7 @@ static int __init snd_ctl_led_init(void)
 
 	device_initialize(&snd_ctl_led_dev);
 	snd_ctl_led_dev.class = sound_class;
+	snd_ctl_led_dev.release = snd_ctl_led_dev_release;
 	dev_set_name(&snd_ctl_led_dev, "ctl-led");
 	if (device_add(&snd_ctl_led_dev)) {
 		put_device(&snd_ctl_led_dev);
@@ -733,15 +751,16 @@ static int __init snd_ctl_led_init(void)
 		INIT_LIST_HEAD(&led->controls);
 		device_initialize(&led->dev);
 		led->dev.parent = &snd_ctl_led_dev;
+		led->dev.release = snd_ctl_led_release;
 		led->dev.groups = snd_ctl_led_dev_attr_groups;
 		dev_set_name(&led->dev, led->name);
 		if (device_add(&led->dev)) {
 			put_device(&led->dev);
 			for (; group > 0; group--) {
 				led = &snd_ctl_leds[group - 1];
-				device_del(&led->dev);
+				device_unregister(&led->dev);
 			}
-			device_del(&snd_ctl_led_dev);
+			device_unregister(&snd_ctl_led_dev);
 			return -ENOMEM;
 		}
 	}
@@ -767,9 +786,9 @@ static void __exit snd_ctl_led_exit(void)
 	}
 	for (group = 0; group < MAX_LED; group++) {
 		led = &snd_ctl_leds[group];
-		device_del(&led->dev);
+		device_unregister(&led->dev);
 	}
-	device_del(&snd_ctl_led_dev);
+	device_unregister(&snd_ctl_led_dev);
 	snd_ctl_led_clean(NULL);
 }
 
-- 
cgit v1.2.3


From 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d Mon Sep 17 00:00:00 2001
From: Mark Pearson <markpearson@lenovo.com>
Date: Mon, 31 May 2021 10:55:02 -0400
Subject: ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms

More laptops identified where the AMD ACP bridge needs to be blocked
or the microphone will not work when connected to HDMI.

Use DMI to block the microphone PCM device for these platforms.

Suggested-by: Gabriel Craciunescu <nix.or.die@gmail.com>
Signed-off-by: Mark Pearson <markpearson@lenovo.com>
Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++++++++
 1 file changed, 35 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c
index 19438da5dfa5..c9fb1c8fbf8c 100644
--- a/sound/soc/amd/renoir/rn-pci-acp3x.c
+++ b/sound/soc/amd/renoir/rn-pci-acp3x.c
@@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = {
 			DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"),
 		}
 	},
+	{
+		/* Lenovo ThinkPad P14s Gen 1 (20Y1) */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_BOARD_NAME, "20Y1"),
+		}
+	},
+	{
+		/* Lenovo ThinkPad T14s Gen1 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_BOARD_NAME, "20UH"),
+		}
+	},
+	{
+		/* Lenovo ThinkPad T14s Gen1 Campus */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_BOARD_NAME, "20UJ"),
+		}
+	},
+	{
+		/* Lenovo ThinkPad T14 Gen 1*/
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_BOARD_NAME, "20UD"),
+		}
+	},
+	{
+		/* Lenovo ThinkPad X13 Gen 1*/
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_BOARD_NAME, "20UF"),
+		}
+	},
 	{}
 };
 
-- 
cgit v1.2.3


From 9c1fe96bded935369f8340c2ac2e9e189f697d5d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Jun 2021 13:38:23 +0200
Subject: ALSA: timer: Fix master timer notification

snd_timer_notify1() calls the notification to each slave for a master
event, but it passes a wrong event number.  It should be +10 offset,
corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with
+100 offset.  Casually this was spotted by UBSAN check via syzkaller.

Reported-by: syzbot+d102fa5b35335a7e544e@syzkaller.appspotmail.com
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com
Link: https://lore.kernel.org/r/20210602113823.23777-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/timer.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/core/timer.c b/sound/core/timer.c
index 6898b1ac0d7f..92b7008fcdb8 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event)
 		return;
 	if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
 		return;
+	event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */
 	list_for_each_entry(ts, &ti->slave_active_head, active_list)
 		if (ts->ccallback)
-			ts->ccallback(ts, event + 100, &tstamp, resolution);
+			ts->ccallback(ts, event, &tstamp, resolution);
 }
 
 /* start/continue a master timer */
-- 
cgit v1.2.3


From b8b90c17602689eeaa5b219d104bbc215d1225cc Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 2 Jun 2021 22:54:24 +0800
Subject: ALSA: hda: update the power_state during the direct-complete

The patch_realtek.c needs to check if the power_state.event equals
PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and
resume() will be skipped if the codec is already rt_suspended, in this
case, the patch_realtek.c will always get PM_EVENT_ON even the system
is really resumed from S3.

We could set power_state to PMSG_SUSPEND in the prepare(), if other
PM functions are called before complete(), those functions will
override power_state; if no other PM functions are called before
complete(), we could know the suspend() and resume() are skipped since
only S3 pm functions could be skipped by direct-complete, in this case
set power_state to PMSG_RESUME in the complete(). This could guarantee
the first time of calling hda_codec_runtime_resume() after complete()
has the correct power_state.

Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210602145424.3132-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a31009afc025..5462f771c2f9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2917,6 +2917,7 @@ static int hda_codec_runtime_resume(struct device *dev)
 #ifdef CONFIG_PM_SLEEP
 static int hda_codec_pm_prepare(struct device *dev)
 {
+	dev->power.power_state = PMSG_SUSPEND;
 	return pm_runtime_suspended(dev);
 }
 
@@ -2924,6 +2925,10 @@ static void hda_codec_pm_complete(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	/* If no other pm-functions are called between prepare() and complete() */
+	if (dev->power.power_state.event == PM_EVENT_SUSPEND)
+		dev->power.power_state = PMSG_RESUME;
+
 	if (pm_runtime_suspended(dev) && (codec->jackpoll_interval ||
 	    hda_codec_need_resume(codec) || codec->forced_resume))
 		pm_request_resume(dev);
-- 
cgit v1.2.3


From 320232caf1d8febea17312dab4b2dfe02e033520 Mon Sep 17 00:00:00 2001
From: Mark Pearson <markpearson@lenovo.com>
Date: Wed, 2 Jun 2021 13:12:51 -0400
Subject: ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms

Unfortunately the previous patch to fix issues using the AMD ACP bridge
has the side effect of breaking the dmic in other cases and needs to be
reverted.

Removing the changes while we revisit the fix and find something better.
Apologies for the churn.

Suggested-by: Gabriel Craciunescu <unix.or.die@gmail.com>
Signed-off-by: Mark Pearson <markpearson@lenovo.com>
Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/renoir/rn-pci-acp3x.c | 35 -----------------------------------
 1 file changed, 35 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c
index c9fb1c8fbf8c..19438da5dfa5 100644
--- a/sound/soc/amd/renoir/rn-pci-acp3x.c
+++ b/sound/soc/amd/renoir/rn-pci-acp3x.c
@@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = {
 			DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"),
 		}
 	},
-	{
-		/* Lenovo ThinkPad P14s Gen 1 (20Y1) */
-		.matches = {
-			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
-			DMI_MATCH(DMI_BOARD_NAME, "20Y1"),
-		}
-	},
-	{
-		/* Lenovo ThinkPad T14s Gen1 */
-		.matches = {
-			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
-			DMI_MATCH(DMI_BOARD_NAME, "20UH"),
-		}
-	},
-	{
-		/* Lenovo ThinkPad T14s Gen1 Campus */
-		.matches = {
-			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
-			DMI_MATCH(DMI_BOARD_NAME, "20UJ"),
-		}
-	},
-	{
-		/* Lenovo ThinkPad T14 Gen 1*/
-		.matches = {
-			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
-			DMI_MATCH(DMI_BOARD_NAME, "20UD"),
-		}
-	},
-	{
-		/* Lenovo ThinkPad X13 Gen 1*/
-		.matches = {
-			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
-			DMI_MATCH(DMI_BOARD_NAME, "20UF"),
-		}
-	},
 	{}
 };
 
-- 
cgit v1.2.3


From 8bef925e37bdc9b6554b85eda16ced9a8e3c135f Mon Sep 17 00:00:00 2001
From: Richard Weinberger <richard@nod.at>
Date: Sun, 30 May 2021 22:34:46 +0200
Subject: ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values

TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0.
So all values need to be left shifted by 1.

Signed-off-by: Richard Weinberger <richard@nod.at>
Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2562.h | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h
index 81866aeb3fbf..55b2a1f52ca3 100644
--- a/sound/soc/codecs/tas2562.h
+++ b/sound/soc/codecs/tas2562.h
@@ -57,13 +57,13 @@
 #define TAS2562_TDM_CFG0_RAMPRATE_MASK		BIT(5)
 #define TAS2562_TDM_CFG0_RAMPRATE_44_1		BIT(5)
 #define TAS2562_TDM_CFG0_SAMPRATE_MASK		GENMASK(3, 1)
-#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ	0x0
-#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ	0x1
-#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ	0x2
-#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ	0x3
-#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ	0x4
-#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ	0x5
-#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ	0x6
+#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ	(0x0 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ	(0x1 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ	(0x2 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ	(0x3 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ	(0x4 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ	(0x5 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ	(0x6 << 1)
 
 #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY	BIT(6)
 
-- 
cgit v1.2.3


From 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Fri, 4 Jun 2021 14:31:50 +0800
Subject: ASoC: rt5682: Fix the fast discharge for headset unplugging in
 soundwire mode

Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also
should be set in soundwire mode.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5682-sdw.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index fed80c8f994f..e78ba3b064c4 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
 
 	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
 		RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
-	regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042);
+	regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142);
+	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600);
 	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3,
 		RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
 	regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1,
-- 
cgit v1.2.3


From 15d295b560e6dd45f839a53ae69e4f63b54eb32f Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Sat, 5 Jun 2021 16:25:36 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite
 Dragonfly G2

The HP Elite Dragonfly G2 using ALC285 codec which using 0x04 to control
mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210605082539.41797-1-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 43e37145eb5d..9f65171a902d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8310,6 +8310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
+	SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
-- 
cgit v1.2.3


From 61d3e87468fad82dc8e8cb6de7db563ada64b532 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Sat, 5 Jun 2021 16:25:37 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP EliteBook
 x360 1040 G8

The HP EliteBook x360 1040 G8 using ALC285 codec which using 0x04 to control
mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210605082539.41797-2-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 9f65171a902d..11324163ebe1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8311,6 +8311,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+	SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
-- 
cgit v1.2.3


From dfb06401b4cdfc71e2fc3e19b877ab845cc9f7f7 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Sat, 5 Jun 2021 16:25:38 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 840 Aero G8

The HP EliteBook 840 Aero G8 using ALC285 codec which using 0x04 to
control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
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/20210605082539.41797-3-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 11324163ebe1..215beb3ac678 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8330,6 +8330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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, 0x884b, "HP EliteBook 840 Aero 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(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
-- 
cgit v1.2.3


From 9981b20a5e3694f4625ab5a1ddc98ce7503f6d12 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sat, 5 Jun 2021 18:10:54 +0900
Subject: ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()

In the workqueue to queue wake-up event, isochronous context is not
processed, thus it's useless to check context for the workqueue to switch
status of runtime for PCM substream to XRUN. On the other hand, in
software IRQ context of 1394 OHCI, it's needed.

This commit fixes the bug introduced when tasklet was replaced with
workqueue.

Cc: <stable@vger.kernel.org>
Fixes: 2b3d2987d800 ("ALSA: firewire: Replace tasklet with work")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210605091054.68866-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index e0faa6601966..5805c5de39fb 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -804,7 +804,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs,
 static inline void cancel_stream(struct amdtp_stream *s)
 {
 	s->packet_index = -1;
-	if (current_work() == &s->period_work)
+	if (in_interrupt())
 		amdtp_stream_pcm_abort(s);
 	WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN);
 }
-- 
cgit v1.2.3


From c8a4556d98510ca05bad8d02265a4918b03a8c0b Mon Sep 17 00:00:00 2001
From: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Date: Fri, 4 Jun 2021 23:45:45 +0800
Subject: ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin

This patch fixes PoP noise of around 15ms observed during audio
capture begin.
Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for
introducing some delay before capture start.

(am from https://patchwork.kernel.org/patch/12276369/)
(also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org)

Co-developed-by: Judy Hsiao <judyhsiao@chromium.org>
Signed-off-by: Judy Hsiao <judyhsiao@chromium.org>
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/qcom/lpass.h     |  4 +++
 2 files changed, 83 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 28c7497344e3..a6e95db6b3fb 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+	struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+	unsigned int id = dai->driver->id;
 
 	clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+	/*
+	 * Ensure LRCLK is disabled even in device node validation.
+	 * Will not impact if disabled in lpass_cpu_daiops_trigger()
+	 * suspend.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE);
+	else
+		regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE);
+
+	/*
+	 * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before
+	 * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in
+	 * lpass_cpu_daiops_prepare.
+	 */
+	if (drvdata->mi2s_was_prepared[dai->driver->id]) {
+		drvdata->mi2s_was_prepared[dai->driver->id] = false;
+		clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
+	}
+
 	clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
 }
 
@@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		/*
+		 * Ensure lpass BCLK/LRCLK is enabled during
+		 * device resume as lpass_cpu_daiops_prepare() is not called
+		 * after the device resumes. We don't check mi2s_was_prepared before
+		 * enable/disable BCLK in trigger events because:
+		 *  1. These trigger events are paired, so the BCLK
+		 *     enable_count is balanced.
+		 *  2. the BCLK can be shared (ex: headset and headset mic),
+		 *     we need to increase the enable_count so that we don't
+		 *     turn off the shared BCLK while other devices are using
+		 *     it.
+		 */
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			ret = regmap_fields_write(i2sctl->spken, id,
 						 LPAIF_I2SCTL_SPKEN_ENABLE);
@@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		/*
+		 * To ensure lpass BCLK/LRCLK is disabled during
+		 * device suspend.
+		 */
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			ret = regmap_fields_write(i2sctl->spken, id,
 						 LPAIF_I2SCTL_SPKEN_DISABLE);
@@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+	struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+	unsigned int id = dai->driver->id;
+	int ret;
+
+	/*
+	 * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture
+	 * data flow starts. This allows other codec to have some delay before
+	 * the data flow.
+	 * (ex: to drop start up pop noise before capture starts).
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE);
+	else
+		ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE);
+
+	if (ret) {
+		dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can
+	 * be called multiple times. It's paired with the clk_disable in
+	 * lpass_cpu_daiops_shutdown.
+	 */
+	if (!drvdata->mi2s_was_prepared[dai->driver->id]) {
+		ret = clk_enable(drvdata->mi2s_bit_clk[id]);
+		if (ret) {
+			dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+			return ret;
+		}
+		drvdata->mi2s_was_prepared[dai->driver->id] = true;
+	}
+	return 0;
+}
+
 const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = {
 	.set_sysclk	= lpass_cpu_daiops_set_sysclk,
 	.startup	= lpass_cpu_daiops_startup,
 	.shutdown	= lpass_cpu_daiops_shutdown,
 	.hw_params	= lpass_cpu_daiops_hw_params,
 	.trigger	= lpass_cpu_daiops_trigger,
+	.prepare	= lpass_cpu_daiops_prepare,
 };
 EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops);
 
diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h
index 83b2e08ade06..7f72214404ba 100644
--- a/sound/soc/qcom/lpass.h
+++ b/sound/soc/qcom/lpass.h
@@ -67,6 +67,10 @@ struct lpass_data {
 	/* MI2S SD lines to use for playback/capture */
 	unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS];
 	unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS];
+
+	/* The state of MI2S prepare dai_ops was called */
+	bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS];
+
 	int hdmi_port_enable;
 
 	/* low-power audio interface (LPAIF) registers */
-- 
cgit v1.2.3


From 57c9e21a49b1c196cda28f54de9a5d556ac93f20 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Tue, 8 Jun 2021 10:46:00 +0800
Subject: ALSA: hda/realtek: headphone and mic don't work on an Acer laptop

There are 2 issues on this machine, the 1st one is mic's plug/unplug
can't be detected, that is because the mic is set to manual detecting
mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto
detecting mode. The other one is headphone's plug/unplug can't be
detected by pulseaudio, that is because the pulseaudio will use
ucm2/sof-hda-dsp on this machine, and the ucm2 only handle
'Headphone Jack', but on this machine the headphone's pincfg sets the
location to Front, then the alsa mixer name is "Front Headphone Jack"
instead of "Headphone Jack", so override the pincfg to change location
to Left.

BugLink: http://bugs.launchpad.net/bugs/1930188
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210608024600.6198-1-hui.wang@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 215beb3ac678..11ba8e351ad4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6568,6 +6568,7 @@ enum {
 	ALC285_FIXUP_HP_SPECTRE_X360,
 	ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP,
 	ALC623_FIXUP_LENOVO_THINKSTATION_P340,
+	ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC283_FIXUP_HEADSET_MIC,
 	},
+	[ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x21, 0x03211030 }, /* Change the Headphone location to Left */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -8740,6 +8751,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
 	{.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"},
 	{.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"},
+	{.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From 600dd2a7e8b62170d177381cc1303861f48f9780 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Tue, 8 Jun 2021 19:47:48 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8

The HP ZBook Power G8 using ALC236 codec which 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/20210608114750.32009-1-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 11ba8e351ad4..ab5113cccffa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8346,6 +8346,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+	SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
-- 
cgit v1.2.3


From 83e197a8414c0ba545e7e3916ce05f836f349273 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 10 Jun 2021 17:20:59 +0200
Subject: ALSA: seq: Fix race of snd_seq_timer_open()

The timer instance per queue is exclusive, and snd_seq_timer_open()
should have managed the concurrent accesses.  It looks as if it's
checking the already existing timer instance at the beginning, but
it's not right, because there is no protection, hence any later
concurrent call of snd_seq_timer_open() may override the timer
instance easily.  This may result in UAF, as the leftover timer
instance can keep running while the queue itself gets closed, as
spotted by syzkaller recently.

For avoiding the race, add a proper check at the assignment of
tmr->timeri again, and return -EBUSY if it's been already registered.

Reported-by: syzbot+ddc1260a83ed1cbf6fb5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com
Link: https://lore.kernel.org/r/20210610152059.24633-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/seq/seq_timer.c | 10 +++++++++-
 1 file changed, 9 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 1645e4142e30..9863be6fd43e 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
 		return err;
 	}
 	spin_lock_irq(&tmr->lock);
-	tmr->timeri = t;
+	if (tmr->timeri)
+		err = -EBUSY;
+	else
+		tmr->timeri = t;
 	spin_unlock_irq(&tmr->lock);
+	if (err < 0) {
+		snd_timer_close(t);
+		snd_timer_instance_free(t);
+		return err;
+	}
 	return 0;
 }
 
-- 
cgit v1.2.3


From 688d47cdd9344b1485eb28c2a7aa99743ed529a3 Mon Sep 17 00:00:00 2001
From: Claudius Heine <ch@denx.de>
Date: Thu, 17 Jun 2021 10:52:28 +0200
Subject: ASoC: tlv320aic32x4: add type to device private data struct

While this driver can already handle different device variants, the
variant information cannot be used in the driver code and therefor
cannot have different code paths depending on the device variant.

This change adds a `type` value into the `aic32x4_priv` structure, that
contains a device variant identifier, which was set when the driver was
bound to the device.

Signed-off-by: Claudius Heine <ch@denx.de>
Link: https://lore.kernel.org/r/20210617085230.1851503-2-ch@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic32x4-i2c.c | 20 ++++++++++++++++----
 sound/soc/codecs/tlv320aic32x4-spi.c | 23 +++++++++++++++++++----
 sound/soc/codecs/tlv320aic32x4.c     |  3 +++
 sound/soc/codecs/tlv320aic32x4.h     |  5 +++++
 4 files changed, 43 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
index 6d54cbf70a0b..247fb1e13674 100644
--- a/sound/soc/codecs/tlv320aic32x4-i2c.c
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -16,6 +16,8 @@
 
 #include "tlv320aic32x4.h"
 
+static const struct of_device_id aic32x4_of_id[];
+
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -27,6 +29,16 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 	config.val_bits = 8;
 
 	regmap = devm_regmap_init_i2c(i2c, &config);
+
+	if (i2c->dev.of_node) {
+		const struct of_device_id *oid;
+
+		oid = of_match_node(aic32x4_of_id, i2c->dev.of_node);
+		dev_set_drvdata(&i2c->dev, (void *)oid->data);
+	} else if (id) {
+		dev_set_drvdata(&i2c->dev, (void *)id->driver_data);
+	}
+
 	return aic32x4_probe(&i2c->dev, regmap);
 }
 
@@ -36,15 +48,15 @@ static int aic32x4_i2c_remove(struct i2c_client *i2c)
 }
 
 static const struct i2c_device_id aic32x4_i2c_id[] = {
-	{ "tlv320aic32x4", 0 },
-	{ "tlv320aic32x6", 1 },
+	{ "tlv320aic32x4", (kernel_ulong_t)AIC32X4_TYPE_AIC32X4 },
+	{ "tlv320aic32x6", (kernel_ulong_t)AIC32X4_TYPE_AIC32X6 },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
 
 static const struct of_device_id aic32x4_of_id[] = {
-	{ .compatible = "ti,tlv320aic32x4", },
-	{ .compatible = "ti,tlv320aic32x6", },
+	{ .compatible = "ti,tlv320aic32x4", .data = (void *)AIC32X4_TYPE_AIC32X4 },
+	{ .compatible = "ti,tlv320aic32x6", .data = (void *)AIC32X4_TYPE_AIC32X6 },
 	{ /* senitel */ }
 };
 MODULE_DEVICE_TABLE(of, aic32x4_of_id);
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
index a22e7700bfc8..e81c72958a82 100644
--- a/sound/soc/codecs/tlv320aic32x4-spi.c
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -16,6 +16,8 @@
 
 #include "tlv320aic32x4.h"
 
+static const struct of_device_id aic32x4_of_id[];
+
 static int aic32x4_spi_probe(struct spi_device *spi)
 {
 	struct regmap *regmap;
@@ -28,6 +30,19 @@ static int aic32x4_spi_probe(struct spi_device *spi)
 	config.read_flag_mask = 0x01;
 
 	regmap = devm_regmap_init_spi(spi, &config);
+
+	if (spi->dev.of_node) {
+		const struct of_device_id *oid;
+
+		oid = of_match_node(aic32x4_of_id, spi->dev.of_node);
+		dev_set_drvdata(&spi->dev, (void *)oid->data);
+	} else {
+		const struct spi_device_id *id_entry;
+
+		id_entry = spi_get_device_id(spi);
+		dev_set_drvdata(&spi->dev, (void *)id_entry->driver_data);
+	}
+
 	return aic32x4_probe(&spi->dev, regmap);
 }
 
@@ -37,15 +52,15 @@ static int aic32x4_spi_remove(struct spi_device *spi)
 }
 
 static const struct spi_device_id aic32x4_spi_id[] = {
-	{ "tlv320aic32x4", 0 },
-	{ "tlv320aic32x6", 1 },
+	{ "tlv320aic32x4", (kernel_ulong_t)AIC32X4_TYPE_AIC32X4 },
+	{ "tlv320aic32x6", (kernel_ulong_t)AIC32X4_TYPE_AIC32X6 },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(spi, aic32x4_spi_id);
 
 static const struct of_device_id aic32x4_of_id[] = {
-	{ .compatible = "ti,tlv320aic32x4", },
-	{ .compatible = "ti,tlv320aic32x6", },
+	{ .compatible = "ti,tlv320aic32x4", .data = (void *)AIC32X4_TYPE_AIC32X4 },
+	{ .compatible = "ti,tlv320aic32x6", .data = (void *)AIC32X4_TYPE_AIC32X6 },
 	{ /* senitel */ }
 };
 MODULE_DEVICE_TABLE(of, aic32x4_of_id);
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index b689f26fc4be..70a1574fb72a 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -48,6 +48,7 @@ struct aic32x4_priv {
 
 	struct aic32x4_setup_data *setup;
 	struct device *dev;
+	enum aic32x4_type type;
 };
 
 static int aic32x4_reset_adc(struct snd_soc_dapm_widget *w,
@@ -1198,6 +1199,8 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
 		return -ENOMEM;
 
 	aic32x4->dev = dev;
+	aic32x4->type = (enum aic32x4_type)dev_get_drvdata(dev);
+
 	dev_set_drvdata(dev, aic32x4);
 
 	if (pdata) {
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 7550122e9f8a..8a18dbec76a6 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -10,6 +10,11 @@
 struct device;
 struct regmap_config;
 
+enum aic32x4_type {
+	AIC32X4_TYPE_AIC32X4 = 0,
+	AIC32X4_TYPE_AIC32X6,
+};
+
 extern const struct regmap_config aic32x4_regmap_config;
 int aic32x4_probe(struct device *dev, struct regmap *regmap);
 int aic32x4_remove(struct device *dev);
-- 
cgit v1.2.3


From b4525b6196cd7f83eba16d8679a55f8bb9571052 Mon Sep 17 00:00:00 2001
From: Claudius Heine <ch@denx.de>
Date: Thu, 17 Jun 2021 10:52:29 +0200
Subject: ASoC: tlv320aic32x4: add support for TAS2505

This adds support for TAS2505 and TAS2521 to the tlv320aic32x4 driver.

The TAS2505 seems to be a stripped down version of the TLV320AIC32X4 so
it makes sense to handle them in the same driver.

Signed-off-by: Claudius Heine <ch@denx.de>
Link: https://lore.kernel.org/r/20210617085230.1851503-3-ch@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic32x4-i2c.c |   2 +
 sound/soc/codecs/tlv320aic32x4.c     | 136 ++++++++++++++++++++++++++++++++++-
 sound/soc/codecs/tlv320aic32x4.h     |   5 ++
 3 files changed, 142 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
index 247fb1e13674..04ad38311360 100644
--- a/sound/soc/codecs/tlv320aic32x4-i2c.c
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -50,6 +50,7 @@ static int aic32x4_i2c_remove(struct i2c_client *i2c)
 static const struct i2c_device_id aic32x4_i2c_id[] = {
 	{ "tlv320aic32x4", (kernel_ulong_t)AIC32X4_TYPE_AIC32X4 },
 	{ "tlv320aic32x6", (kernel_ulong_t)AIC32X4_TYPE_AIC32X6 },
+	{ "tas2505", (kernel_ulong_t)AIC32X4_TYPE_TAS2505 },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
@@ -57,6 +58,7 @@ MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
 static const struct of_device_id aic32x4_of_id[] = {
 	{ .compatible = "ti,tlv320aic32x4", .data = (void *)AIC32X4_TYPE_AIC32X4 },
 	{ .compatible = "ti,tlv320aic32x6", .data = (void *)AIC32X4_TYPE_AIC32X6 },
+	{ .compatible = "ti,tas2505", .data = (void *)AIC32X4_TYPE_TAS2505 },
 	{ /* senitel */ }
 };
 MODULE_DEVICE_TABLE(of, aic32x4_of_id);
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 70a1574fb72a..c63b717040ed 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -251,6 +251,9 @@ static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
 /* -12dB min, 0.5dB steps */
 static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
 
+static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
+
 static const char * const lo_cm_text[] = {
 	"Full Chip", "1.65V",
 };
@@ -1059,6 +1062,129 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
+	SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+			AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+	SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
+	SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
+			AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
+			tlv_driver_gain),
+	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
+			AIC32X4_HPLGAIN, 6, 0x01, 1),
+
+	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+
+	SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
+			0, 0, 117, 1, tlv_spk_vol),
+	SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
+			4, 5, 0, tlv_amp_vol),
+};
+
+static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget aic32x4_tas2505_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", AIC32X4_DACSETUP, 7, 0),
+	SND_SOC_DAPM_MIXER("HP Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &hp_output_mixer_controls[0],
+			   ARRAY_SIZE(hp_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("HP Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Speaker Driver", TAS2505_SPK, 1, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("HP"),
+	SND_SOC_DAPM_OUTPUT("Speaker"),
+};
+
+static const struct snd_soc_dapm_route aic32x4_tas2505_dapm_routes[] = {
+	/* Left Output */
+	{"HP Output Mixer", "DAC Switch", "DAC"},
+
+	{"HP Power", NULL, "HP Output Mixer"},
+	{"HP", NULL, "HP Power"},
+
+	{"Speaker Driver", NULL, "DAC"},
+	{"Speaker", NULL, "Speaker Driver"},
+};
+
+static struct snd_soc_dai_driver aic32x4_tas2505_dai = {
+	.name = "tas2505-hifi",
+	.playback = {
+			 .stream_name = "Playback",
+			 .channels_min = 1,
+			 .channels_max = 1,
+			 .rates = SNDRV_PCM_RATE_8000_96000,
+			 .formats = AIC32X4_FORMATS,},
+	.ops = &aic32x4_ops,
+	.symmetric_rate = 1,
+};
+
+static int aic32x4_tas2505_component_probe(struct snd_soc_component *component)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
+	u32 tmp_reg;
+	int ret;
+
+	struct clk_bulk_data clocks[] = {
+		{ .id = "codec_clkin" },
+		{ .id = "pll" },
+		{ .id = "bdiv" },
+		{ .id = "mdac" },
+	};
+
+	ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks);
+	if (ret)
+		return ret;
+
+	if (aic32x4->setup)
+		aic32x4_setup_gpios(component);
+
+	clk_set_parent(clocks[0].clk, clocks[1].clk);
+	clk_set_parent(clocks[2].clk, clocks[3].clk);
+
+	/* Power platform configuration */
+	if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE)
+		snd_soc_component_write(component, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
+
+	tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
+			AIC32X4_LDOCTLEN : 0;
+	snd_soc_component_write(component, AIC32X4_LDOCTL, tmp_reg);
+
+	tmp_reg = snd_soc_component_read(component, AIC32X4_CMMODE);
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36)
+		tmp_reg |= AIC32X4_LDOIN_18_36;
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED)
+		tmp_reg |= AIC32X4_LDOIN2HP;
+	snd_soc_component_write(component, AIC32X4_CMMODE, tmp_reg);
+
+	/*
+	 * Enable the fast charging feature and ensure the needed 40ms ellapsed
+	 * before using the analog circuits.
+	 */
+	snd_soc_component_write(component, TAS2505_REFPOWERUP,
+				AIC32X4_REFPOWERUP_40MS);
+	msleep(40);
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = {
+	.probe			= aic32x4_tas2505_component_probe,
+	.set_bias_level		= aic32x4_set_bias_level,
+	.controls		= aic32x4_tas2505_snd_controls,
+	.num_controls		= ARRAY_SIZE(aic32x4_tas2505_snd_controls),
+	.dapm_widgets		= aic32x4_tas2505_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aic32x4_tas2505_dapm_widgets),
+	.dapm_routes		= aic32x4_tas2505_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(aic32x4_tas2505_dapm_routes),
+	.suspend_bias_off	= 1,
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
 static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
 		struct device_node *np)
 {
@@ -1250,8 +1376,16 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
 	if (ret)
 		goto err_disable_regulators;
 
-	ret = devm_snd_soc_register_component(dev,
+	switch (aic32x4->type) {
+	case AIC32X4_TYPE_TAS2505:
+		ret = devm_snd_soc_register_component(dev,
+			&soc_component_dev_aic32x4_tas2505, &aic32x4_tas2505_dai, 1);
+		break;
+	default:
+		ret = devm_snd_soc_register_component(dev,
 			&soc_component_dev_aic32x4, &aic32x4_dai, 1);
+	}
+
 	if (ret) {
 		dev_err(dev, "Failed to register component\n");
 		goto err_disable_regulators;
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 8a18dbec76a6..e9fd2e55d6c3 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -13,6 +13,7 @@ struct regmap_config;
 enum aic32x4_type {
 	AIC32X4_TYPE_AIC32X4 = 0,
 	AIC32X4_TYPE_AIC32X6,
+	AIC32X4_TYPE_TAS2505,
 };
 
 extern const struct regmap_config aic32x4_regmap_config;
@@ -93,6 +94,9 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name);
 #define	AIC32X4_LOLGAIN		AIC32X4_REG(1, 18)
 #define	AIC32X4_LORGAIN		AIC32X4_REG(1, 19)
 #define AIC32X4_HEADSTART	AIC32X4_REG(1, 20)
+#define TAS2505_SPK		AIC32X4_REG(1, 45)
+#define TAS2505_SPKVOL1		AIC32X4_REG(1, 46)
+#define TAS2505_SPKVOL2		AIC32X4_REG(1, 48)
 #define AIC32X4_MICBIAS		AIC32X4_REG(1, 51)
 #define AIC32X4_LMICPGAPIN	AIC32X4_REG(1, 52)
 #define AIC32X4_LMICPGANIN	AIC32X4_REG(1, 54)
@@ -101,6 +105,7 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name);
 #define AIC32X4_FLOATINGINPUT	AIC32X4_REG(1, 58)
 #define AIC32X4_LMICPGAVOL	AIC32X4_REG(1, 59)
 #define AIC32X4_RMICPGAVOL	AIC32X4_REG(1, 60)
+#define TAS2505_REFPOWERUP	AIC32X4_REG(1, 122)
 #define AIC32X4_REFPOWERUP	AIC32X4_REG(1, 123)
 
 /* Bits, masks, and shifts */
-- 
cgit v1.2.3