From 5436f59bc5bcc27ca0244ebdb02b4228bf8b1aae Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 17 Aug 2020 10:21:38 +0200
Subject: [PATCH 01/70] ALSA: usb-audio: Move device rename and profile quirks
 to an internal table

So far we've added the devices that need vendor/product string renames
or the profile setup into the standard quirk table in quirks-table.h.
This table is imported into the primary USB audio device entry, hence
it's all exported for the probing so that udev and co can take a look
at it.  OTOH, for renaming or profile setup, we don't need to expose
those explicit entries because the probe itself follows the standard
way.  That said, we're exposing unnecessarily too many entries.

This patch moves such internal quirk entries into the own table, and
reduces the exported device table size.  Along with the moving items,
re-arrange the entries in the proper order.

Link: https://lore.kernel.org/r/20200817082140.20232-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/card.c         | 131 ++++++++++++++++++++++++++++++++++++---
 sound/usb/quirks-table.h | 120 -----------------------------------
 sound/usb/usbaudio.h     |   1 -
 3 files changed, 123 insertions(+), 129 deletions(-)

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 696e788c5d31..fa764b61fe9c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -332,6 +332,106 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 	return 0;
 }
 
+/*
+ * Profile name preset table
+ */
+struct usb_audio_device_name {
+	u32 id;
+	const char *vendor_name;
+	const char *product_name;
+	const char *profile_name;	/* override card->longname */
+};
+
+#define PROFILE_NAME(vid, pid, vendor, product, profile)	 \
+	{ .id = USB_ID(vid, pid), .vendor_name = (vendor),	 \
+	  .product_name = (product), .profile_name = (profile) }
+#define DEVICE_NAME(vid, pid, vendor, product) \
+	PROFILE_NAME(vid, pid, vendor, product, NULL)
+
+/* vendor/product and profile name presets, sorted in device id order */
+static const struct usb_audio_device_name usb_audio_names[] = {
+	/* HP Thunderbolt Dock Audio Headset */
+	PROFILE_NAME(0x03f0, 0x0269, "HP", "Thunderbolt Dock Audio Headset",
+		     "HP-Thunderbolt-Dock-Audio-Headset"),
+	/* HP Thunderbolt Dock Audio Module */
+	PROFILE_NAME(0x03f0, 0x0567, "HP", "Thunderbolt Dock Audio Module",
+		     "HP-Thunderbolt-Dock-Audio-Module"),
+
+	/* Two entries for Gigabyte TRX40 Aorus Master:
+	 * TRX40 Aorus Master has two USB-audio devices, one for the front
+	 * headphone with ESS SABRE9218 DAC chip, while another for the rest
+	 * I/O (the rear panel and the front mic) with Realtek ALC1220-VB.
+	 * Here we provide two distinct names for making UCM profiles easier.
+	 */
+	PROFILE_NAME(0x0414, 0xa000, "Gigabyte", "Aorus Master Front Headphone",
+		     "Gigabyte-Aorus-Master-Front-Headphone"),
+	PROFILE_NAME(0x0414, 0xa001, "Gigabyte", "Aorus Master Main Audio",
+		     "Gigabyte-Aorus-Master-Main-Audio"),
+
+	/* Gigabyte TRX40 Aorus Pro WiFi */
+	PROFILE_NAME(0x0414, 0xa002,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	/* Creative/E-Mu devices */
+	DEVICE_NAME(0x041e, 0x3010, "Creative Labs", "Sound Blaster MP3+"),
+	/* Creative/Toshiba Multimedia Center SB-0500 */
+	DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"),
+
+	DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"),
+
+	/* Dell WD15 Dock */
+	PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"),
+	/* Dell WD19 Dock */
+	PROFILE_NAME(0x0bda, 0x402e, "Dell", "WD19 Dock", "Dell-WD15-Dock"),
+
+	DEVICE_NAME(0x0ccd, 0x0028, "TerraTec", "Aureon5.1MkII"),
+
+	/*
+	 * The original product_name is "USB Sound Device", however this name
+	 * is also used by the CM106 based cards, so make it unique.
+	 */
+	DEVICE_NAME(0x0d8c, 0x0102, NULL, "ICUSBAUDIO7D"),
+	DEVICE_NAME(0x0d8c, 0x0103, NULL, "Audio Advantage MicroII"),
+
+	/* MSI TRX40 Creator */
+	PROFILE_NAME(0x0db0, 0x0d64,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+	/* MSI TRX40 */
+	PROFILE_NAME(0x0db0, 0x543d,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
+	DEVICE_NAME(0x103d, 0x0100, "Stanton", "ScratchAmp"),
+	DEVICE_NAME(0x103d, 0x0101, "Stanton", "ScratchAmp"),
+
+	/* aka. Serato Scratch Live DJ Box */
+	DEVICE_NAME(0x13e5, 0x0001, "Rane", "SL-1"),
+
+	/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
+	PROFILE_NAME(0x17aa, 0x1046, "Lenovo", "ThinkStation P620 Rear",
+		     "Lenovo-ThinkStation-P620-Rear"),
+	/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
+	PROFILE_NAME(0x17aa, 0x104d, "Lenovo", "ThinkStation P620 Main",
+		     "Lenovo-ThinkStation-P620-Main"),
+
+	/* Asrock TRX40 Creator */
+	PROFILE_NAME(0x26ce, 0x0a01,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	{ } /* terminator */
+};
+
+static const struct usb_audio_device_name *
+lookup_device_name(u32 id)
+{
+	static const struct usb_audio_device_name *p;
+
+	for (p = usb_audio_names; p->id; p++)
+		if (p->id == id)
+			return p;
+	return NULL;
+}
+
 /*
  * free the chip instance
  *
@@ -357,10 +457,16 @@ static void usb_audio_make_shortname(struct usb_device *dev,
 				     const struct snd_usb_audio_quirk *quirk)
 {
 	struct snd_card *card = chip->card;
+	const struct usb_audio_device_name *preset;
+	const char *s = NULL;
 
-	if (quirk && quirk->product_name && *quirk->product_name) {
-		strlcpy(card->shortname, quirk->product_name,
-			sizeof(card->shortname));
+	preset = lookup_device_name(chip->usb_id);
+	if (preset && preset->product_name)
+		s = preset->product_name;
+	else if (quirk && quirk->product_name)
+		s = quirk->product_name;
+	if (s && *s) {
+		strlcpy(card->shortname, s, sizeof(card->shortname));
 		return;
 	}
 
@@ -382,17 +488,26 @@ static void usb_audio_make_longname(struct usb_device *dev,
 				    const struct snd_usb_audio_quirk *quirk)
 {
 	struct snd_card *card = chip->card;
+	const struct usb_audio_device_name *preset;
+	const char *s = NULL;
 	int len;
 
+	preset = lookup_device_name(chip->usb_id);
+
 	/* shortcut - if any pre-defined string is given, use it */
-	if (quirk && quirk->profile_name && *quirk->profile_name) {
-		strlcpy(card->longname, quirk->profile_name,
-			sizeof(card->longname));
+	if (preset && preset->profile_name)
+		s = preset->profile_name;
+	if (s && *s) {
+		strlcpy(card->longname, s, sizeof(card->longname));
 		return;
 	}
 
-	if (quirk && quirk->vendor_name && *quirk->vendor_name) {
-		len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
+	if (preset && preset->vendor_name)
+		s = preset->vendor_name;
+	else if (quirk && quirk->vendor_name)
+		s = quirk->vendor_name;
+	if (s && *s) {
+		len = strlcpy(card->longname, s, sizeof(card->longname));
 	} else {
 		/* retrieve the vendor and device strings as longname */
 		if (dev->descriptor.iManufacturer)
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f4fb002e3ef4..978f7113bd81 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -25,33 +25,6 @@
 	.idProduct = prod, \
 	.bInterfaceClass = USB_CLASS_VENDOR_SPEC
 
-#define QUIRK_RENAME_DEVICE(_vendor, _device)				\
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
-		.vendor_name = _vendor,					\
-		.product_name = _device,				\
-		.ifnum = QUIRK_NO_INTERFACE				\
-	}
-
-#define QUIRK_DEVICE_PROFILE(_vendor, _device, _profile)		\
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
-		.vendor_name = _vendor,					\
-		.product_name = _device,				\
-		.profile_name = _profile,				\
-		.ifnum = QUIRK_NO_INTERFACE				\
-	}
-
-/* HP Thunderbolt Dock Audio Headset */
-{
-	USB_DEVICE(0x03f0, 0x0269),
-	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset",
-			     "HP-Thunderbolt-Dock-Audio-Headset"),
-},
-/* HP Thunderbolt Dock Audio Module */
-{
-	USB_DEVICE(0x03f0, 0x0567),
-	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module",
-			     "HP-Thunderbolt-Dock-Audio-Module"),
-},
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
@@ -85,16 +58,6 @@
 	}
 },
 
-/* Creative/E-Mu devices */
-{
-	USB_DEVICE(0x041e, 0x3010),
-	QUIRK_RENAME_DEVICE("Creative Labs", "Sound Blaster MP3+")
-},
-/* Creative/Toshiba Multimedia Center SB-0500 */
-{
-	USB_DEVICE(0x041e, 0x3048),
-	QUIRK_RENAME_DEVICE("Toshiba", "SB-0500")
-},
 {
 	/* E-Mu 0202 USB */
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
@@ -226,7 +189,6 @@
 	.idProduct = 0x0990,
 	.bInterfaceClass = USB_CLASS_AUDIO,
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
-	QUIRK_RENAME_DEVICE("Logitech, Inc.", "QuickCam Pro 9000")
 },
 
 /*
@@ -2609,10 +2571,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.type = QUIRK_MIDI_STANDARD_INTERFACE
 	}
 },
-{
-	USB_DEVICE(0x0ccd, 0x0028),
-	QUIRK_RENAME_DEVICE("TerraTec", "Aureon5.1MkII")
-},
 {
 	USB_DEVICE(0x0ccd, 0x0035),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
@@ -2623,16 +2581,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
-/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
-{
-	USB_DEVICE(0x103d, 0x0100),
-	QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp")
-},
-{
-	USB_DEVICE(0x103d, 0x0101),
-	QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp")
-},
-
 /* Novation EMS devices */
 {
 	USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001),
@@ -2817,26 +2765,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
-/* */
-{
-	/* aka. Serato Scratch Live DJ Box */
-	USB_DEVICE(0x13e5, 0x0001),
-	QUIRK_RENAME_DEVICE("Rane", "SL-1")
-},
-
-/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
-{
-	USB_DEVICE(0x17aa, 0x1046),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
-			     "Lenovo-ThinkStation-P620-Rear"),
-},
-/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
-{
-	USB_DEVICE(0x17aa, 0x104d),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
-			     "Lenovo-ThinkStation-P620-Main"),
-},
-
 /* Native Instruments MK2 series */
 {
 	/* Komplete Audio 6 */
@@ -3295,19 +3223,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
-/*
- * The original product_name is "USB Sound Device", however this name
- * is also used by the CM106 based cards, so make it unique.
- */
-{
-	USB_DEVICE(0x0d8c, 0x0102),
-	QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D")
-},
-{
-	USB_DEVICE(0x0d8c, 0x0103),
-	QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII")
-},
-
 /* disabled due to regression for other devices;
  * see https://bugzilla.kernel.org/show_bug.cgi?id=199905
  */
@@ -3408,18 +3323,10 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
-/* Dell WD15 Dock */
-{
-	USB_DEVICE(0x0bda, 0x4014),
-	QUIRK_DEVICE_PROFILE("Dell", "WD15 Dock", "Dell-WD15-Dock")
-},
 /* Dell WD19 Dock */
 {
 	USB_DEVICE(0x0bda, 0x402e),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Dell",
-		.product_name = "WD19 Dock",
-		.profile_name = "Dell-WD15-Dock",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_SETUP_FMT_AFTER_RESUME
 	}
@@ -3645,33 +3552,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
-#define ALC1220_VB_DESKTOP(vend, prod) { \
-	USB_DEVICE(vend, prod),	\
-	QUIRK_DEVICE_PROFILE("Realtek", "ALC1220-VB-DT", \
-			     "Realtek-ALC1220-VB-Desktop") \
-}
-ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */
-ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */
-ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */
-ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
-#undef ALC1220_VB_DESKTOP
-
-/* Two entries for Gigabyte TRX40 Aorus Master:
- * TRX40 Aorus Master has two USB-audio devices, one for the front headphone
- * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear
- * panel and the front mic) with Realtek ALC1220-VB.
- * Here we provide two distinct names for making UCM profiles easier.
- */
-{
-	USB_DEVICE(0x0414, 0xa000),
-	QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Front Headphone",
-			     "Gigabyte-Aorus-Master-Front-Headphone")
-},
-{
-	USB_DEVICE(0x0414, 0xa001),
-	QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Main Audio",
-			     "Gigabyte-Aorus-Master-Main-Audio")
-},
 {
 	/*
 	 * Pioneer DJ DJM-900NXS2
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b91c4c0807ec..2275295fe71f 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -109,7 +109,6 @@ enum quirk_type {
 struct snd_usb_audio_quirk {
 	const char *vendor_name;
 	const char *product_name;
-	const char *profile_name;	/* override the card->longname */
 	int16_t ifnum;
 	uint16_t type;
 	bool shares_media_device;

From fa10635fca359f047df6a18b3befd2f1e7304e1a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 17 Aug 2020 10:21:39 +0200
Subject: [PATCH 02/70] ALSA: usb-audio: Simplify quirk entries with a macro

Introduce a new macro USB_AUDIO_DEVICE() for the entries matching with
the pid/vid pair and the class/subclass, and remove the open-code.

Link: https://lore.kernel.org/r/20200817082140.20232-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 105 ++++++++-------------------------------
 1 file changed, 21 insertions(+), 84 deletions(-)

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 978f7113bd81..988bb9d00192 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -25,6 +25,16 @@
 	.idProduct = prod, \
 	.bInterfaceClass = USB_CLASS_VENDOR_SPEC
 
+/* A standard entry matching with vid/pid and the audio class/subclass */
+#define USB_AUDIO_DEVICE(vend, prod) \
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
+		       USB_DEVICE_ID_MATCH_INT_CLASS | \
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
+	.idVendor = vend, \
+	.idProduct = prod, \
+	.bInterfaceClass = USB_CLASS_AUDIO, \
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
+
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
@@ -127,69 +137,13 @@
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
  * class matches do not take effect without an explicit ID match.
  */
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x0850,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08ae,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08c6,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f0,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f5,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f6,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x0990,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
-},
+{ USB_AUDIO_DEVICE(0x046d, 0x0850) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08ae) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08c6) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f0) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f5) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f6) },
+{ USB_AUDIO_DEVICE(0x046d, 0x0990) },
 
 /*
  * Yamaha devices
@@ -2831,13 +2785,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
  */
 
 #define AU0828_DEVICE(vid, pid, vname, pname) { \
-	.idVendor = vid, \
-	.idProduct = pid, \
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
-		       USB_DEVICE_ID_MATCH_INT_CLASS | \
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
-	.bInterfaceClass = USB_CLASS_AUDIO, \
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \
+	USB_AUDIO_DEVICE(vid, pid), \
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \
 		.vendor_name = vname, \
 		.product_name = pname, \
@@ -2867,13 +2815,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 
 /* Syntek STK1160 */
 {
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x05e1,
-	.idProduct = 0x0408,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	USB_AUDIO_DEVICE(0x05e1, 0x0408),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "Syntek",
 		.product_name = "STK1160",
@@ -3628,13 +3570,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
  * channels to be swapped and out of phase, which is dealt with in quirks.c.
  */
 {
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x534d,
-	.idProduct = 0x2109,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	USB_AUDIO_DEVICE(0x534d, 0x2109),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MacroSilicon",
 		.product_name = "MS2109",
@@ -3675,3 +3611,4 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 
 #undef USB_DEVICE_VENDOR_SPEC
+#undef USB_AUDIO_DEVICE

From 51ab5d77dcb8a50283c0eded08c67a6d275cb910 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 17 Aug 2020 10:21:40 +0200
Subject: [PATCH 03/70] ALSA: usb-audio: Properly match with audio interface
 class

There are a few entries in the quirk table that set the device ID with
USB_DEVICE() macro while having an extra bInterfaceClass field.  But
bInterfaceClass field is never checked unless the proper match_flags
is set, so those may match incorrectly with all interfaces.

Introduce another macro to match with the vid/pid pair and the audio
class interface, and apply it to such entries, so that they can match
properly.

Link: https://lore.kernel.org/r/20200817082140.20232-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 55 ++++++++++++++--------------------------
 1 file changed, 19 insertions(+), 36 deletions(-)

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 988bb9d00192..7a80ef31bbe4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -35,6 +35,14 @@
 	.bInterfaceClass = USB_CLASS_AUDIO, \
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 
+/* Another standard entry matching with vid/pid and the audio class */
+#define USB_AUDIO_CLASS(vend, prod) \
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
+		       USB_DEVICE_ID_MATCH_INT_CLASS, \
+	.idVendor = vend, \
+	.idProduct = prod, \
+	.bInterfaceClass = USB_CLASS_AUDIO
+
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
@@ -68,34 +76,14 @@
 	}
 },
 
-{
-	/* E-Mu 0202 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f02,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu 0404 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f04,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu Tracker Pre */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f0a,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu 0204 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f19,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
+/* E-Mu 0202 USB */
+{ USB_AUDIO_CLASS(0x041e, 0x3f02) },
+/* E-Mu 0404 USB */
+{ USB_AUDIO_CLASS(0x041e, 0x3f04) },
+/* E-Mu Tracker Pre */
+{ USB_AUDIO_CLASS(0x041e, 0x3f0a) },
+/* E-Mu 0204 USB */
+{ USB_AUDIO_CLASS(0x041e, 0x3f19) },
 
 /*
  * HP Wireless Audio
@@ -2751,10 +2739,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 },
 
 /* KeithMcMillen Stringport */
-{
-	USB_DEVICE(0x1f38, 0x0001),
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
+{ USB_AUDIO_CLASS(0x1f38, 0x0001) },
 
 /* Miditech devices */
 {
@@ -2977,10 +2962,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 {
 	/* Tascam US122 MKII - playback-only support */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x0644,
-	.idProduct = 0x8021,
-	.bInterfaceClass = USB_CLASS_AUDIO,
+	USB_AUDIO_CLASS(0x0644, 0x8021),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "TASCAM",
 		.product_name = "US122 MKII",
@@ -3612,3 +3594,4 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 
 #undef USB_DEVICE_VENDOR_SPEC
 #undef USB_AUDIO_DEVICE
+#undef USB_AUDIO_CLASS

From 14b51ccd26742811ced3ee5e52ee6544f5d6105c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 23 Aug 2020 13:32:51 +0200
Subject: [PATCH 04/70] ALSA: usb-audio: Correct wrongly matching entries with
 audio class

The commit 51ab5d77dcb8 ("ALSA: usb-audio: Properly match with audio
interface class") converted the quirk entries that have both vid/pid
pair and bInterface fields to match with all those with a new macro
USB_AUDIO_CLASS().  However, it turned out that those are false
conversions; all those (but the unknown KeithMcMillen device) are
actually with vendor-specific interface class, hence the conversions
broke the matching.

This patch corrects those entries to the right one,
USB_DEVICE_VENDOR_SPEC() (and USB_DEVICE() for KeithMcMillen to be
sure), and drop the unused USB_AUDIO_CLASS macro again.

Fixes: 51ab5d77dcb8 ("ALSA: usb-audio: Properly match with audio interface class")
Reported-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200823113251.10175-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 21 ++++++---------------
 1 file changed, 6 insertions(+), 15 deletions(-)

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 969c79d0c688..7fe9d3e75d59 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -35,14 +35,6 @@
 	.bInterfaceClass = USB_CLASS_AUDIO, \
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 
-/* Another standard entry matching with vid/pid and the audio class */
-#define USB_AUDIO_CLASS(vend, prod) \
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
-		       USB_DEVICE_ID_MATCH_INT_CLASS, \
-	.idVendor = vend, \
-	.idProduct = prod, \
-	.bInterfaceClass = USB_CLASS_AUDIO
-
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
@@ -77,13 +69,13 @@
 },
 
 /* E-Mu 0202 USB */
-{ USB_AUDIO_CLASS(0x041e, 0x3f02) },
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f02) },
 /* E-Mu 0404 USB */
-{ USB_AUDIO_CLASS(0x041e, 0x3f04) },
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f04) },
 /* E-Mu Tracker Pre */
-{ USB_AUDIO_CLASS(0x041e, 0x3f0a) },
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f0a) },
 /* E-Mu 0204 USB */
-{ USB_AUDIO_CLASS(0x041e, 0x3f19) },
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) },
 
 /*
  * HP Wireless Audio
@@ -2756,7 +2748,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 },
 
 /* KeithMcMillen Stringport */
-{ USB_AUDIO_CLASS(0x1f38, 0x0001) },
+{ USB_DEVICE(0x1f38, 0x0001) }, /* FIXME: should be more restrictive matching */
 
 /* Miditech devices */
 {
@@ -2979,7 +2971,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 {
 	/* Tascam US122 MKII - playback-only support */
-	USB_AUDIO_CLASS(0x0644, 0x8021),
+	USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "TASCAM",
 		.product_name = "US122 MKII",
@@ -3611,4 +3603,3 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 
 #undef USB_DEVICE_VENDOR_SPEC
 #undef USB_AUDIO_DEVICE
-#undef USB_AUDIO_CLASS

From bf2aa9ccc8e598d788132ded9c7e94c6af3a9d1e Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:20 -0400
Subject: [PATCH 05/70] ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function.

Cleanup the ca0132_mmio_init function, separating into two separate
functions, one for Sound Blaster Z/ZxR/Recon3D, and another for the
AE-5.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-2-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 176 +++++++++++++++++++++--------------
 1 file changed, 108 insertions(+), 68 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7dbf2e7f77a..7491e2044638 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8108,78 +8108,118 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec)
  * what they do, or if they're necessary. Could possibly
  * be removed. Figure they're better to leave in.
  */
+static const unsigned int ca0113_mmio_init_address_sbz[] = {
+	0x400, 0x408, 0x40c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c,
+	0xc0c, 0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04
+};
+
+static const unsigned int ca0113_mmio_init_data_sbz[] = {
+	0x00000030, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000003, 0x000000c1, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_data_zxr[] = {
+	0x00000030, 0x00000000, 0x00000000, 0x00000003, 0x00000003,
+	0x00000003, 0x00000001, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_address_ae5[] = {
+	0x400, 0x42c, 0x46c, 0x4ac, 0x4ec, 0x43c, 0x47c, 0x4bc, 0x4fc, 0x408,
+	0x100, 0x410, 0x40c, 0x100, 0x100, 0x830, 0x86c, 0x800, 0x86c, 0x800,
+	0x804, 0x20c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c, 0xc0c,
+	0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04, 0x01c
+};
+
+static const unsigned int ca0113_mmio_init_data_ae5[] = {
+	0x00000001, 0x00000000, 0x00000000, 0x00000000, 0x00000000,
+	0x00000000, 0x00000000, 0x00000000, 0x00000000, 0x00000001,
+	0x00000600, 0x00000014, 0x00000001, 0x0000060f, 0x0000070f,
+	0x00000aff, 0x00000000, 0x0000006b, 0x00000001, 0x0000006b,
+	0x00000057, 0x00800000, 0x00880680, 0x00000080, 0x00000030,
+	0x00000000, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000001, 0x000000f1, 0x00000001, 0x000000c7, 0x000000c1,
+	0x00000080, 0x00880680
+};
+
+static void ca0132_mmio_init_sbz(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp[2], i, count, cur_addr;
+	const unsigned int *addr, *data;
+
+	addr = ca0113_mmio_init_address_sbz;
+	for (i = 0; i < 3; i++)
+		writel(0x00000000, spec->mem_base + addr[i]);
+
+	cur_addr = i;
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		tmp[0] = 0x00880480;
+		tmp[1] = 0x00000080;
+		break;
+	case QUIRK_SBZ:
+		tmp[0] = 0x00820680;
+		tmp[1] = 0x00000083;
+		break;
+	case QUIRK_R3D:
+		tmp[0] = 0x00880680;
+		tmp[1] = 0x00000083;
+		break;
+	default:
+		tmp[0] = 0x00000000;
+		tmp[1] = 0x00000000;
+		break;
+	}
+
+	for (i = 0; i < 2; i++)
+		writel(tmp[i], spec->mem_base + addr[cur_addr + i]);
+
+	cur_addr += i;
+
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_zxr);
+		data = ca0113_mmio_init_data_zxr;
+		break;
+	default:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_sbz);
+		data = ca0113_mmio_init_data_sbz;
+		break;
+	}
+
+	for (i = 0; i < count; i++)
+		writel(data[i], spec->mem_base + addr[cur_addr + i]);
+}
+
+static void ca0132_mmio_init_ae5(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int *addr, *data;
+	unsigned int i, count;
+
+	addr = ca0113_mmio_init_address_ae5;
+	data = ca0113_mmio_init_data_ae5;
+	count = ARRAY_SIZE(ca0113_mmio_init_data_ae5);
+
+	for (i = 0; i < count; i++)
+		writel(data[i], spec->mem_base + addr[i]);
+}
+
 static void ca0132_mmio_init(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x400);
-	else
-		writel(0x00000000, spec->mem_base + 0x400);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x408);
-	else
-		writel(0x00000000, spec->mem_base + 0x408);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x40c);
-	else
-		writel(0x00000000, spec->mem_base + 0x40C);
-
-	if (ca0132_quirk(spec) == QUIRK_ZXR)
-		writel(0x00880640, spec->mem_base + 0x01C);
-	else
-		writel(0x00880680, spec->mem_base + 0x01C);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000080, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000083, spec->mem_base + 0xC0C);
-
-	writel(0x00000030, spec->mem_base + 0xC00);
-	writel(0x00000000, spec->mem_base + 0xC04);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000000, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000003, spec->mem_base + 0xC0C);
-
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0xC08);
-	else
-		writel(0x000000C1, spec->mem_base + 0xC08);
-
-	writel(0x000000F1, spec->mem_base + 0xC08);
-	writel(0x00000001, spec->mem_base + 0xC08);
-	writel(0x000000C7, spec->mem_base + 0xC08);
-	writel(0x000000C1, spec->mem_base + 0xC08);
-	writel(0x00000080, spec->mem_base + 0xC04);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
-		writel(0x00000000, spec->mem_base + 0x42c);
-		writel(0x00000000, spec->mem_base + 0x46c);
-		writel(0x00000000, spec->mem_base + 0x4ac);
-		writel(0x00000000, spec->mem_base + 0x4ec);
-		writel(0x00000000, spec->mem_base + 0x43c);
-		writel(0x00000000, spec->mem_base + 0x47c);
-		writel(0x00000000, spec->mem_base + 0x4bc);
-		writel(0x00000000, spec->mem_base + 0x4fc);
-		writel(0x00000600, spec->mem_base + 0x100);
-		writel(0x00000014, spec->mem_base + 0x410);
-		writel(0x0000060f, spec->mem_base + 0x100);
-		writel(0x0000070f, spec->mem_base + 0x100);
-		writel(0x00000aff, spec->mem_base + 0x830);
-		writel(0x00000000, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000001, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000057, spec->mem_base + 0x804);
-		writel(0x00800000, spec->mem_base + 0x20c);
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_R3D:
+	case QUIRK_SBZ:
+	case QUIRK_ZXR:
+		ca0132_mmio_init_sbz(codec);
+		break;
+	case QUIRK_AE5:
+		ca0132_mmio_init_ae5(codec);
+		break;
 	}
 }
 

From 896e361e82423aed4490f485dc25de1958c724ed Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:21 -0400
Subject: [PATCH 06/70] ALSA: hda/ca0132 - Add speaker tuning initialization
 commands.

Add speaker tuning initialization DSP commands, and also define
previously unknown DSP command values.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 119 +++++++++++++++++++++++++++++++++++
 1 file changed, 119 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 7491e2044638..2e664aeee1c4 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -589,6 +589,60 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
 	}
 };
 
+/*
+ * Definitions for the DSP req's to handle speaker tuning. These all belong to
+ * module ID 0x96, the output effects module.
+ */
+enum speaker_tuning_reqs {
+	/*
+	 * Currently, this value is always set to 0.0f. However, on Windows,
+	 * when selecting certain headphone profiles on the new Sound Blaster
+	 * connect software, the QUERY_SPEAKER_EQ_ADDRESS req on mid 0x80 is
+	 * sent. This gets the speaker EQ address area, which is then used to
+	 * send over (presumably) an equalizer profile for the specific
+	 * headphone setup. It is sent using the same method the DSP
+	 * firmware is uploaded with, which I believe is why the 'ctspeq.bin'
+	 * file exists in linux firmware tree but goes unused. It would also
+	 * explain why the QUERY_SPEAKER_EQ_ADDRESS req is defined but unused.
+	 * Once this profile is sent over, SPEAKER_TUNING_USE_SPEAKER_EQ is
+	 * set to 1.0f.
+	 */
+	SPEAKER_TUNING_USE_SPEAKER_EQ           = 0x1f,
+	SPEAKER_TUNING_ENABLE_CENTER_EQ         = 0x20,
+	SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL     = 0x21,
+	SPEAKER_TUNING_FRONT_RIGHT_VOL_LEVEL    = 0x22,
+	SPEAKER_TUNING_CENTER_VOL_LEVEL         = 0x23,
+	SPEAKER_TUNING_LFE_VOL_LEVEL            = 0x24,
+	SPEAKER_TUNING_REAR_LEFT_VOL_LEVEL      = 0x25,
+	SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL     = 0x26,
+	SPEAKER_TUNING_SURROUND_LEFT_VOL_LEVEL  = 0x27,
+	SPEAKER_TUNING_SURROUND_RIGHT_VOL_LEVEL = 0x28,
+	/*
+	 * Inversion is used when setting headphone virtualization to line
+	 * out. Not sure why this is, but it's the only place it's ever used.
+	 */
+	SPEAKER_TUNING_FRONT_LEFT_INVERT        = 0x29,
+	SPEAKER_TUNING_FRONT_RIGHT_INVERT       = 0x2a,
+	SPEAKER_TUNING_CENTER_INVERT            = 0x2b,
+	SPEAKER_TUNING_LFE_INVERT               = 0x2c,
+	SPEAKER_TUNING_REAR_LEFT_INVERT         = 0x2d,
+	SPEAKER_TUNING_REAR_RIGHT_INVERT        = 0x2e,
+	SPEAKER_TUNING_SURROUND_LEFT_INVERT     = 0x2f,
+	SPEAKER_TUNING_SURROUND_RIGHT_INVERT    = 0x30,
+	/* Delay is used when setting surround speaker distance in Windows. */
+	SPEAKER_TUNING_FRONT_LEFT_DELAY         = 0x31,
+	SPEAKER_TUNING_FRONT_RIGHT_DELAY        = 0x32,
+	SPEAKER_TUNING_CENTER_DELAY             = 0x33,
+	SPEAKER_TUNING_LFE_DELAY                = 0x34,
+	SPEAKER_TUNING_REAR_LEFT_DELAY          = 0x35,
+	SPEAKER_TUNING_REAR_RIGHT_DELAY         = 0x36,
+	SPEAKER_TUNING_SURROUND_LEFT_DELAY      = 0x37,
+	SPEAKER_TUNING_SURROUND_RIGHT_DELAY     = 0x38,
+	/* Of these two, only mute seems to ever be used. */
+	SPEAKER_TUNING_MAIN_VOLUME              = 0x39,
+	SPEAKER_TUNING_MUTE                     = 0x3a,
+};
+
 /* DSP command sequences for ca0132_alt_select_out */
 #define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
 struct ca0132_alt_out_set {
@@ -6874,6 +6928,67 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 	}
 }
 
+/*
+ * Default speaker tuning values setup for alternative codecs.
+ */
+static const unsigned int sbz_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000198. */
+	0x394f9e38, 0x394f9e38, 0x00000000, 0x00000000, 0x00000000, 0x00000000
+};
+
+static const unsigned int zxr_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000220. */
+	0x00000000, 0x00000000, 0x3966afcd, 0x3966afcd, 0x3966afcd, 0x3966afcd
+};
+
+static const unsigned int ae5_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000100. */
+	0x00000000, 0x00000000, 0x38d1b717, 0x38d1b717, 0x38d1b717, 0x38d1b717
+};
+
+/*
+ * If we never change these, probably only need them on initialization.
+ */
+static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, tmp, start_req, end_req;
+	const unsigned int *values;
+
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_SBZ:
+		values = sbz_default_delay_values;
+		break;
+	case QUIRK_ZXR:
+		values = zxr_default_delay_values;
+		break;
+	case QUIRK_AE5:
+		values = ae5_default_delay_values;
+		break;
+	default:
+		values = sbz_default_delay_values;
+		break;
+	}
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_ENABLE_CENTER_EQ, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_INVERT;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_INVERT;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+
+	for (i = 0; i < 6; i++)
+		dspio_set_uint_param(codec, 0x96,
+				SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
+}
+
 /*
  * Creates a dummy stream to bind the output to. This seems to have to be done
  * after changing the main outputs source and destination streams.
@@ -7373,6 +7488,8 @@ static void sbz_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 
@@ -7440,6 +7557,8 @@ static void ae5_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 

From 01464a566eed527dbf977e2430afa249c257c008 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:22 -0400
Subject: [PATCH 07/70] ALSA: hda/ca0132 - Add surround channel config control.

Add a surround channel configuration enumeration control. Setting up
different channel configurations allows the DSP to upmix stereo audio
into multi-channel audio, and allows for redirection of bass to a
subwoofer.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-4-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 110 ++++++++++++++++++++++++++++++++++-
 1 file changed, 108 insertions(+), 2 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 2e664aeee1c4..1a5fb30b69ab 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -38,6 +38,8 @@
 #define FLOAT_ONE	0x3f800000
 #define FLOAT_TWO	0x40000000
 #define FLOAT_THREE     0x40400000
+#define FLOAT_FIVE	0x40a00000
+#define FLOAT_SIX       0x40c00000
 #define FLOAT_EIGHT     0x41000000
 #define FLOAT_MINUS_5	0xc0a00000
 
@@ -143,7 +145,8 @@ enum {
 	MIC_BOOST_ENUM,
 	AE5_HEADPHONE_GAIN_ENUM,
 	AE5_SOUND_FILTER_ENUM,
-	ZXR_HEADPHONE_GAIN
+	ZXR_HEADPHONE_GAIN,
+	SPEAKER_CHANNEL_CFG_ENUM,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -686,6 +689,39 @@ static const struct ca0132_alt_out_set alt_out_presets[] = {
 	}
 };
 
+/* Surround output channel count configuration structures. */
+#define SPEAKER_CHANNEL_CFG_COUNT 5
+enum {
+	SPEAKER_CHANNELS_2_0,
+	SPEAKER_CHANNELS_2_1,
+	SPEAKER_CHANNELS_4_0,
+	SPEAKER_CHANNELS_4_1,
+	SPEAKER_CHANNELS_5_1,
+};
+
+struct ca0132_alt_speaker_channel_cfg {
+	char *name;
+	unsigned int val;
+};
+
+static const struct ca0132_alt_speaker_channel_cfg speaker_channel_cfgs[] = {
+	{ .name = "2.0",
+	  .val = FLOAT_ONE
+	},
+	{ .name = "2.1",
+	  .val = FLOAT_TWO
+	},
+	{ .name = "4.0",
+	  .val = FLOAT_FIVE
+	},
+	{ .name = "4.1",
+	  .val = FLOAT_SIX
+	},
+	{ .name = "5.1",
+	  .val = FLOAT_EIGHT
+	}
+};
+
 /*
  * DSP volume setting structs. Req 1 is left volume, req 2 is right volume,
  * and I don't know what the third req is, but it's always zero. I assume it's
@@ -1063,6 +1099,7 @@ struct ca0132_spec {
 	/* ca0132_alt control related values */
 	unsigned char in_enum_val;
 	unsigned char out_enum_val;
+	unsigned char channel_cfg_val;
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
@@ -4476,7 +4513,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
 				    pin_ctl | PIN_OUT);
 
-		dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
+		dspio_set_uint_param(codec, 0x80, 0x04, tmp);
 		break;
 	}
 	/*
@@ -5582,6 +5620,54 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+/* Select surround output type: 2.1, 4.0, 4.1, or 5.1. */
+static int ca0132_alt_speaker_channel_cfg_get_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = items;
+	if (uinfo->value.enumerated.item >= items)
+		uinfo->value.enumerated.item = items - 1;
+	strcpy(uinfo->value.enumerated.name,
+			speaker_channel_cfgs[uinfo->value.enumerated.item].name);
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->channel_cfg_val;
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_speaker_channels: sel=%d, channels=%s\n",
+		    sel, speaker_channel_cfgs[sel].name);
+
+	spec->channel_cfg_val = sel;
+
+	if (spec->out_enum_val == SURROUND_OUT)
+		ca0132_alt_select_out(codec);
+
+	return 1;
+}
+
 /*
  * Smart Volume output setting control. Three different settings, Normal,
  * which takes the value from the smart volume slider. The two others, loud
@@ -6226,6 +6312,23 @@ static int ca0132_alt_add_output_enum(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Add a control for selecting channel count on speaker output. Setting this
+ * allows the DSP to do bass redirection and channel upmixing on surround
+ * configurations.
+ */
+static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Surround Channel Config",
+				    SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_speaker_channel_cfg_get_info;
+	knew.get = ca0132_alt_speaker_channel_cfg_get;
+	knew.put = ca0132_alt_speaker_channel_cfg_put;
+	return snd_hda_ctl_add(codec, SPEAKER_CHANNEL_CFG_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6530,6 +6633,9 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 */
 	if (ca0132_use_alt_functions(spec)) {
 		err = ca0132_alt_add_output_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_speaker_channel_cfg_enum(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);

From 670c5f484a4403e5c218442d5a0e337a2224741f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:23 -0400
Subject: [PATCH 08/70] ALSA: hda/ca0132 - Add full-range speaker selection
 controls.

Add functions for setting full-range speakers and controls to
enable/disable the setting. Setting a speaker to full-range means that
the channels won't have their bass redirected to the LFE channel.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-5-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 117 +++++++++++++++++++++++++++++++++++
 1 file changed, 117 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 1a5fb30b69ab..469cefe9a51a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -147,6 +147,8 @@ enum {
 	AE5_SOUND_FILTER_ENUM,
 	ZXR_HEADPHONE_GAIN,
 	SPEAKER_CHANNEL_CFG_ENUM,
+	SPEAKER_FULL_RANGE_FRONT,
+	SPEAKER_FULL_RANGE_REAR,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -592,6 +594,24 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
 	}
 };
 
+/*
+ * DSP reqs for handling full-range speakers/bass redirection. If a speaker is
+ * set as not being full range, and bass redirection is enabled, all
+ * frequencies below the crossover frequency are redirected to the LFE
+ * channel. If the surround configuration has no LFE channel, this can't be
+ * enabled. X-Bass must be disabled when using these.
+ */
+enum speaker_range_reqs {
+	SPEAKER_BASS_REDIRECT            = 0x15,
+	SPEAKER_BASS_REDIRECT_XOVER_FREQ = 0x16,
+	/* Between 0x16-0x1a are the X-Bass reqs. */
+	SPEAKER_FULL_RANGE_FRONT_L_R     = 0x1a,
+	SPEAKER_FULL_RANGE_CENTER_LFE    = 0x1b,
+	SPEAKER_FULL_RANGE_REAR_L_R      = 0x1c,
+	SPEAKER_FULL_RANGE_SURROUND_L_R  = 0x1d,
+	SPEAKER_BASS_REDIRECT_SUB_GAIN   = 0x1e,
+};
+
 /*
  * Definitions for the DSP req's to handle speaker tuning. These all belong to
  * module ID 0x96, the output effects module.
@@ -1100,6 +1120,7 @@ struct ca0132_spec {
 	unsigned char in_enum_val;
 	unsigned char out_enum_val;
 	unsigned char channel_cfg_val;
+	unsigned char speaker_range_val[2];
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
@@ -4259,6 +4280,50 @@ static void ae5_mmio_select_out(struct hda_codec *codec)
 			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
 }
 
+static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int err;
+
+	/* 2.0/4.0 setup has no LFE channel, so setting full-range does nothing. */
+	if (spec->channel_cfg_val == SPEAKER_CHANNELS_4_0
+			|| spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+		return 0;
+
+	/* Set front L/R full range. Zero for full-range, one for redirection. */
+	tmp = spec->speaker_range_val[0] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_FRONT_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/* When setting full-range rear, both rear and center/lfe are set. */
+	tmp = spec->speaker_range_val[1] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_CENTER_LFE, tmp);
+	if (err < 0)
+		return err;
+
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_REAR_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/*
+	 * Only the AE series cards set this value when setting full-range,
+	 * and it's always 1.0f.
+	 */
+	if (ca0132_quirk(spec) == QUIRK_AE5) {
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * These are the commands needed to setup output on each of the different card
  * types.
@@ -4539,6 +4604,9 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 			goto exit;
 	}
 
+	if (spec->cur_out_type == SURROUND_OUT)
+		err = ca0132_alt_set_full_range_speaker(codec);
+
 exit:
 	snd_hda_power_down_pm(codec);
 
@@ -5269,6 +5337,7 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 
 	*valp = spec->xbass_xover_freq;
+
 	return 0;
 }
 
@@ -5894,6 +5963,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		*valp = spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT];
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -5972,6 +6046,14 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		goto exit;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp;
+		if (spec->cur_out_type == SURROUND_OUT)
+			ca0132_alt_set_full_range_speaker(codec);
+
+		changed = 0;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -6329,6 +6411,31 @@ static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Full range front stereo and rear surround switches. When these are set to
+ * full range, the lower frequencies from these channels are no longer
+ * redirected to the LFE channel.
+ */
+static int ca0132_alt_add_front_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Front Speakers",
+				    SPEAKER_FULL_RANGE_FRONT, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_FRONT,
+				snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Rear Speakers",
+				    SPEAKER_FULL_RANGE_REAR, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_REAR,
+				snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6636,6 +6743,12 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_speaker_channel_cfg_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_front_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_rear_full_range_switch(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
@@ -7982,6 +8095,10 @@ static void ca0132_init_chip(struct hda_codec *codec)
 	 * ca0132 codecs. Also sets x-bass crossover frequency to 80hz.
 	 */
 	if (ca0132_use_alt_controls(spec)) {
+		/* Set speakers to default to full range. */
+		spec->speaker_range_val[0] = 1;
+		spec->speaker_range_val[1] = 1;
+
 		spec->xbass_xover_freq = 8;
 		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
 			spec->fx_ctl_val[i] = effect_slider_defaults[i];

From f49b3063ad0d8638da2e7eb994549863f2c74923 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:24 -0400
Subject: [PATCH 09/70] ALSA: hda/ca0132 - Add bass redirection controls.

Add bass redirection controls for surround outputs. This uses the DSP to
redirect audio below the bass redirection crossover frequency to the LFE
channel from the front/rear L/R speakers. This only goes into effect if
the speakers aren't set as full range, and only if the surround
configuration has an LFE channel.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-6-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 130 +++++++++++++++++++++++++++++++++--
 1 file changed, 126 insertions(+), 4 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 469cefe9a51a..8ad2fc5ab30b 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -149,6 +149,8 @@ enum {
 	SPEAKER_CHANNEL_CFG_ENUM,
 	SPEAKER_FULL_RANGE_FRONT,
 	SPEAKER_FULL_RANGE_REAR,
+	BASS_REDIRECTION,
+	BASS_REDIRECTION_XOVER,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -1123,6 +1125,8 @@ struct ca0132_spec {
 	unsigned char speaker_range_val[2];
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
+	unsigned char bass_redirection_val;
+	long bass_redirect_xover_freq;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
 	long xbass_xover_freq;
 	long eq_preset_val;
@@ -4324,6 +4328,35 @@ static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 	return 0;
 }
 
+static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec,
+		bool val)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int err;
+
+	if (val && spec->channel_cfg_val != SPEAKER_CHANNELS_4_0 &&
+			spec->channel_cfg_val != SPEAKER_CHANNELS_2_0)
+		tmp = FLOAT_ONE;
+	else
+		tmp = FLOAT_ZERO;
+
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_BASS_REDIRECT, tmp);
+	if (err < 0)
+		return err;
+
+	/* If it is enabled, make sure to set the crossover frequency. */
+	if (tmp) {
+		tmp = float_xbass_xover_lookup[spec->xbass_xover_freq];
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_BASS_REDIRECT_XOVER_FREQ, tmp);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * These are the commands needed to setup output on each of the different card
  * types.
@@ -4593,6 +4626,15 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
+	if (spec->cur_out_type == SURROUND_OUT)
+		err = ca0132_alt_surround_set_bass_redirection(codec,
+				spec->bass_redirection_val);
+	else
+		err = ca0132_alt_surround_set_bass_redirection(codec, 0);
+
+	if (err < 0)
+		goto exit;
+
 	/* run through the output dsp commands for the selected output. */
 	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
 		err = dspio_set_uint_param(codec,
@@ -5282,6 +5324,18 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 /* End of control change helpers. */
+
+static void ca0132_alt_bass_redirection_xover_set(struct hda_codec *codec,
+		long idx)
+{
+	snd_hda_power_up(codec);
+
+	dspio_set_param(codec, 0x96, 0x20, SPEAKER_BASS_REDIRECT_XOVER_FREQ,
+			&(float_xbass_xover_lookup[idx]), sizeof(unsigned int));
+
+	snd_hda_power_down(codec);
+}
+
 /*
  * Below I've added controls to mess with the effect levels, I've only enabled
  * them on the Sound Blaster Z, but they would probably also work on the
@@ -5290,6 +5344,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
  */
 
 /* Sets DSP effect level from the sliders above the controls */
+
 static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid,
 			  const unsigned int *lookup, int idx)
 {
@@ -5335,8 +5390,12 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct ca0132_spec *spec = codec->spec;
 	long *valp = ucontrol->value.integer.value;
+	hda_nid_t nid = get_amp_nid(kcontrol);
 
-	*valp = spec->xbass_xover_freq;
+	if (nid == BASS_REDIRECTION_XOVER)
+		*valp = spec->bass_redirect_xover_freq;
+	else
+		*valp = spec->xbass_xover_freq;
 
 	return 0;
 }
@@ -5391,16 +5450,25 @@ static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol,
 	struct ca0132_spec *spec = codec->spec;
 	hda_nid_t nid = get_amp_nid(kcontrol);
 	long *valp = ucontrol->value.integer.value;
+	long *cur_val;
 	int idx;
 
+	if (nid == BASS_REDIRECTION_XOVER)
+		cur_val = &spec->bass_redirect_xover_freq;
+	else
+		cur_val = &spec->xbass_xover_freq;
+
 	/* any change? */
-	if (spec->xbass_xover_freq == *valp)
+	if (*cur_val == *valp)
 		return 0;
 
-	spec->xbass_xover_freq = *valp;
+	*cur_val = *valp;
 
 	idx = *valp;
-	ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
+	if (nid == BASS_REDIRECTION_XOVER)
+		ca0132_alt_bass_redirection_xover_set(codec, *cur_val);
+	else
+		ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
 
 	return 0;
 }
@@ -5968,6 +6036,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == BASS_REDIRECTION) {
+		*valp = spec->bass_redirection_val;
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -6054,6 +6127,14 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		changed = 0;
 	}
 
+	if (nid == BASS_REDIRECTION) {
+		spec->bass_redirection_val = *valp;
+		if (spec->cur_out_type == SURROUND_OUT)
+			ca0132_alt_surround_set_bass_redirection(codec, *valp);
+
+		changed = 0;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -6436,6 +6517,39 @@ static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Bass redirection redirects audio below the crossover frequency to the LFE
+ * channel on speakers that are set as not being full-range. On configurations
+ * without an LFE channel, it does nothing. Bass redirection seems to be the
+ * replacement for X-Bass on configurations with an LFE channel.
+ */
+static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection Crossover";
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0,
+				HDA_OUTPUT);
+
+	knew.tlv.c = NULL;
+	knew.info = ca0132_alt_xbass_xover_slider_info;
+	knew.get = ca0132_alt_xbass_xover_slider_ctl_get;
+	knew.put = ca0132_alt_xbass_xover_slider_put;
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION_XOVER,
+			snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_bass_redirection_switch(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection";
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO(namestr, BASS_REDIRECTION, 1,
+				HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION,
+			snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6749,6 +6863,12 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_rear_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_crossover(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_switch(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
@@ -8102,6 +8222,8 @@ static void ca0132_init_chip(struct hda_codec *codec)
 		spec->xbass_xover_freq = 8;
 		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
 			spec->fx_ctl_val[i] = effect_slider_defaults[i];
+
+		spec->bass_redirect_xover_freq = 8;
 	}
 
 	spec->voicefx_val = 0;

From ed8156c86f958744ba2088d57b1959b86fb32344 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:25 -0400
Subject: [PATCH 10/70] ALSA: hda/ca0132 - Remove surround output selection.

Remove the surround output selection and merge it with the speaker
output selection. Now that the extra commands that were being run on
surround output setting are known, there's no need to have it be
separate.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-7-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 151 +++++++++++------------------------
 1 file changed, 46 insertions(+), 105 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8ad2fc5ab30b..5743bdd7cc88 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -82,11 +82,10 @@ MODULE_FIRMWARE(R3DI_EFX_FILE);
 
 static const char *const dirstr[2] = { "Playback", "Capture" };
 
-#define NUM_OF_OUTPUTS 3
+#define NUM_OF_OUTPUTS 2
 enum {
 	SPEAKER_OUT,
 	HEADPHONE_OUT,
-	SURROUND_OUT
 };
 
 enum {
@@ -699,16 +698,6 @@ static const struct ca0132_alt_out_set alt_out_presets[] = {
 		    0x00000000, 0x00000000, 0x00000000,
 		    0x00000000 }
 	},
-	{ .name = "Surround",
-	  .commands = 8,
-	  .mids = { 0x96, 0x8F, 0x96, 0x96,
-		    0x96, 0x96, 0x96, 0x96 },
-	  .reqs = { 0x18, 0x01, 0x1F, 0x15,
-		    0x3A, 0x1A, 0x1B, 0x1C },
-	  .vals = { 0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000 }
-	}
 };
 
 /* Surround output channel count configuration structures. */
@@ -785,10 +774,6 @@ static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
 	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
 	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
 	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
 	}
 };
 
@@ -4440,42 +4425,6 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 			break;
 		}
 		break;
-	case SURROUND_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
-		}
-		break;
 	}
 }
 
@@ -4492,11 +4441,10 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 static int ca0132_alt_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp, outfx_set, i;
 	unsigned int pin_ctl;
 	int jack_present;
 	int auto_jack;
-	unsigned int i;
-	unsigned int tmp;
 	int err;
 	/* Default Headphone is rear headphone */
 	hda_nid_t headphone_nid = spec->out_pins[1];
@@ -4523,6 +4471,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	} else
 		spec->cur_out_type = spec->out_enum_val;
 
+	outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
+
 	/* Begin DSP output switch */
 	tmp = FLOAT_ONE;
 	err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp);
@@ -4536,6 +4486,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		codec_dbg(codec, "%s speaker\n", __func__);
 
 		/* disable headphone node */
+
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
 					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
@@ -4549,15 +4500,34 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
 
-		/* If PlayEnhancement is enabled, set different source */
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+		/* enable center/lfe out node */
+		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
+				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
+				pin_ctl | PIN_OUT);
+		/* Now set rear surround node as out. */
+		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
+				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
+				pin_ctl | PIN_OUT);
+
+		/*
+		 * Without PlayEnhancement being enabled, if we've got a 2.0
+		 * setup, set it to floating point eight to disable any DSP
+		 * processing effects.
+		 */
+		if (!outfx_set && spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+			tmp = FLOAT_EIGHT;
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+			tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
+
+		err = dspio_set_uint_param(codec, 0x80, 0x04, tmp);
+		if (err < 0)
+			goto exit;
+
 		break;
 	case HEADPHONE_OUT:
 		codec_dbg(codec, "%s hp\n", __func__);
-
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
@@ -4568,7 +4538,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 				pin_ctl & ~PIN_HP);
 
 		/* enable headphone, either front or rear */
-
 		if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp))
 			headphone_nid = spec->out_pins[2];
 		else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
@@ -4579,54 +4548,22 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_set_pin_ctl(codec, headphone_nid,
 				    pin_ctl | PIN_HP);
 
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+		if (outfx_set)
 			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
 		else
 			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
 		break;
-	case SURROUND_OUT:
-		codec_dbg(codec, "%s surround\n", __func__);
-
-		/* enable line out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-						pin_ctl | PIN_OUT);
-		/* Disable headphone out */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* Enable EAPD on line out */
-		snd_hda_codec_write(codec, spec->out_pins[0], 0,
-			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
-		/* enable center/lfe out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
-				    pin_ctl | PIN_OUT);
-		/* Now set rear surround node as out. */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
-				    pin_ctl | PIN_OUT);
-
-		tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
-		dspio_set_uint_param(codec, 0x80, 0x04, tmp);
-		break;
 	}
 	/*
-	 * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
-	 * With this set though, X_BASS cannot be enabled. So, if we have OutFX
-	 * enabled, we need to make sure X_BASS is off, otherwise everything
-	 * sounds all muffled. Running ca0132_effects_set with X_BASS as the
-	 * effect should sort this out.
+	 * If output effects are enabled, set the X-Bass effect value again to
+	 * make sure that it's properly enabled/disabled for speaker
+	 * configurations with an LFE channel.
 	 */
-	if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+	if (outfx_set)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
-	if (spec->cur_out_type == SURROUND_OUT)
+	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_surround_set_bass_redirection(codec,
 				spec->bass_redirection_val);
 	else
@@ -4646,7 +4583,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 			goto exit;
 	}
 
-	if (spec->cur_out_type == SURROUND_OUT)
+	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_set_full_range_speaker(codec);
 
 exit:
@@ -5054,7 +4991,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable)
 static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int on, tmp;
+	unsigned int on, tmp, channel_cfg;
 	int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	int err = 0;
 	int idx = nid - EFFECT_START_NID;
@@ -5067,8 +5004,12 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		/* if PE if off, turn off out effects. */
 		if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
 			val = 0;
-		if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
-			val = 0;
+		if (spec->cur_out_type == SPEAKER_OUT && nid == X_BASS) {
+			channel_cfg = spec->channel_cfg_val;
+			if (channel_cfg != SPEAKER_CHANNELS_2_0 &&
+					channel_cfg != SPEAKER_CHANNELS_4_0)
+				val = 0;
+		}
 	}
 
 	/* for in effect, qualify with CrystalVoice */
@@ -5799,7 +5740,7 @@ static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol,
 
 	spec->channel_cfg_val = sel;
 
-	if (spec->out_enum_val == SURROUND_OUT)
+	if (spec->out_enum_val == SPEAKER_OUT)
 		ca0132_alt_select_out(codec);
 
 	return 1;
@@ -6121,7 +6062,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 
 	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
 		spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp;
-		if (spec->cur_out_type == SURROUND_OUT)
+		if (spec->cur_out_type == SPEAKER_OUT)
 			ca0132_alt_set_full_range_speaker(codec);
 
 		changed = 0;
@@ -6129,7 +6070,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 
 	if (nid == BASS_REDIRECTION) {
 		spec->bass_redirection_val = *valp;
-		if (spec->cur_out_type == SURROUND_OUT)
+		if (spec->cur_out_type == SPEAKER_OUT)
 			ca0132_alt_surround_set_bass_redirection(codec, *valp);
 
 		changed = 0;

From 8e00dc7cedb37c43a2eeb2147d67409ae2e8abf5 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:26 -0400
Subject: [PATCH 11/70] ALSA: hda/ca0132 - Clean up ca0132_alt_out_select.

Remove the output structures that were in use before and instead set the
DSP commands line by line. Now that the commands use is known, it makes
the functionality more clear this way.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-8-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 142 ++++++++++++++---------------------
 1 file changed, 57 insertions(+), 85 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 5743bdd7cc88..39e333866be3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -83,6 +83,7 @@ MODULE_FIRMWARE(R3DI_EFX_FILE);
 static const char *const dirstr[2] = { "Playback", "Capture" };
 
 #define NUM_OF_OUTPUTS 2
+static const char *const out_type_str[2] = { "Speakers", "Headphone" };
 enum {
 	SPEAKER_OUT,
 	HEADPHONE_OUT,
@@ -667,39 +668,6 @@ enum speaker_tuning_reqs {
 	SPEAKER_TUNING_MUTE                     = 0x3a,
 };
 
-/* DSP command sequences for ca0132_alt_select_out */
-#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
-struct ca0132_alt_out_set {
-	char *name; /*preset name*/
-	unsigned char commands;
-	unsigned int mids[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int vals[ALT_OUT_SET_MAX_COMMANDS];
-};
-
-static const struct ca0132_alt_out_set alt_out_presets[] = {
-	{ .name = "Line Out",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
-	},
-	{ .name = "Headphone",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
-	},
-};
-
 /* Surround output channel count configuration structures. */
 #define SPEAKER_CHANNEL_CFG_COUNT 5
 enum {
@@ -4428,21 +4396,31 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 	}
 }
 
+static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid,
+		bool out_enable, bool hp_enable)
+{
+	unsigned int pin_ctl;
+
+	pin_ctl = snd_hda_codec_read(codec, nid, 0,
+			AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+
+	pin_ctl = hp_enable ? pin_ctl | PIN_HP_AMP : pin_ctl & ~PIN_HP_AMP;
+	pin_ctl = out_enable ? pin_ctl | PIN_OUT : pin_ctl & ~PIN_OUT;
+	snd_hda_set_pin_ctl(codec, nid, pin_ctl);
+}
+
 /*
  * This function behaves similarly to the ca0132_select_out funciton above,
  * except with a few differences. It adds the ability to select the current
  * output with an enumerated control "output source" if the auto detect
  * mute switch is set to off. If the auto detect mute switch is enabled, it
  * will detect either headphone or lineout(SPEAKER_OUT) from jack detection.
- * It also adds the ability to auto-detect the front headphone port. The only
- * way to select surround is to disable auto detect, and set Surround with the
- * enumerated control.
+ * It also adds the ability to auto-detect the front headphone port.
  */
 static int ca0132_alt_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp, outfx_set, i;
-	unsigned int pin_ctl;
+	unsigned int tmp, outfx_set;
 	int jack_present;
 	int auto_jack;
 	int err;
@@ -4473,9 +4451,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 
 	outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
 
-	/* Begin DSP output switch */
-	tmp = FLOAT_ONE;
-	err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp);
+	/* Begin DSP output switch, mute DSP volume. */
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_MUTE, FLOAT_ONE);
 	if (err < 0)
 		goto exit;
 
@@ -4485,31 +4462,18 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	case SPEAKER_OUT:
 		codec_dbg(codec, "%s speaker\n", __func__);
 
-		/* disable headphone node */
-
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* enable line-out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				    pin_ctl | PIN_OUT);
 		/* Enable EAPD */
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
 
-		/* enable center/lfe out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
-				pin_ctl | PIN_OUT);
-		/* Now set rear surround node as out. */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
-				pin_ctl | PIN_OUT);
+		/* Disable headphone node. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[1], 0, 0);
+		/* Set front L-R to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 1, 0);
+		/* Set Center/LFE to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 1, 0);
+		/* Set rear surround to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 1, 0);
 
 		/*
 		 * Without PlayEnhancement being enabled, if we've got a 2.0
@@ -4531,11 +4495,10 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
-		/* disable speaker*/
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				pin_ctl & ~PIN_HP);
+		/* Disable all speaker nodes. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 0, 0);
 
 		/* enable headphone, either front or rear */
 		if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp))
@@ -4543,15 +4506,15 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
 			headphone_nid = spec->out_pins[1];
 
-		pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, headphone_nid,
-				    pin_ctl | PIN_HP);
+		ca0132_set_out_node_pincfg(codec, headphone_nid, 1, 1);
 
 		if (outfx_set)
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
+
+		if (err < 0)
+			goto exit;
 		break;
 	}
 	/*
@@ -4563,29 +4526,38 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
+	/* Set speaker EQ bypass attenuation to 0. */
+	err = dspio_set_uint_param(codec, 0x8f, 0x01, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
+	/*
+	 * Although unused on all cards but the AE series, this is always set
+	 * to zero when setting the output.
+	 */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_USE_SPEAKER_EQ, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
 	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_surround_set_bass_redirection(codec,
 				spec->bass_redirection_val);
 	else
 		err = ca0132_alt_surround_set_bass_redirection(codec, 0);
 
+	/* Unmute DSP now that we're done with output selection. */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_MUTE, FLOAT_ZERO);
 	if (err < 0)
 		goto exit;
 
-	/* run through the output dsp commands for the selected output. */
-	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
-		err = dspio_set_uint_param(codec,
-		alt_out_presets[spec->cur_out_type].mids[i],
-		alt_out_presets[spec->cur_out_type].reqs[i],
-		alt_out_presets[spec->cur_out_type].vals[i]);
-
+	if (spec->cur_out_type == SPEAKER_OUT) {
+		err = ca0132_alt_set_full_range_speaker(codec);
 		if (err < 0)
 			goto exit;
 	}
 
-	if (spec->cur_out_type == SPEAKER_OUT)
-		err = ca0132_alt_set_full_range_speaker(codec);
-
 exit:
 	snd_hda_power_down_pm(codec);
 
@@ -5659,7 +5631,7 @@ static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
 	if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS)
 		uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1;
 	strcpy(uinfo->value.enumerated.name,
-			alt_out_presets[uinfo->value.enumerated.item].name);
+			out_type_str[uinfo->value.enumerated.item]);
 	return 0;
 }
 
@@ -5686,7 +5658,7 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
-		    sel, alt_out_presets[sel].name);
+		    sel, out_type_str[sel]);
 
 	spec->out_enum_val = sel;
 

From def3f0a5c7007c899a9a4d4034130e9d175d952d Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:27 -0400
Subject: [PATCH 12/70] ALSA: hda/ca0132 - Add quirk output selection
 structures.

Add structures containing the changes that need to happen on output
selection for each quirk. This should streamline the addition of new
quirks.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-9-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 338 +++++++++++++++++++++++++----------
 1 file changed, 242 insertions(+), 96 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 39e333866be3..ab84ea397552 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1256,6 +1256,164 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	{}
 };
 
+/* Output selection quirk info structures. */
+#define MAX_QUIRK_MMIO_GPIO_SET_VALS 3
+#define MAX_QUIRK_SCP_SET_VALS 2
+struct ca0132_alt_out_set_info {
+	unsigned int dac2port; /* ParamID 0x0d value. */
+
+	bool has_hda_gpio;
+	char hda_gpio_pin;
+	char hda_gpio_set;
+
+	unsigned int mmio_gpio_count;
+	char mmio_gpio_pin[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+	char mmio_gpio_set[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+
+	unsigned int scp_cmds_count;
+	unsigned int scp_cmd_mid[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_req[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_val[MAX_QUIRK_SCP_SET_VALS];
+
+	bool has_chipio_write;
+	unsigned int chipio_write_addr;
+	unsigned int chipio_write_data;
+};
+
+struct ca0132_alt_out_set_quirk_data {
+	int quirk_id;
+
+	bool has_headphone_gain;
+	bool is_ae_series;
+
+	struct ca0132_alt_out_set_info out_set_info[NUM_OF_OUTPUTS];
+};
+
+static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = {
+	{ .quirk_id = QUIRK_R3DI,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 1,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 0,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_R3D,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_SBZ,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x18,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false, },
+		/* Headphones. */
+		{ .dac2port         = 0x12,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_ZXR,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_AE5,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0xa4,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		},
+		/* Headphones. */
+		{ .dac2port          = 0xa1,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		} },
+	}
+};
+
 /*
  * CA0132 codec access
  */
@@ -3513,26 +3671,6 @@ static void r3di_gpio_mic_set(struct hda_codec *codec,
 			    AC_VERB_SET_GPIO_DATA, cur_gpio);
 }
 
-static void r3di_gpio_out_set(struct hda_codec *codec,
-		enum r3di_out_select cur_out)
-{
-	unsigned int cur_gpio;
-
-	/* Get the current GPIO Data setup */
-	cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0);
-
-	switch (cur_out) {
-	case R3DI_HEADPHONE_OUT:
-		cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT);
-		break;
-	case R3DI_LINE_OUT:
-		cur_gpio |= (1 << R3DI_OUT_SELECT_BIT);
-		break;
-	}
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_DATA, cur_gpio);
-}
-
 static void r3di_gpio_dsp_status_set(struct hda_codec *codec,
 		enum r3di_dsp_status dsp_status)
 {
@@ -4314,88 +4452,95 @@ static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec,
  * These are the commands needed to setup output on each of the different card
  * types.
  */
-static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+static void ca0132_alt_select_out_get_quirk_data(struct hda_codec *codec,
+		const struct ca0132_alt_out_set_quirk_data **quirk_data)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp;
+	int quirk = ca0132_quirk(spec);
+	unsigned int i;
 
-	switch (spec->cur_out_type) {
-	case SPEAKER_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	*quirk_data = NULL;
+	for (i = 0; i < ARRAY_SIZE(quirk_out_set_data); i++) {
+		if (quirk_out_set_data[i].quirk_id == quirk) {
+			*quirk_data = &quirk_out_set_data[i];
+			return;
 		}
-		break;
-	case HEADPHONE_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, true);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, false);
-			chipio_set_control_param(codec, 0x0d, 0x12);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, false);
-			ca0113_mmio_gpio_set(codec, 3, false);
-			ca0113_mmio_gpio_set(codec, 5, true);
-			zxr_headphone_gain_set(codec, spec->zxr_gain_set);
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			ca0113_mmio_gpio_set(codec, 0x1, false);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec,
-					spec->ae5_headphone_gain_val);
-			tmp = FLOAT_ONE;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa1);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
-		}
-		break;
 	}
 }
 
+static int ca0132_alt_select_out_quirk_set(struct hda_codec *codec)
+{
+	const struct ca0132_alt_out_set_quirk_data *quirk_data;
+	const struct ca0132_alt_out_set_info *out_info;
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, gpio_data;
+	int err;
+
+	ca0132_alt_select_out_get_quirk_data(codec, &quirk_data);
+	if (!quirk_data)
+		return 0;
+
+	out_info = &quirk_data->out_set_info[spec->cur_out_type];
+	if (quirk_data->is_ae_series)
+		ae5_mmio_select_out(codec);
+
+	if (out_info->has_hda_gpio) {
+		gpio_data = snd_hda_codec_read(codec, codec->core.afg, 0,
+				AC_VERB_GET_GPIO_DATA, 0);
+
+		if (out_info->hda_gpio_set)
+			gpio_data |= (1 << out_info->hda_gpio_pin);
+		else
+			gpio_data &= ~(1 << out_info->hda_gpio_pin);
+
+		snd_hda_codec_write(codec, codec->core.afg, 0,
+				    AC_VERB_SET_GPIO_DATA, gpio_data);
+	}
+
+	if (out_info->mmio_gpio_count) {
+		for (i = 0; i < out_info->mmio_gpio_count; i++) {
+			ca0113_mmio_gpio_set(codec, out_info->mmio_gpio_pin[i],
+					out_info->mmio_gpio_set[i]);
+		}
+	}
+
+	if (out_info->scp_cmds_count) {
+		for (i = 0; i < out_info->scp_cmds_count; i++) {
+			err = dspio_set_uint_param(codec,
+					out_info->scp_cmd_mid[i],
+					out_info->scp_cmd_req[i],
+					out_info->scp_cmd_val[i]);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	chipio_set_control_param(codec, 0x0d, out_info->dac2port);
+
+	if (out_info->has_chipio_write) {
+		chipio_write(codec, out_info->chipio_write_addr,
+				out_info->chipio_write_data);
+	}
+
+	if (quirk_data->has_headphone_gain) {
+		if (spec->cur_out_type != HEADPHONE_OUT) {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec, 2);
+			else
+				zxr_headphone_gain_set(codec, 0);
+		} else {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec,
+						spec->ae5_headphone_gain_val);
+			else
+				zxr_headphone_gain_set(codec,
+						spec->zxr_gain_set);
+		}
+	}
+
+	return 0;
+}
+
 static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid,
 		bool out_enable, bool hp_enable)
 {
@@ -4456,7 +4601,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	if (err < 0)
 		goto exit;
 
-	ca0132_alt_select_out_quirk_handler(codec);
+	if (ca0132_alt_select_out_quirk_set(codec) < 0)
+		goto exit;
 
 	switch (spec->cur_out_type) {
 	case SPEAKER_OUT:

From b7a8b9e8e797236a62716118c513e13a2670fd07 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:28 -0400
Subject: [PATCH 13/70] ALSA: hda/ca0132 - Fix Recon3D Center/LFE output.

Properly set the GPIO pin to un-mute the Center/LFE channel on the
Recon3D.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-10-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ab84ea397552..138403fd1639 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -7819,6 +7819,12 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 	if (ca0132_quirk(spec) == QUIRK_R3DI)
 		r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
 
+	/* Disable mute on Center/LFE. */
+	if (ca0132_quirk(spec) == QUIRK_R3D) {
+		ca0113_mmio_gpio_set(codec, 2, false);
+		ca0113_mmio_gpio_set(codec, 4, true);
+	}
+
 	/* Setup effect defaults */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
 	for (idx = 0; idx < num_fx; idx++) {

From 620f08eea6d6961b789af3fa3ea86725c8c93ece Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:29 -0400
Subject: [PATCH 14/70] ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster
 AE-7.

Add a new PCI subsystem ID for the SoundBlaster AE-7 card.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-11-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 138403fd1639..284f63dc2749 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1134,6 +1134,7 @@ enum {
 	QUIRK_R3DI,
 	QUIRK_R3D,
 	QUIRK_AE5,
+	QUIRK_AE7,
 };
 
 #ifdef CONFIG_PCI
@@ -1253,6 +1254,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+	SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
 	{}
 };
 

From a35e37a3a2c5fc9a6dcf1e0445e8472cffac37c1 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:30 -0400
Subject: [PATCH 15/70] ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg.

Add AE-7 pincfg, based on the values set within Windows.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-12-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 284f63dc2749..6791aaf18e63 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1238,6 +1238,20 @@ static const struct hda_pintbl r3di_pincfgs[] = {
 	{}
 };
 
+static const struct hda_pintbl ae7_pincfgs[] = {
+	{ 0x0b, 0x01017010 },
+	{ 0x0c, 0x014510f0 },
+	{ 0x0d, 0x414510f0 },
+	{ 0x0e, 0x01c520f0 },
+	{ 0x0f, 0x01017114 },
+	{ 0x10, 0x01017011 },
+	{ 0x11, 0x018170ff },
+	{ 0x12, 0x01a170f0 },
+	{ 0x13, 0x908700f0 },
+	{ 0x18, 0x500000f0 },
+	{}
+};
+
 static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4),
 	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
@@ -9105,6 +9119,10 @@ static void ca0132_config(struct hda_codec *codec)
 		codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
 		snd_hda_apply_pincfgs(codec, ae5_pincfgs);
 		break;
+	case QUIRK_AE7:
+		codec_dbg(codec, "%s: QUIRK_AE7 applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, ae7_pincfgs);
+		break;
 	default:
 		break;
 	}
@@ -9186,6 +9204,7 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->dig_in = 0x09;
 		break;
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
 		spec->out_pins[1] = 0x11; /* Rear headphone out */

From 76d257d67f4104bd3089be1a486047e3f3b40c24 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:31 -0400
Subject: [PATCH 16/70] ALSA: hda/ca0132 - Set AE-7 bools and select mixer.

Set the boolean values used for desktop cards, and select the desktop
mixer.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-13-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 6791aaf18e63..bd5d4f0bd6f5 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -9403,6 +9403,10 @@ static int patch_ca0132(struct hda_codec *codec)
 		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
 		break;
+	case QUIRK_AE7:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Sound Blaster AE-7");
+		break;
 	default:
 		spec->mixers[0] = ca0132_mixer;
 		break;
@@ -9413,6 +9417,7 @@ static int patch_ca0132(struct hda_codec *codec)
 	case QUIRK_SBZ:
 	case QUIRK_R3D:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 	case QUIRK_ZXR:
 		spec->use_alt_controls = true;
 		spec->use_alt_functions = true;

From 4e356d56df9dbd4631ce2e40a5fb491829f4e8ce Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:32 -0400
Subject: [PATCH 17/70] ALSA: hda/ca0132 - Add ca0132_mmio_init data for
 SoundBlaster AE-7.

Modify the AE-5 ca0132_mmio_init function to add AE-7 specific writes.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-14-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 20 +++++++++++++++++++-
 1 file changed, 19 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index bd5d4f0bd6f5..10aaa4806946 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8748,8 +8748,26 @@ static void ca0132_mmio_init_ae5(struct hda_codec *codec)
 	data = ca0113_mmio_init_data_ae5;
 	count = ARRAY_SIZE(ca0113_mmio_init_data_ae5);
 
-	for (i = 0; i < count; i++)
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		writel(0x00000680, spec->mem_base + 0x1c);
+		writel(0x00880680, spec->mem_base + 0x1c);
+	}
+
+	for (i = 0; i < count; i++) {
+		/*
+		 * AE-7 shares all writes with the AE-5, except that it writes
+		 * a different value to 0x20c.
+		 */
+		if (i == 21 && ca0132_quirk(spec) == QUIRK_AE7) {
+			writel(0x00800001, spec->mem_base + addr[i]);
+			continue;
+		}
+
 		writel(data[i], spec->mem_base + addr[i]);
+	}
+
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		writel(0x00880680, spec->mem_base + 0x1c);
 }
 
 static void ca0132_mmio_init(struct hda_codec *codec)

From 77bdbae90445771f76510e8c6a4ac0b722a19472 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:33 -0400
Subject: [PATCH 18/70] ALSA: hda/ca0132 - Add pre-init function for
 SoundBlaster AE-7.

Add pre DSP initialization function for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-15-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 70 ++++++++++++++++++++++++++++--------
 1 file changed, 55 insertions(+), 15 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 10aaa4806946..cd46112c827e 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8786,6 +8786,16 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 	}
 }
 
+static const unsigned int ca0132_ae5_register_set_addresses[] = {
+	0x304, 0x304, 0x304, 0x304, 0x100, 0x304, 0x100, 0x304, 0x100, 0x304,
+	0x100, 0x304, 0x86c, 0x800, 0x86c, 0x800, 0x804
+};
+
+static const unsigned char ca0132_ae5_register_set_data[] = {
+	0x0f, 0x0e, 0x1f, 0x0c, 0x3f, 0x08, 0x7f, 0x00, 0xff, 0x00, 0x6b,
+	0x01, 0x6b, 0x57
+};
+
 /*
  * This function writes to some SFR's, does some region2 writes, and then
  * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
@@ -8794,6 +8804,18 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 static void ae5_register_set(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int count = ARRAY_SIZE(ca0132_ae5_register_set_addresses);
+	const unsigned int *addr = ca0132_ae5_register_set_addresses;
+	const unsigned char *data = ca0132_ae5_register_set_data;
+	unsigned int i, cur_addr;
+	unsigned char tmp[3];
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+	}
 
 	chipio_8051_write_direct(codec, 0x93, 0x10);
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
@@ -8801,25 +8823,43 @@ static void ae5_register_set(struct hda_codec *codec)
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
 			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
 
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0e, spec->mem_base + 0x100);
-	writeb(0x1f, spec->mem_base + 0x304);
-	writeb(0x0c, spec->mem_base + 0x100);
-	writeb(0x3f, spec->mem_base + 0x304);
-	writeb(0x08, spec->mem_base + 0x100);
-	writeb(0x7f, spec->mem_base + 0x304);
-	writeb(0x00, spec->mem_base + 0x100);
-	writeb(0xff, spec->mem_base + 0x304);
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		tmp[0] = 0x03;
+		tmp[1] = 0x03;
+		tmp[2] = 0x07;
+	} else {
+		tmp[0] = 0x0f;
+		tmp[1] = 0x0f;
+		tmp[2] = 0x0f;
+	}
 
-	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	for (i = cur_addr = 0; i < 3; i++, cur_addr++)
+		writeb(tmp[i], spec->mem_base + addr[cur_addr]);
+
+	/*
+	 * First writes are in single bytes, final are in 4 bytes. So, we use
+	 * writeb, then writel.
+	 */
+	for (i = 0; cur_addr < 12; i++, cur_addr++)
+		writeb(data[i], spec->mem_base + addr[cur_addr]);
+
+	for (; cur_addr < count; i++, cur_addr++)
+		writel(data[i], spec->mem_base + addr[cur_addr]);
+
+	writel(0x00800001, spec->mem_base + 0x20c);
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+		ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+	} else {
+		ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	}
 
 	chipio_8051_write_direct(codec, 0x90, 0x00);
 	chipio_8051_write_direct(codec, 0x90, 0x10);
 
-	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
 
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
 
@@ -8918,7 +8958,7 @@ static int ca0132_init(struct hda_codec *codec)
 
 	snd_hda_power_up_pm(codec);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
+	if (ca0132_quirk(spec) == QUIRK_AE5 || ca0132_quirk(spec) == QUIRK_AE7)
 		ae5_register_set(codec);
 
 	ca0132_init_unsol(codec);

From cfa736e2f02d768dad1b611abdb85fe6c10d8000 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:34 -0400
Subject: [PATCH 19/70] ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7.

Add initialization data for the SoundBlaster AE-7 card.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-16-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index cd46112c827e..dc1eb9bfcc5e 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -3582,6 +3582,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
 	switch (ca0132_quirk(spec)) {
 	case QUIRK_SBZ:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
 		snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
 		snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -8911,6 +8912,19 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
 		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
 		break;
+	case QUIRK_AE7:
+		ca0132_gpio_init(codec);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		chipio_write(codec, 0x18b008, 0x000000f8);
+		chipio_write(codec, 0x18b008, 0x000000f0);
+		chipio_write(codec, 0x18b030, 0x00000020);
+		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+		break;
 	case QUIRK_ZXR:
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);

From e5b21888882bb340bf0fbcbcea909b09eef2933f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:35 -0400
Subject: [PATCH 20/70] ALSA: hda/ca0132 - Add DSP setup functions for AE-7.

Add DSP setup functions for the Sound Blaster AE-7 post DSP download.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-17-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 290 +++++++++++++++++++++++++++++++++++
 1 file changed, 290 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index dc1eb9bfcc5e..8519119ef7a6 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -7378,6 +7378,7 @@ static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
 		values = zxr_default_delay_values;
 		break;
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		values = ae5_default_delay_values;
 		break;
 	default:
@@ -7551,6 +7552,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
 		switch (ca0132_quirk(spec)) {
 		case QUIRK_SBZ:
 		case QUIRK_AE5:
+		case QUIRK_AE7:
 			tmp = 0x00000003;
 			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
 			tmp = 0x00000000;
@@ -7760,6 +7762,206 @@ static void ae5_post_dsp_startup_data(struct hda_codec *codec)
 	mutex_unlock(&spec->chipio_mutex);
 }
 
+static const unsigned int ae7_port_set_data[] = {
+	0x0001e0c0, 0x0001e1c1, 0x0001e4c2, 0x0001e5c3, 0x0001e2c4, 0x0001e3c5,
+	0x0001e8c6, 0x0001e9c7, 0x0001ecc8, 0x0001edc9, 0x0001eaca, 0x0001ebcb
+};
+
+static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, count, addr;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	count = ARRAY_SIZE(ae7_port_set_data);
+	addr = 0x190030;
+	for (i = 0; i < count; i++) {
+		chipio_write_no_mutex(codec, addr, ae7_port_set_data[i]);
+
+		/* Addresses are incremented by 4-bytes. */
+		addr += 0x04;
+	}
+
+	/*
+	 * Port setting always ends with a write of 0x1 to address 0x19042c.
+	 */
+	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x0d, 0x40);
+	ca0113_mmio_command_set(codec, 0x48, 0x17, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x19, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x11, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x12, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x13, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x14, 0x7f);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+
+	chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+	chipio_set_stream_control(codec, 0x18, 1);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_pll_setup(struct hda_codec *codec)
+{
+	const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 };
+	const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d };
+	unsigned int i;
+
+	for (i = 0; i < ARRAY_SIZE(addr); i++) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]);
+	}
+}
+
+static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11,
+					0x12, 0x13, 0x14 };
+	const unsigned int data[]   = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff,
+					0xff, 0xff, 0x7f };
+	unsigned int i;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+	chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	for (i = 0; i < ARRAY_SIZE(target); i++)
+		ca0113_mmio_command_set(codec, 0x48, target[i], data[i]);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x21, 0x64, 0x56);
+	chipio_set_stream_channels(codec, 0x21, 2);
+	chipio_set_conn_rate_no_mutex(codec, 0x56, SR_8_000);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	/*
+	 * In the 8051's memory, this param is referred to as 'n2sid', which I
+	 * believe is 'node to streamID'. It seems to be a way to assign a
+	 * stream to a given HDA node.
+	 */
+	chipio_set_control_param_no_mutex(codec, 0x20, 0x21);
+
+	chipio_write_no_mutex(codec, 0x18b038, 0x00000088);
+
+	/*
+	 * Now, at this point on Windows, an actual stream is setup and
+	 * seemingly sends data to the HDA node 0x09, which is the digital
+	 * audio input node. This is left out here, because obviously I don't
+	 * know what data is being sent. Interestingly, the AE-5 seems to go
+	 * through the motions of getting here and never actually takes this
+	 * step, but the AE-7 does.
+	 */
+
+	ca0113_mmio_gpio_set(codec, 0, 1);
+	ca0113_mmio_gpio_set(codec, 1, 1);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	chipio_write_no_mutex(codec, 0x18b03c, 0x00000000);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+
+	/*
+	 * Runs again, this has been repeated a few times, but I'm just
+	 * following what the Windows driver does.
+	 */
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+/*
+ * The Windows driver has commands that seem to setup ASI, which I believe to
+ * be some sort of audio serial interface. My current speculation is that it's
+ * related to communicating with the new DAC.
+ */
+static void ae7_post_dsp_asi_setup(struct hda_codec *codec)
+{
+	chipio_8051_write_direct(codec, 0x93, 0x10);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	chipio_set_control_param(codec, 3, 3);
+	chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+	snd_hda_codec_write(codec, 0x17, 0, 0x794, 0x00);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+
+	ae7_post_dsp_pll_setup(codec);
+	ae7_post_dsp_asi_stream_setup(codec);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	ae7_post_dsp_asi_setup_ports(codec);
+}
+
 /*
  * Setup default parameters for DSP
  */
@@ -7983,6 +8185,91 @@ static void ae5_setup_defaults(struct hda_codec *codec)
 	ca0132_alt_create_dummy_stream(codec);
 }
 
+/*
+ * Setup default parameters for the Sound Blaster AE-7 DSP.
+ */
+static void ae7_setup_defaults(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int num_fx;
+	int idx, i;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return;
+
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
+	ae7_post_dsp_setup_ports(codec);
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_LEFT_INVERT, tmp);
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_RIGHT_INVERT, tmp);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	/* New, unknown SCP req's */
+	dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+	dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+	ca0113_mmio_gpio_set(codec, 0, false);
+
+	/* Internal loopback off */
+	tmp = FLOAT_ONE;
+	dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+	dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+	/*remove DSP headroom*/
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+	/* set WUH source */
+	tmp = FLOAT_TWO;
+	dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+	chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+	/* Set speaker source? */
+	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+	ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+	/*
+	 * This is the second time we've called this, but this is seemingly
+	 * what Windows does.
+	 */
+	ca0132_alt_init_analog_mics(codec);
+
+	ae7_post_dsp_asi_setup(codec);
+
+	/*
+	 * Not sure why, but these are both set to 1. They're only set to 0
+	 * upon shutdown.
+	 */
+	ca0113_mmio_gpio_set(codec, 0, true);
+	ca0113_mmio_gpio_set(codec, 1, true);
+
+	/* Volume control related. */
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x04);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x04);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x80);
+
+	/* out, in effects + voicefx */
+	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+	for (idx = 0; idx < num_fx; idx++) {
+		for (i = 0; i <= ca0132_effects[idx].params; i++) {
+			dspio_set_uint_param(codec,
+					ca0132_effects[idx].mid,
+					ca0132_effects[idx].reqs[i],
+					ca0132_effects[idx].def_vals[i]);
+		}
+	}
+
+	ca0132_alt_init_speaker_tuning(codec);
+
+	ca0132_alt_create_dummy_stream(codec);
+}
+
 /*
  * Initialization of flags in chip
  */
@@ -9000,6 +9287,9 @@ static int ca0132_init(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_setup_defaults(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_setup_defaults(codec);
+		break;
 	default:
 		ca0132_setup_defaults(codec);
 		ca0132_init_analog_mic2(codec);

From 91b94a933f28b0b819224bc638866b0844317776 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:36 -0400
Subject: [PATCH 21/70] ALSA: hda/ca0132 - Add output selection for
 SoundBlaster AE-7.

Add output selection quirk table information for SoundBlaster AE-7, and
slightly modify the AE-5's ca0113 command table to accommodate the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-18-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 88 ++++++++++++++++++++++++++++--------
 1 file changed, 68 insertions(+), 20 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8519119ef7a6..dcc8d29d934c 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -727,22 +727,29 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
 };
 
 /* Values for ca0113_mmio_command_set for selecting output. */
-#define AE5_CA0113_OUT_SET_COMMANDS 6
-struct ae5_ca0113_output_set {
-	unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+#define AE_CA0113_OUT_SET_COMMANDS 6
+struct ae_ca0113_output_set {
+	unsigned int group[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int target[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int vals[NUM_OF_OUTPUTS][AE_CA0113_OUT_SET_COMMANDS];
 };
 
-static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
-	}
+static const struct ae_ca0113_output_set ae5_ca0113_output_presets = {
+	.group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals =   { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } },
+};
+
+static const struct ae_ca0113_output_set ae7_ca0113_output_presets = {
+	.group  = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals   = { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x02, 0x00 } },
 };
 
 /* ae5 ca0113 command sequences to set headphone gain levels. */
@@ -1427,6 +1434,41 @@ static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = {
 		  .chipio_write_addr = 0x0018b03c,
 		  .chipio_write_data = 0x00000012
 		} },
+	},
+	{ .quirk_id = QUIRK_AE7,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000000
+		},
+		/* Headphones. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000010
+		} },
 	}
 };
 
@@ -4383,18 +4425,24 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
 static void ae5_mmio_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	const struct ae_ca0113_output_set *out_cmds;
 	unsigned int i;
 
-	for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
-		ca0113_mmio_command_set(codec,
-			ae5_ca0113_output_presets[spec->cur_out_type].group[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].target[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		out_cmds = &ae5_ca0113_output_presets;
+	else
+		out_cmds = &ae7_ca0113_output_presets;
+
+	for (i = 0; i < AE_CA0113_OUT_SET_COMMANDS; i++)
+		ca0113_mmio_command_set(codec, out_cmds->group[i],
+				out_cmds->target[i],
+				out_cmds->vals[spec->cur_out_type][i]);
 }
 
 static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	int quirk = ca0132_quirk(spec);
 	unsigned int tmp;
 	int err;
 
@@ -4426,7 +4474,7 @@ static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 	 * Only the AE series cards set this value when setting full-range,
 	 * and it's always 1.0f.
 	 */
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
+	if (quirk == QUIRK_AE5 || quirk == QUIRK_AE7) {
 		err = dspio_set_uint_param(codec, 0x96,
 				SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
 		if (err < 0)

From ed93f9750c6c2ed371347d0aac3dcd31cb9cf256 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:37 -0400
Subject: [PATCH 22/70] ALSA: hda/ca0132 - Add AE-7 microphone selection
 commands.

Add AE-7 quirk data for setting of microphone. The AE-7 has no front
panel connector, so only rear-mic/line-in have new commands.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-19-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 22 +++++++++++++++++++++-
 1 file changed, 21 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index dcc8d29d934c..8c6e38734489 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -4997,6 +4997,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			tmp = FLOAT_THREE;
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+			tmp = FLOAT_THREE;
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -5042,6 +5051,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		case QUIRK_AE5:
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			break;
 		}
@@ -5051,7 +5068,10 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		if (ca0132_quirk(spec) == QUIRK_R3DI)
 			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
 
-		tmp = FLOAT_ZERO;
+		if (ca0132_quirk(spec) == QUIRK_AE7)
+			tmp = FLOAT_THREE;
+		else
+			tmp = FLOAT_ZERO;
 		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
 		switch (ca0132_quirk(spec)) {

From 24a28eaeb23b2ff7791350bb44875ca4dc3bef3f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:38 -0400
Subject: [PATCH 23/70] ALSA: hda/ca0132 - Add AE-7 custom controls.

Add headphone gain and DAC filter controls, which use the same commands
as the AE-5. Also, change input source enumerated control item count to
exclude front microphone.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-20-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++++----
 1 file changed, 16 insertions(+), 4 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8c6e38734489..52f6d3740e0a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -5839,6 +5839,13 @@ static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol,
 	int sel = ucontrol->value.enumerated.item[0];
 	unsigned int items = IN_SRC_NUM_OF_INPUTS;
 
+	/*
+	 * The AE-7 has no front microphone, so limit items to 2: rear mic and
+	 * line-in.
+	 */
+	if (ca0132_quirk(spec) == QUIRK_AE7)
+		items = 2;
+
 	if (sel >= items)
 		return 0;
 
@@ -7029,20 +7036,25 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		}
 	}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_AE5:
+	case QUIRK_AE7:
 		err = ae5_add_headphone_gain_enum(codec);
 		if (err < 0)
 			return err;
 		err = ae5_add_sound_filter_enum(codec);
 		if (err < 0)
 			return err;
-	}
-
-	if (ca0132_quirk(spec) == QUIRK_ZXR) {
+		break;
+	case QUIRK_ZXR:
 		err = zxr_add_headphone_gain_switch(codec);
 		if (err < 0)
 			return err;
+		break;
+	default:
+		break;
 	}
+
 #ifdef ENABLE_TUNING_CONTROLS
 	add_tuning_ctls(codec);
 #endif

From 685a04a53747d07d22f5139c04ae35745200a91d Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:39 -0400
Subject: [PATCH 24/70] ALSA: hda/ca0132 - Add AE-7 exit commands.

Add exit commands for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-21-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 29 +++++++++++++++++++++++++++++
 1 file changed, 29 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 52f6d3740e0a..9779978e4bc7 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8838,6 +8838,32 @@ static void ae5_exit_chip(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
 }
 
+static void ae7_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_source_dest(codec, 0x21, 0xc8, 0xc8);
+	chipio_set_stream_channels(codec, 0x21, 0);
+	chipio_set_control_param(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	chipio_set_control_param(codec, 0x20, 0x01);
+
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_control(codec, 0x0c, 0);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+	snd_hda_codec_write(codec, 0x15, 0, 0x724, 0x83);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x00);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 1, false);
+	ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+}
+
 static void zxr_exit_chip(struct hda_codec *codec)
 {
 	chipio_set_stream_control(codec, 0x03, 0);
@@ -9457,6 +9483,9 @@ static void ca0132_free(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_exit_chip(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_exit_chip(codec);
+		break;
 	case QUIRK_R3DI:
 		r3di_gpio_shutdown(codec);
 		break;

From 931522b90813a6f1c9c673ec9a0eb985b39554dc Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:16 -0500
Subject: [PATCH 25/70] ALSA: core: pcm: simplify locking for timers

Fix sparse warning:

sound/core/pcm.c:999:9: warning: context imbalance in
'snd_pcm_detach_substream' - different lock contexts for basic block

There's no real reason to test the same thing twice, and it's simpler
have linear sequences.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index b6d2331a82f7..be5714f1bb58 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -991,11 +991,13 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
 		       PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
 	kfree(runtime->hw_constraints.rules);
 	/* Avoid concurrent access to runtime via PCM timer interface */
-	if (substream->timer)
+	if (substream->timer) {
 		spin_lock_irq(&substream->timer->lock);
-	substream->runtime = NULL;
-	if (substream->timer)
+		substream->runtime = NULL;
 		spin_unlock_irq(&substream->timer->lock);
+	} else {
+		substream->runtime = NULL;
+	}
 	kfree(runtime);
 	put_pid(substream->pid);
 	substream->pid = NULL;

From b1c14124507b86ab554be75ed5fcb2e50bb33c6c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:17 -0500
Subject: [PATCH 26/70] ALSA: core: memalloc: fix fallthrough position

Fix cppcheck, the fallthrough only makes sense within the conditional
block

sound/core/memalloc.c:161:3: style:inconclusive: Statements following
return, break, continue, goto or throw will never be
executed. [unreachableCode]
  fallthrough;
  ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/memalloc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index ad74ea9cbff5..0aeeb6244ff6 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -157,8 +157,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		 * so if we fail to malloc, try to fetch memory traditionally.
 		 */
 		dmab->dev.type = SNDRV_DMA_TYPE_DEV;
-#endif /* CONFIG_GENERIC_ALLOCATOR */
 		fallthrough;
+#endif /* CONFIG_GENERIC_ALLOCATOR */
 	case SNDRV_DMA_TYPE_DEV:
 	case SNDRV_DMA_TYPE_DEV_UC:
 		snd_malloc_dev_pages(dmab, size);

From b658cbabf8e3e41816f43820f2b03f498e5556d2 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:18 -0500
Subject: [PATCH 27/70] ALSA: core: pcm_memory: dereference pointer after NULL
 checks

Fix cppcheck warnings:

sound/core/pcm_memory.c:380:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
 struct snd_card *card = substream->pcm->card;
                         ^
sound/core/pcm_memory.c:384:6: note: Assuming that condition
'!substream' is not redundant
 if (PCM_RUNTIME_CHECK(substream))
     ^
sound/core/pcm_memory.c:380:26: note: Null pointer dereference
 struct snd_card *card = substream->pcm->card;
                         ^
sound/core/pcm_memory.c:433:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
 struct snd_card *card = substream->pcm->card;
                         ^
sound/core/pcm_memory.c:436:6: note: Assuming that condition
'!substream' is not redundant
 if (PCM_RUNTIME_CHECK(substream))
     ^
sound/core/pcm_memory.c:433:26: note: Null pointer dereference
 struct snd_card *card = substream->pcm->card;
                         ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_memory.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 1bf6a3d9e0c2..4f03ba8ed0ae 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -377,7 +377,7 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne
  */
 int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
 {
-	struct snd_card *card = substream->pcm->card;
+	struct snd_card *card;
 	struct snd_pcm_runtime *runtime;
 	struct snd_dma_buffer *dmab = NULL;
 
@@ -387,6 +387,7 @@ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
 		       SNDRV_DMA_TYPE_UNKNOWN))
 		return -EINVAL;
 	runtime = substream->runtime;
+	card = substream->pcm->card;
 
 	if (runtime->dma_buffer_p) {
 		/* perphaps, we might free the large DMA memory region

From 63632563c0dfe036215dcba222a0f899d35f7be6 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:19 -0500
Subject: [PATCH 28/70] ALSA: core: timer: remove redundant assignment

Cppcheck complains about a possible NULL pointer dereference but it
actually looks like the NULL assignment is not needed (same loop is
used in other parts of the file without it).

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/timer.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/core/timer.c b/sound/core/timer.c
index d9f85f2d66a3..1349f5b5be09 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -173,7 +173,7 @@ EXPORT_SYMBOL(snd_timer_instance_free);
  */
 static struct snd_timer *snd_timer_find(struct snd_timer_id *tid)
 {
-	struct snd_timer *timer = NULL;
+	struct snd_timer *timer;
 
 	list_for_each_entry(timer, &snd_timer_list, device_list) {
 		if (timer->tmr_class != tid->dev_class)

From 3bcf8eeb7d979402d3db96fb58bed456a3c66668 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:20 -0500
Subject: [PATCH 29/70] ALSA: core: timer: clarify operator precedence

fix cppcheck warning:

sound/core/timer.c:1286:9: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
        ? "running" : "stopped");
        ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/timer.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/core/timer.c b/sound/core/timer.c
index 1349f5b5be09..b915d39e7acb 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1281,8 +1281,8 @@ static void snd_timer_proc_read(struct snd_info_entry *entry,
 		list_for_each_entry(ti, &timer->open_list_head, open_list)
 			snd_iprintf(buffer, "  Client %s : %s\n",
 				    ti->owner ? ti->owner : "unknown",
-				    ti->flags & (SNDRV_TIMER_IFLG_START |
-						 SNDRV_TIMER_IFLG_RUNNING)
+				    (ti->flags & (SNDRV_TIMER_IFLG_START |
+						  SNDRV_TIMER_IFLG_RUNNING))
 				    ? "running" : "stopped");
 	}
 	mutex_unlock(&register_mutex);

From cd91fd9f0af5453e66bda9984916ca67686689cf Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:21 -0500
Subject: [PATCH 30/70] ALSA: compress_offload: dereference after checking for
 NULL pointer

Fix cppcheck warning and only dereference once the initial checks are
done:

sound/core/compress_offload.c:516:38: warning: Either the condition
'!stream' is redundant or there is possible null pointer dereference:
stream. [nullPointerRedundantCheck]
 struct snd_compr_runtime *runtime = stream->runtime;
                                     ^
sound/core/compress_offload.c:518:17: note: Assuming that condition
'!stream' is not redundant
if (snd_BUG_ON(!(stream) || !(stream)->runtime))
                ^
sound/core/compress_offload.c:516:38: note: Null pointer dereference
 struct snd_compr_runtime *runtime = stream->runtime;
                                     ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200902212133.30964-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 0e53f6f31916..e3eb314acb10 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -513,10 +513,11 @@ EXPORT_SYMBOL(snd_compr_malloc_pages);
 
 int snd_compr_free_pages(struct snd_compr_stream *stream)
 {
-	struct snd_compr_runtime *runtime = stream->runtime;
+	struct snd_compr_runtime *runtime;
 
 	if (snd_BUG_ON(!(stream) || !(stream)->runtime))
 		return -EINVAL;
+	runtime = stream->runtime;
 	if (runtime->dma_area == NULL)
 		return 0;
 	if (runtime->dma_buffer_p != &stream->dma_buffer) {

From 9725ce3949a7b3eebf003cb69fa0d7d821179233 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:22 -0500
Subject: [PATCH 31/70] ALSA: compress_offload: remove redundant initialization

Fix cppcheck warning:

sound/core/compress_offload.c:1044:6: style: Redundant initialization
for 'ret'. The initialized value is overwritten before it is
read. [redundantInitialization]

 ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS,
     ^
sound/core/compress_offload.c:1034:10: note: ret is initialized
 int ret = -EINVAL;
         ^
sound/core/compress_offload.c:1044:6: note: ret is overwritten
 ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS,
     ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200902212133.30964-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index e3eb314acb10..c1fec932c49d 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -1032,7 +1032,7 @@ static const struct file_operations snd_compr_file_ops = {
 
 static int snd_compress_dev_register(struct snd_device *device)
 {
-	int ret = -EINVAL;
+	int ret;
 	struct snd_compr *compr;
 
 	if (snd_BUG_ON(!device || !device->device_data))

From bec206db9db59cbc14993583288e7489e8ab02b1 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:23 -0500
Subject: [PATCH 32/70] ALSA: core: init: use DECLARE_COMPLETION_ONSTACK()
 macro

Follow recommendation in Documentation/scheduler/completion.rst and use
macro to declare local 'struct completion'

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/init.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

diff --git a/sound/core/init.c b/sound/core/init.c
index 0478847ba2b8..764dbe673d48 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -519,10 +519,9 @@ EXPORT_SYMBOL(snd_card_free_when_closed);
  */
 int snd_card_free(struct snd_card *card)
 {
-	struct completion released;
+	DECLARE_COMPLETION_ONSTACK(released);
 	int ret;
 
-	init_completion(&released);
 	card->release_completion = &released;
 	ret = snd_card_free_when_closed(card);
 	if (ret)

From 5656a7a06f0d5d90025f05c59424eb03d66e0039 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:24 -0500
Subject: [PATCH 33/70] ALSA: aoa: i2sbus: use DECLARE_COMPLETION_ONSTACK()
 macro

Follow recommendation in Documentation/scheduler/completion.rst and
use macro to declare local 'struct completion'

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/aoa/soundbus/i2sbus/pcm.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c
index d350dbd24305..1c8e8131a716 100644
--- a/sound/aoa/soundbus/i2sbus/pcm.c
+++ b/sound/aoa/soundbus/i2sbus/pcm.c
@@ -254,12 +254,11 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
 				 struct pcm_info *pi)
 {
 	unsigned long flags;
-	struct completion done;
+	DECLARE_COMPLETION_ONSTACK(done);
 	long timeout;
 
 	spin_lock_irqsave(&i2sdev->low_lock, flags);
 	if (pi->dbdma_ring.stopping) {
-		init_completion(&done);
 		pi->stop_completion = &done;
 		spin_unlock_irqrestore(&i2sdev->low_lock, flags);
 		timeout = wait_for_completion_timeout(&done, HZ);

From e9bd25885c7f799134534989b9f13c3db5ce6a16 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:25 -0500
Subject: [PATCH 34/70] ALSA: hda: auto_parser: remove shadowed variable
 declaration

Fix cppcheck warning:

sound/pci/hda/hda_auto_parser.c:353:7: style: Local variable 'i'
shadows outer variable [shadowVariable]
  int i = 0;
      ^
sound/pci/hda/hda_auto_parser.c:182:6: note: Shadowed declaration
 int i;
     ^
sound/pci/hda/hda_auto_parser.c:353:7: note: Shadow variable
  int i = 0;
      ^

It's not clear why a new declaration was added, remove and reuse
variable declared with larger scope.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 824f4ac1a8ce..4dc01647753c 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -350,7 +350,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	 */
 	if (!cfg->line_outs && cfg->hp_outs > 1 &&
 	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
-		int i = 0;
+		i = 0;
 		while (i < cfg->hp_outs) {
 			/* The real HPs should have the sequence 0x0f */
 			if ((hp_out[i].seq & 0x0f) == 0x0f) {

From 74610eaf310ab9b9b00a015adb54afcae038bc06 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:26 -0500
Subject: [PATCH 35/70] ALSA: hda: (cosmetic) align function parameters

Fix cppcheck warnings and use same names in headers and C code.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.h  | 2 +-
 sound/pci/hda/hda_local.h | 8 ++++----
 2 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 727b6d3ba454..8ceaf0ef5df1 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -77,7 +77,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
 
 struct hda_jack_callback *
 snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid,
-					int dev_id, hda_jack_callback_fn cb);
+					int dev_id, hda_jack_callback_fn func);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 8c28b1022f49..5beb8aa44ecd 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -100,7 +100,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol);
 int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
-			  unsigned int size, unsigned int __user *tlv);
+			  unsigned int size, unsigned int __user *_tlv);
 int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo);
 int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
@@ -119,7 +119,7 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid,
 			     int ch, int dir, int idx, int mask, int val);
 int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
-			     int dir, int idx, int mask, int val);
+			     int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 			   int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
@@ -198,7 +198,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 			  unsigned int *cur_val);
 int snd_hda_add_imux_item(struct hda_codec *codec,
 			  struct hda_input_mux *imux, const char *label,
-			  int index, int *type_index_ret);
+			  int index, int *type_idx);
 
 /*
  * Multi-channel / digital-out PCM helper
@@ -642,7 +642,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
  */
 int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_info *uinfo,
-			     int num_entries, const char * const *texts);
+			     int num_items, const char * const *texts);
 #define snd_hda_enum_bool_helper_info(kcontrol, uinfo) \
 	snd_hda_enum_helper_info(kcontrol, uinfo, 0, NULL)
 

From 04d0b5e3b1ba4aca95534767e47d140be1d6643f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:27 -0500
Subject: [PATCH 36/70] ALSA: usb: scarless_gen2: fix endianness issue

Fix Sparse warning:

sound/usb/mixer_scarlett_gen2.c:1949:24: warning: cast to restricted
__le32

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_scarlett_gen2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 0ffff7640892..d33df146d6ce 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -1946,7 +1946,7 @@ static void scarlett2_mixer_interrupt(struct urb *urb)
 		goto requeue;
 
 	if (len == 8) {
-		data = le32_to_cpu(*(u32 *)urb->transfer_buffer);
+		data = le32_to_cpu(*(__le32 *)urb->transfer_buffer);
 		if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE)
 			scarlett2_mixer_interrupt_vol_change(mixer);
 		if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE)

From 2d7a5c6c69029ae7e386cb3a04086a1e0b61cb8f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:28 -0500
Subject: [PATCH 37/70] ALSA: ac97: (cosmetic) align argument names

Fix cppcheck warning:

sound/ac97/bus.c:133:60: style:inconclusive: Function
'snd_ac97_bus_scan_one' argument 1 names different: declaration 'ac97'
definition 'adrv'. [funcArgNamesDifferent]

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/ac97/ac97_core.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h
index 0c5956e4b2f3..5a9677c3d4c3 100644
--- a/sound/ac97/ac97_core.h
+++ b/sound/ac97/ac97_core.h
@@ -3,7 +3,7 @@
  * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
 
-unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97,
+unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
 				   unsigned int codec_num);
 
 static inline bool ac97_ids_match(unsigned int id1, unsigned int id2,

From a971b42cbf8c0dd553ac0ca7fe82f4b8e9cad088 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:29 -0500
Subject: [PATCH 38/70] ALSA: atmel: ac97: clarify operator precedence

Fix cppcheck warnings:

sound/atmel/ac97c.c:478:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
                             ^
sound/atmel/ac97c.c:479:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
                             ^
sound/atmel/ac97c.c:480:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
                             ^
sound/atmel/ac97c.c:481:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
                             ^
sound/atmel/ac97c.c:482:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    casr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
                             ^
sound/atmel/ac97c.c:524:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    cosr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
                             ^
sound/atmel/ac97c.c:525:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    cosr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
                             ^
sound/atmel/ac97c.c:526:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
                             ^
sound/atmel/ac97c.c:527:30: style: Clarify calculation precedence for
'&' and '?'. [clarifyCalculation]
    cosr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
                             ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/atmel/ac97c.c | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 1006458f7f85..66ecbd4d034e 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -475,12 +475,12 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
 		struct snd_pcm_runtime *runtime;
 		int offset, next_period, block_size;
 		dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
-				casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
-				casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				casr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!casr                    ? " NONE"    : "");
+			(casr & AC97C_CSR_OVRUN)   ? " OVRUN"   : "",
+			(casr & AC97C_CSR_RXRDY)   ? " RXRDY"   : "",
+			(casr & AC97C_CSR_UNRUN)   ? " UNRUN"   : "",
+			(casr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "",
+			(casr & AC97C_CSR_TXRDY)   ? " TXRDY"   : "",
+			!casr                      ? " NONE"    : "");
 		if ((casr & camr) & AC97C_CSR_ENDTX) {
 			runtime = chip->playback_substream->runtime;
 			block_size = frames_to_bytes(runtime, runtime->period_size);
@@ -521,11 +521,11 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
 
 	if (sr & AC97C_SR_COEVT) {
 		dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n",
-				cosr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				cosr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				cosr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!cosr                    ? " NONE"    : "");
+			 (cosr & AC97C_CSR_OVRUN)   ? " OVRUN"   : "",
+			 (cosr & AC97C_CSR_RXRDY)   ? " RXRDY"   : "",
+			 (cosr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "",
+			 (cosr & AC97C_CSR_TXRDY)   ? " TXRDY"   : "",
+			 !cosr                      ? " NONE"    : "");
 		retval = IRQ_HANDLED;
 	}
 

From 93fcef86caafba893f2e13782f4c58b386eab311 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:30 -0500
Subject: [PATCH 39/70] ALSA: rawmidi: (cosmetic) align function parameters

fix cppcheck:

sound/core/rawmidi.c:1711:49: style:inconclusive: Function
'snd_rawmidi_free' argument 1 names different: declaration 'rawmidi'
definition 'rmidi'. [funcArgNamesDifferent]

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/rawmidi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 2a688b711a9a..c78720a3299c 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -35,7 +35,7 @@ module_param_array(amidi_map, int, NULL, 0444);
 MODULE_PARM_DESC(amidi_map, "Raw MIDI device number assigned to 2nd OSS device.");
 #endif /* CONFIG_SND_OSSEMUL */
 
-static int snd_rawmidi_free(struct snd_rawmidi *rawmidi);
+static int snd_rawmidi_free(struct snd_rawmidi *rmidi);
 static int snd_rawmidi_dev_free(struct snd_device *device);
 static int snd_rawmidi_dev_register(struct snd_device *device);
 static int snd_rawmidi_dev_disconnect(struct snd_device *device);

From b7dcd6ac0225aa3bc4c78fc5f13610789c9a7f15 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:31 -0500
Subject: [PATCH 40/70] ALSA: vx: vx_core: clarify operator precedence

Fix cppcheck warning

sound/drivers/vx/vx_core.c:600:49: style: Clarify calculation
precedence for '&' and '?'. [clarifyCalculation]
      chip->chip_status & VX_STAT_XILINX_LOADED ? "Loaded" : "No");
                                                ^
sound/drivers/vx/vx_core.c:602:47: style: Clarify calculation
precedence for '&' and '?'. [clarifyCalculation]
      chip->chip_status & VX_STAT_DEVICE_INIT ? "Yes" : "No");
                                              ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/vx/vx_core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index 26d591fe6a6b..d5c65cab195b 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -597,9 +597,9 @@ static void vx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *b
 	
 	snd_iprintf(buffer, "%s\n", chip->card->longname);
 	snd_iprintf(buffer, "Xilinx Firmware: %s\n",
-		    chip->chip_status & VX_STAT_XILINX_LOADED ? "Loaded" : "No");
+		    (chip->chip_status & VX_STAT_XILINX_LOADED) ? "Loaded" : "No");
 	snd_iprintf(buffer, "Device Initialized: %s\n",
-		    chip->chip_status & VX_STAT_DEVICE_INIT ? "Yes" : "No");
+		    (chip->chip_status & VX_STAT_DEVICE_INIT) ? "Yes" : "No");
 	snd_iprintf(buffer, "DSP audio info:");
 	if (chip->audio_info & VX_AUDIO_INFO_REAL_TIME)
 		snd_iprintf(buffer, " realtime");

From e408ab068aeca958b83bb6e5ca724356f8c7d78e Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:32 -0500
Subject: [PATCH 41/70] ALSA: vx: vx_pcm: remove redundant assignment

Fix cppcheck warning:

sound/drivers/vx/vx_pcm.c:539:30: style: Variable
'chip->playback_pipes[audio]' is reassigned a value before the old one
has been used. [redundantAssignment]
 chip->playback_pipes[audio] = pipe;
                             ^
sound/drivers/vx/vx_pcm.c:533:31: note: chip->playback_pipes[audio] is
assigned
  chip->playback_pipes[audio] = pipe;
                              ^
sound/drivers/vx/vx_pcm.c:539:30: note: chip->playback_pipes[audio] is
overwritten
 chip->playback_pipes[audio] = pipe;
                             ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/vx/vx_pcm.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 664b9efa9a50..47361e26be80 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -530,7 +530,6 @@ static int vx_pcm_playback_open(struct snd_pcm_substream *subs)
 		err = vx_alloc_pipe(chip, 0, audio, 2, &pipe); /* stereo playback */
 		if (err < 0)
 			return err;
-		chip->playback_pipes[audio] = pipe;
 	}
 	/* open for playback */
 	pipe->references++;

From b248b9dd5975c1e16b7918c1dc7eaef7eccfde5f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:33 -0500
Subject: [PATCH 42/70] ALSA: vx: vx_pcm: remove redundant assignment

Fix cppcheck warning:

sound/drivers/vx/vx_pcm.c:63:7: style: Variable 'buf' is assigned a
value that is never used. [unreadVariable]
  buf = (unsigned char *)runtime->dma_area;
      ^

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/vx/vx_pcm.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 47361e26be80..3d2e3bcafca8 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -60,7 +60,6 @@ static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime *
 	*buf++ = vx_inb(chip, RXL);
 	if (++offset >= pipe->buffer_bytes) {
 		offset = 0;
-		buf = (unsigned char *)runtime->dma_area;
 	}
 	pipe->hw_ptr = offset;
 }

From 68f86a905e2c1d7a6d5d0bcc57f4e44ae204c171 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:21 +0200
Subject: [PATCH 43/70] ALSA: pcsp: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  This patch replaces the usage of
tasklet in pcsp driver with a simple work.  In pcsp driver, a global
tasklet is used for offloading the period-elapse handling in the
hrtimer callback (introduced in commit 96c7d478efad "ALSA: pcsp - Fix
locking messes in snd-pcsp").  It can be achieved gracefully with a
work queued in the high-prio system workqueue.

This also changes tasklet_kill() with cancel_work_sync() in the
sync_stop callback, which is anyway better to assure canceling the
pending tasks.

Link: https://lore.kernel.org/r/20200903104131.21097-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/pcsp/pcsp_lib.c | 12 +++++-------
 1 file changed, 5 insertions(+), 7 deletions(-)

diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 4e79293d7f11..ed40d0f7432c 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -23,10 +23,10 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
 #define DMIX_WANTS_S16	1
 
 /*
- * Call snd_pcm_period_elapsed in a tasklet
+ * Call snd_pcm_period_elapsed in a work
  * This avoids spinlock messes and long-running irq contexts
  */
-static void pcsp_call_pcm_elapsed(unsigned long priv)
+static void pcsp_call_pcm_elapsed(struct work_struct *work)
 {
 	if (atomic_read(&pcsp_chip.timer_active)) {
 		struct snd_pcm_substream *substream;
@@ -36,7 +36,7 @@ static void pcsp_call_pcm_elapsed(unsigned long priv)
 	}
 }
 
-static DECLARE_TASKLET_OLD(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed);
+static DECLARE_WORK(pcsp_pcm_work, pcsp_call_pcm_elapsed);
 
 /* write the port and returns the next expire time in ns;
  * called at the trigger-start and in hrtimer callback
@@ -119,11 +119,9 @@ static void pcsp_pointer_update(struct snd_pcsp *chip)
 	if (periods_elapsed) {
 		chip->period_ptr += periods_elapsed * period_bytes;
 		chip->period_ptr %= buffer_bytes;
+		queue_work(system_highpri_wq, &pcsp_pcm_work);
 	}
 	spin_unlock_irqrestore(&chip->substream_lock, flags);
-
-	if (periods_elapsed)
-		tasklet_schedule(&pcsp_pcm_tasklet);
 }
 
 enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
@@ -196,7 +194,7 @@ void pcsp_sync_stop(struct snd_pcsp *chip)
 	pcsp_stop_playing(chip);
 	local_irq_enable();
 	hrtimer_cancel(&chip->timer);
-	tasklet_kill(&pcsp_pcm_tasklet);
+	cancel_work_sync(&pcsp_pcm_work);
 }
 
 static int snd_pcsp_playback_close(struct snd_pcm_substream *substream)

From bf0835957f553aeddec896f3de386562536feee4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:22 +0200
Subject: [PATCH 44/70] ALSA: timer: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In ALSA core timer API, the
callbacks can be offlined to a tasklet when a flag is set in the timer
backend.  It can be achieved gracefully with a work queued in the
high-prio system workqueue.

This patch replaces the usage of tasklet in ALSA timer API with a
simple work.  Currently the tasklet feature is used only in the system
timer and hrtimer backends, so both are patched to use the new flag
name SNDRV_TIMER_HW_WORK, too.

Link: https://lore.kernel.org/r/20200903104131.21097-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/timer.h |  8 ++++----
 sound/core/hrtimer.c  |  2 +-
 sound/core/timer.c    | 20 ++++++++++----------
 3 files changed, 15 insertions(+), 15 deletions(-)

diff --git a/include/sound/timer.h b/include/sound/timer.h
index 23e885d31525..760e132cc0cd 100644
--- a/include/sound/timer.h
+++ b/include/sound/timer.h
@@ -21,13 +21,13 @@
 #define SNDRV_TIMER_HW_STOP	0x00000002	/* call stop before start */
 #define SNDRV_TIMER_HW_SLAVE	0x00000004	/* only slave timer (variable resolution) */
 #define SNDRV_TIMER_HW_FIRST	0x00000008	/* first tick can be incomplete */
-#define SNDRV_TIMER_HW_TASKLET	0x00000010	/* timer is called from tasklet */
+#define SNDRV_TIMER_HW_WORK	0x00000010	/* timer is called from work */
 
 #define SNDRV_TIMER_IFLG_SLAVE	  0x00000001
 #define SNDRV_TIMER_IFLG_RUNNING  0x00000002
 #define SNDRV_TIMER_IFLG_START	  0x00000004
 #define SNDRV_TIMER_IFLG_AUTO	  0x00000008	/* auto restart */
-#define SNDRV_TIMER_IFLG_FAST	  0x00000010	/* fast callback (do not use tasklet) */
+#define SNDRV_TIMER_IFLG_FAST	  0x00000010	/* fast callback (do not use work) */
 #define SNDRV_TIMER_IFLG_CALLBACK 0x00000020	/* timer callback is active */
 #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040	/* exclusive owner - no more instances */
 #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080	/* write early event to the poll queue */
@@ -74,7 +74,7 @@ struct snd_timer {
 	struct list_head active_list_head;
 	struct list_head ack_list_head;
 	struct list_head sack_list_head; /* slow ack list head */
-	struct tasklet_struct task_queue;
+	struct work_struct task_work;
 	int max_instances;	/* upper limit of timer instances */
 	int num_instances;	/* current number of timer instances */
 };
@@ -96,7 +96,7 @@ struct snd_timer_instance {
 	unsigned long ticks;		/* auto-load ticks when expired */
 	unsigned long cticks;		/* current ticks */
 	unsigned long pticks;		/* accumulated ticks for callback */
-	unsigned long resolution;	/* current resolution for tasklet */
+	unsigned long resolution;	/* current resolution for work */
 	unsigned long lost;		/* lost ticks */
 	int slave_class;
 	unsigned int slave_id;
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index c61ba52a530a..e97ff8cccb64 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -114,7 +114,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
 }
 
 static const struct snd_timer_hardware hrtimer_hw __initconst = {
-	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
+	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_WORK,
 	.open =		snd_hrtimer_open,
 	.close =	snd_hrtimer_close,
 	.start =	snd_hrtimer_start,
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 227d8152d325..765ea66665a8 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -813,12 +813,12 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer,
 }
 
 /*
- * timer tasklet
+ * timer work
  *
  */
-static void snd_timer_tasklet(struct tasklet_struct *t)
+static void snd_timer_work(struct work_struct *work)
 {
-	struct snd_timer *timer = from_tasklet(timer, t, task_queue);
+	struct snd_timer *timer = container_of(work, struct snd_timer, task_work);
 	unsigned long flags;
 
 	if (timer->card && timer->card->shutdown) {
@@ -843,7 +843,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	unsigned long resolution;
 	struct list_head *ack_list_head;
 	unsigned long flags;
-	int use_tasklet = 0;
+	bool use_work = false;
 
 	if (timer == NULL)
 		return;
@@ -884,7 +884,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 			--timer->running;
 			list_del_init(&ti->active_list);
 		}
-		if ((timer->hw.flags & SNDRV_TIMER_HW_TASKLET) ||
+		if ((timer->hw.flags & SNDRV_TIMER_HW_WORK) ||
 		    (ti->flags & SNDRV_TIMER_IFLG_FAST))
 			ack_list_head = &timer->ack_list_head;
 		else
@@ -919,11 +919,11 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	snd_timer_process_callbacks(timer, &timer->ack_list_head);
 
 	/* do we have any slow callbacks? */
-	use_tasklet = !list_empty(&timer->sack_list_head);
+	use_work = !list_empty(&timer->sack_list_head);
 	spin_unlock_irqrestore(&timer->lock, flags);
 
-	if (use_tasklet)
-		tasklet_schedule(&timer->task_queue);
+	if (use_work)
+		queue_work(system_highpri_wq, &timer->task_work);
 }
 EXPORT_SYMBOL(snd_timer_interrupt);
 
@@ -967,7 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
 	INIT_LIST_HEAD(&timer->ack_list_head);
 	INIT_LIST_HEAD(&timer->sack_list_head);
 	spin_lock_init(&timer->lock);
-	tasklet_setup(&timer->task_queue, snd_timer_tasklet);
+	INIT_WORK(&timer->task_work, snd_timer_work);
 	timer->max_instances = 1000; /* default limit per timer */
 	if (card != NULL) {
 		timer->module = card->module;
@@ -1200,7 +1200,7 @@ static int snd_timer_s_close(struct snd_timer *timer)
 
 static const struct snd_timer_hardware snd_timer_system =
 {
-	.flags =	SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_TASKLET,
+	.flags =	SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_WORK,
 	.resolution =	1000000000L / HZ,
 	.ticks =	10000000L,
 	.close =	snd_timer_s_close,

From c7d9efdff68e00350e611227ad191ce347b88ea5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:23 +0200
Subject: [PATCH 45/70] ALSA: usb-audio: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In USB-audio driver, a tasklet is
still used in MIDI interface code for handling the output byte
stream.  It can be achieved gracefully with a work queued in the
high-prio system workqueue.

This patch replaces the tasklet usage in USB-audio driver with a
simple work.

Link: https://lore.kernel.org/r/20200903104131.21097-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/midi.c | 13 +++++++------
 1 file changed, 7 insertions(+), 6 deletions(-)

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index e8287a05e36b..c8213652470c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -142,7 +142,7 @@ struct snd_usb_midi_out_endpoint {
 	unsigned int active_urbs;
 	unsigned int drain_urbs;
 	int max_transfer;		/* size of urb buffer */
-	struct tasklet_struct tasklet;
+	struct work_struct work;
 	unsigned int next_urb;
 	spinlock_t buffer_lock;
 
@@ -344,9 +344,10 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep)
 	spin_unlock_irqrestore(&ep->buffer_lock, flags);
 }
 
-static void snd_usbmidi_out_tasklet(struct tasklet_struct *t)
+static void snd_usbmidi_out_work(struct work_struct *work)
 {
-	struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet);
+	struct snd_usb_midi_out_endpoint *ep =
+		container_of(work, struct snd_usb_midi_out_endpoint, work);
 
 	snd_usbmidi_do_output(ep);
 }
@@ -1177,7 +1178,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
 			snd_rawmidi_proceed(substream);
 			return;
 		}
-		tasklet_schedule(&port->ep->tasklet);
+		queue_work(system_highpri_wq, &port->ep->work);
 	}
 }
 
@@ -1440,7 +1441,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
 	}
 
 	spin_lock_init(&ep->buffer_lock);
-	tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet);
+	INIT_WORK(&ep->work, snd_usbmidi_out_work);
 	init_waitqueue_head(&ep->drain_wait);
 
 	for (i = 0; i < 0x10; ++i)
@@ -1503,7 +1504,7 @@ void snd_usbmidi_disconnect(struct list_head *p)
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
 		if (ep->out)
-			tasklet_kill(&ep->out->tasklet);
+			cancel_work_sync(&ep->out->work);
 		if (ep->out) {
 			for (j = 0; j < OUTPUT_URBS; ++j)
 				usb_kill_urb(ep->out->urbs[j].urb);

From 45e4d67f8a53e713833292cc5fef3603294a5914 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:24 +0200
Subject: [PATCH 46/70] ALSA: ua101: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In UA101 driver, a tasklet is still
used for handling the output URBs.  It can be achieved gracefully with
a work queued in the high-prio system workqueue, too.

This patch replaces the tasklet usage in UA101 driver with a simple
work.

Link: https://lore.kernel.org/r/20200903104131.21097-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/misc/ua101.c | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 3b2dce1043f5..6b30155964ec 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -96,7 +96,7 @@ struct ua101 {
 	u8 rate_feedback[MAX_QUEUE_LENGTH];
 
 	struct list_head ready_playback_urbs;
-	struct tasklet_struct playback_tasklet;
+	struct work_struct playback_work;
 	wait_queue_head_t alsa_capture_wait;
 	wait_queue_head_t rate_feedback_wait;
 	wait_queue_head_t alsa_playback_wait;
@@ -188,7 +188,7 @@ static void playback_urb_complete(struct urb *usb_urb)
 		spin_lock_irqsave(&ua->lock, flags);
 		list_add_tail(&urb->ready_list, &ua->ready_playback_urbs);
 		if (ua->rate_feedback_count > 0)
-			tasklet_schedule(&ua->playback_tasklet);
+			queue_work(system_highpri_wq, &ua->playback_work);
 		ua->playback.substream->runtime->delay -=
 				urb->urb.iso_frame_desc[0].length /
 						ua->playback.frame_bytes;
@@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua,
 		*value -= ua->playback.queue_length;
 }
 
-static void playback_tasklet(struct tasklet_struct *t)
+static void playback_work(struct work_struct *work)
 {
-	struct ua101 *ua = from_tasklet(ua, t, playback_tasklet);
+	struct ua101 *ua = container_of(work, struct ua101, playback_work);
 	unsigned long flags;
 	unsigned int frames;
 	struct ua101_urb *urb;
@@ -401,7 +401,7 @@ static void capture_urb_complete(struct urb *urb)
 		}
 		if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) &&
 		    !list_empty(&ua->ready_playback_urbs))
-			tasklet_schedule(&ua->playback_tasklet);
+			queue_work(system_highpri_wq, &ua->playback_work);
 	}
 
 	spin_unlock_irqrestore(&ua->lock, flags);
@@ -532,7 +532,7 @@ static void stop_usb_playback(struct ua101 *ua)
 
 	kill_stream_urbs(&ua->playback);
 
-	tasklet_kill(&ua->playback_tasklet);
+	cancel_work_sync(&ua->playback_work);
 
 	disable_iso_interface(ua, INTF_PLAYBACK);
 }
@@ -550,7 +550,7 @@ static int start_usb_playback(struct ua101 *ua)
 		return 0;
 
 	kill_stream_urbs(&ua->playback);
-	tasklet_kill(&ua->playback_tasklet);
+	cancel_work_sync(&ua->playback_work);
 
 	err = enable_iso_interface(ua, INTF_PLAYBACK);
 	if (err < 0)
@@ -1218,7 +1218,7 @@ static int ua101_probe(struct usb_interface *interface,
 	spin_lock_init(&ua->lock);
 	mutex_init(&ua->mutex);
 	INIT_LIST_HEAD(&ua->ready_playback_urbs);
-	tasklet_setup(&ua->playback_tasklet, playback_tasklet);
+	INIT_WORK(&ua->playback_work, playback_work);
 	init_waitqueue_head(&ua->alsa_capture_wait);
 	init_waitqueue_head(&ua->rate_feedback_wait);
 	init_waitqueue_head(&ua->alsa_playback_wait);

From 6053a712472461c3800d4406bea0c8d5ce98df1a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:25 +0200
Subject: [PATCH 47/70] ALSA: aloop: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In aloop driver, a tasklet is still
used for offloading the timer event task.  It can be achieved
gracefully with a work queued, too.

This patch replaces the tasklet usage in aloop driver with a simple
work.

Link: https://lore.kernel.org/r/20200903104131.21097-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/aloop.c | 23 +++++++++++------------
 1 file changed, 11 insertions(+), 12 deletions(-)

diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 251eaf1152e2..c91356326699 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -110,7 +110,7 @@ struct loopback_cable {
 	struct {
 		int stream;
 		struct snd_timer_id id;
-		struct tasklet_struct event_tasklet;
+		struct work_struct event_work;
 		struct snd_timer_instance *instance;
 	} snd_timer;
 };
@@ -309,8 +309,8 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm)
 	 */
 	snd_timer_close(cable->snd_timer.instance);
 
-	/* wait till drain tasklet has finished if requested */
-	tasklet_kill(&cable->snd_timer.event_tasklet);
+	/* wait till drain work has finished if requested */
+	cancel_work_sync(&cable->snd_timer.event_work);
 
 	snd_timer_instance_free(cable->snd_timer.instance);
 	memset(&cable->snd_timer, 0, sizeof(cable->snd_timer));
@@ -794,11 +794,11 @@ static void loopback_snd_timer_function(struct snd_timer_instance *timeri,
 					  resolution);
 }
 
-static void loopback_snd_timer_tasklet(unsigned long arg)
+static void loopback_snd_timer_work(struct work_struct *work)
 {
-	struct snd_timer_instance *timeri = (struct snd_timer_instance *)arg;
-	struct loopback_cable *cable = timeri->callback_data;
+	struct loopback_cable *cable;
 
+	cable = container_of(work, struct loopback_cable, snd_timer.event_work);
 	loopback_snd_timer_period_elapsed(cable, SNDRV_TIMER_EVENT_MSTOP, 0);
 }
 
@@ -828,9 +828,9 @@ static void loopback_snd_timer_event(struct snd_timer_instance *timeri,
 		 * state the streaming will be aborted by the usual timeout. It
 		 * should not be aborted here because may be the timer sound
 		 * card does only a recovery and the timer is back soon.
-		 * This tasklet triggers loopback_snd_timer_tasklet()
+		 * This work triggers loopback_snd_timer_work()
 		 */
-		tasklet_schedule(&cable->snd_timer.event_tasklet);
+		schedule_work(&cable->snd_timer.event_work);
 	}
 }
 
@@ -1124,7 +1124,7 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm)
 		err = -ENOMEM;
 		goto exit;
 	}
-	/* The callback has to be called from another tasklet. If
+	/* The callback has to be called from another work. If
 	 * SNDRV_TIMER_IFLG_FAST is specified it will be called from the
 	 * snd_pcm_period_elapsed() call of the selected sound card.
 	 * snd_pcm_period_elapsed() helds snd_pcm_stream_lock_irqsave().
@@ -1137,9 +1137,8 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm)
 	timeri->callback_data = (void *)cable;
 	timeri->ccallback = loopback_snd_timer_event;
 
-	/* initialise a tasklet used for draining */
-	tasklet_init(&cable->snd_timer.event_tasklet,
-		     loopback_snd_timer_tasklet, (unsigned long)timeri);
+	/* initialise a work used for draining */
+	INIT_WORK(&cable->snd_timer.event_work, loopback_snd_timer_work);
 
 	/* The mutex loopback->cable_lock is kept locked.
 	 * Therefore snd_timer_open() cannot be called a second time

From 4adab848ae752cc92c4de1403144002f369b079f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:26 +0200
Subject: [PATCH 48/70] ALSA: hdsp: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In HDSP driver, a tasklet is still
used for offloading the MIDI I/O handling (optional via mixer
switch).  It can be achieved gracefully with a work queued, too.

This patch replaces the tasklet usage in HDSP driver with a simple
work.  The conversion is fairly straightforward.  The only significant
difference is that a superfluous tasklet_kill() call is removed from
snd_hdap_midi_input_trigger().

Link: https://lore.kernel.org/r/20200903104131.21097-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 55 ++++++++++++++++++++--------------------
 1 file changed, 27 insertions(+), 28 deletions(-)

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index dda56ecfd33b..cea53a878c36 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -447,8 +447,8 @@ struct hdsp {
 	struct snd_pcm_substream *capture_substream;
 	struct snd_pcm_substream *playback_substream;
         struct hdsp_midi      midi[2];
-	struct tasklet_struct midi_tasklet;
-	int		      use_midi_tasklet;
+	struct work_struct    midi_work;
+	int		      use_midi_work;
 	int                   precise_ptr;
 	u32                   control_register;	     /* cached value */
 	u32                   control2_register;     /* cached value */
@@ -1385,7 +1385,6 @@ static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream,
 		}
 	} else {
 		hdsp->control_register &= ~ie;
-		tasklet_kill(&hdsp->midi_tasklet);
 	}
 
 	hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -2542,37 +2541,37 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn
 	return change;
 }
 
-#define HDSP_USE_MIDI_TASKLET(xname, xindex) \
+#define HDSP_USE_MIDI_WORK(xname, xindex) \
 { .iface = SNDRV_CTL_ELEM_IFACE_CARD, \
   .name = xname, \
   .index = xindex, \
-  .info = snd_hdsp_info_use_midi_tasklet, \
-  .get = snd_hdsp_get_use_midi_tasklet, \
-  .put = snd_hdsp_put_use_midi_tasklet \
+  .info = snd_hdsp_info_use_midi_work, \
+  .get = snd_hdsp_get_use_midi_work, \
+  .put = snd_hdsp_put_use_midi_work \
 }
 
-static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
+static int hdsp_set_use_midi_work(struct hdsp *hdsp, int use_work)
 {
-	if (use_tasklet)
-		hdsp->use_midi_tasklet = 1;
+	if (use_work)
+		hdsp->use_midi_work = 1;
 	else
-		hdsp->use_midi_tasklet = 0;
+		hdsp->use_midi_work = 0;
 	return 0;
 }
 
-#define snd_hdsp_info_use_midi_tasklet		snd_ctl_boolean_mono_info
+#define snd_hdsp_info_use_midi_work		snd_ctl_boolean_mono_info
 
-static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_hdsp_get_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 
 	spin_lock_irq(&hdsp->lock);
-	ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet;
+	ucontrol->value.integer.value[0] = hdsp->use_midi_work;
 	spin_unlock_irq(&hdsp->lock);
 	return 0;
 }
 
-static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_hdsp_put_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
@@ -2582,8 +2581,8 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
 	spin_lock_irq(&hdsp->lock);
-	change = (int)val != hdsp->use_midi_tasklet;
-	hdsp_set_use_midi_tasklet(hdsp, val);
+	change = (int)val != hdsp->use_midi_work;
+	hdsp_set_use_midi_work(hdsp, val);
 	spin_unlock_irq(&hdsp->lock);
 	return change;
 }
@@ -2950,7 +2949,7 @@ HDSP_SPDIF_SYNC_CHECK("SPDIF Lock Status", 0),
 HDSP_ADATSYNC_SYNC_CHECK("ADAT Sync Lock Status", 0),
 HDSP_TOGGLE_SETTING("Line Out", HDSP_LineOut),
 HDSP_PRECISE_POINTER("Precise Pointer", 0),
-HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0),
+HDSP_USE_MIDI_WORK("Use Midi Tasklet", 0),
 };
 
 
@@ -3370,7 +3369,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 	snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0));
 	snd_iprintf(buffer, "MIDI2 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut1));
 	snd_iprintf(buffer, "MIDI2 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn1));
-	snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_tasklet ? "on" : "off");
+	snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_work ? "on" : "off");
 
 	snd_iprintf(buffer, "\n");
 
@@ -3791,9 +3790,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 	return 0;
 }
 
-static void hdsp_midi_tasklet(struct tasklet_struct *t)
+static void hdsp_midi_work(struct work_struct *work)
 {
-	struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet);
+	struct hdsp *hdsp = container_of(work, struct hdsp, midi_work);
 
 	if (hdsp->midi[0].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[0]);
@@ -3838,7 +3837,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 	}
 
 	if (midi0 && midi0status) {
-		if (hdsp->use_midi_tasklet) {
+		if (hdsp->use_midi_work) {
 			/* we disable interrupts for this input until processing is done */
 			hdsp->control_register &= ~HDSP_Midi0InterruptEnable;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -3849,7 +3848,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 		}
 	}
 	if (hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632 && midi1 && midi1status) {
-		if (hdsp->use_midi_tasklet) {
+		if (hdsp->use_midi_work) {
 			/* we disable interrupts for this input until processing is done */
 			hdsp->control_register &= ~HDSP_Midi1InterruptEnable;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -3859,8 +3858,8 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 			snd_hdsp_midi_input_read (&hdsp->midi[1]);
 		}
 	}
-	if (hdsp->use_midi_tasklet && schedule)
-		tasklet_schedule(&hdsp->midi_tasklet);
+	if (hdsp->use_midi_work && schedule)
+		queue_work(system_highpri_wq, &hdsp->midi_work);
 	return IRQ_HANDLED;
 }
 
@@ -5182,7 +5181,7 @@ static int snd_hdsp_create(struct snd_card *card,
 
 	spin_lock_init(&hdsp->lock);
 
-	tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet);
+	INIT_WORK(&hdsp->midi_work, hdsp_midi_work);
 
 	pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
 	hdsp->firmware_rev &= 0xff;
@@ -5235,7 +5234,7 @@ static int snd_hdsp_create(struct snd_card *card,
 	hdsp->irq = pci->irq;
 	card->sync_irq = hdsp->irq;
 	hdsp->precise_ptr = 0;
-	hdsp->use_midi_tasklet = 1;
+	hdsp->use_midi_work = 1;
 	hdsp->dds_value = 0;
 
 	if ((err = snd_hdsp_initialize_memory(hdsp)) < 0)
@@ -5305,7 +5304,7 @@ static int snd_hdsp_free(struct hdsp *hdsp)
 {
 	if (hdsp->port) {
 		/* stop the audio, and cancel all interrupts */
-		tasklet_kill(&hdsp->midi_tasklet);
+		cancel_work_sync(&hdsp->midi_work);
 		hdsp->control_register &= ~(HDSP_Start|HDSP_AudioInterruptEnable|HDSP_Midi0InterruptEnable|HDSP_Midi1InterruptEnable);
 		hdsp_write (hdsp, HDSP_controlRegister, hdsp->control_register);
 	}

From a2e527c5a3ebcf248b51705ef44f6700403fd8e3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:27 +0200
Subject: [PATCH 49/70] ALSA: hdspm: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In HDSP-MADI driver, a tasklet is
still used for offloading the MIDI I/O handling (optional via mixer
switch).  It can be achieved gracefully with a work queued, too.

This patch replaces the tasklet usage in HDSP-MADI driver with a
simple work.  The conversion is fairly straightforward.  The only
significant difference is that the work initialization is moved to the
right place in snd_hdspm_create() and cancel_work_sync() is always
called in snd_hdspm_free() to assure killing the pending works.

Link: https://lore.kernel.org/r/20200903104131.21097-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 13 ++++++-------
 1 file changed, 6 insertions(+), 7 deletions(-)

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 572350aaf18d..e532312a5e1c 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -997,7 +997,7 @@ struct hdspm {
 	u32 settings_register;  /* cached value for AIO / RayDat (sync reference, master/slave) */
 
 	struct hdspm_midi midi[4];
-	struct tasklet_struct midi_tasklet;
+	struct work_struct midi_work;
 
 	size_t period_bytes;
 	unsigned char ss_in_channels;
@@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card,
 }
 
 
-static void hdspm_midi_tasklet(struct tasklet_struct *t)
+static void hdspm_midi_work(struct work_struct *work)
 {
-	struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet);
+	struct hdspm *hdspm = container_of(work, struct hdspm, midi_work);
 	int i = 0;
 
 	while (i < hdspm->midiPorts) {
@@ -5449,7 +5449,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
 		}
 
 		if (schedule)
-			tasklet_hi_schedule(&hdspm->midi_tasklet);
+			queue_work(system_highpri_wq, &hdspm->midi_work);
 	}
 
 	return IRQ_HANDLED;
@@ -6538,6 +6538,7 @@ static int snd_hdspm_create(struct snd_card *card,
 	hdspm->card = card;
 
 	spin_lock_init(&hdspm->lock);
+	INIT_WORK(&hdspm->midi_work, hdspm_midi_work);
 
 	pci_read_config_word(hdspm->pci,
 			PCI_CLASS_REVISION, &hdspm->firmware_rev);
@@ -6836,9 +6837,6 @@ static int snd_hdspm_create(struct snd_card *card,
 
 	}
 
-	tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet);
-
-
 	if (hdspm->io_type != MADIface) {
 		hdspm->serial = (hdspm_read(hdspm,
 				HDSPM_midiStatusIn0)>>8) & 0xFFFFFF;
@@ -6873,6 +6871,7 @@ static int snd_hdspm_free(struct hdspm * hdspm)
 {
 
 	if (hdspm->port) {
+		cancel_work_sync(&hdspm->midi_work);
 
 		/* stop th audio, and cancel all interrupts */
 		hdspm->control_register &=

From 2ac55daffee5af04b13f96e6acef63a8e4bfff85 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:28 +0200
Subject: [PATCH 50/70] ALSA: riptide: Replace tasklet with threaded irq

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In Riptide driver, a tasklet is
still used for offloading the PCM IRQ handling.  It can be achieved
gracefully with a threaded IRQ, too.

This patch replaces the tasklet usage in riptide driver with a
threaded IRQ.

Link: https://lore.kernel.org/r/20200903104131.21097-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/riptide/riptide.c | 20 +++++++++++---------
 1 file changed, 11 insertions(+), 9 deletions(-)

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 098c69b3b7aa..fcc2073c5025 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -445,7 +445,6 @@ struct snd_riptide {
 	union firmware_version firmware;
 
 	spinlock_t lock;
-	struct tasklet_struct riptide_tq;
 	struct snd_info_entry *proc_entry;
 
 	unsigned long received_irqs;
@@ -1070,9 +1069,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval,
 	return 0;
 }
 
-static void riptide_handleirq(struct tasklet_struct *t)
+static irqreturn_t riptide_handleirq(int irq, void *dev_id)
 {
-	struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq);
+	struct snd_riptide *chip = dev_id;
 	struct cmdif *cif = chip->cif;
 	struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1];
 	struct snd_pcm_runtime *runtime;
@@ -1083,7 +1082,7 @@ static void riptide_handleirq(struct tasklet_struct *t)
 	unsigned int flag;
 
 	if (!cif)
-		return;
+		return IRQ_HANDLED;
 
 	for (i = 0; i < PLAYBACK_SUBSTREAMS; i++)
 		substream[i] = chip->playback_substream[i];
@@ -1134,6 +1133,8 @@ static void riptide_handleirq(struct tasklet_struct *t)
 			}
 		}
 	}
+
+	return IRQ_HANDLED;
 }
 
 #ifdef CONFIG_PM_SLEEP
@@ -1699,13 +1700,14 @@ snd_riptide_interrupt(int irq, void *dev_id)
 {
 	struct snd_riptide *chip = dev_id;
 	struct cmdif *cif = chip->cif;
+	irqreturn_t ret = IRQ_HANDLED;
 
 	if (cif) {
 		chip->received_irqs++;
 		if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) ||
 		    IS_EOCIRQ(cif->hwport)) {
 			chip->handled_irqs++;
-			tasklet_schedule(&chip->riptide_tq);
+			ret = IRQ_WAKE_THREAD;
 		}
 		if (chip->rmidi && IS_MPUIRQ(cif->hwport)) {
 			chip->handled_irqs++;
@@ -1714,7 +1716,7 @@ snd_riptide_interrupt(int irq, void *dev_id)
 		}
 		SET_AIACK(cif->hwport);
 	}
-	return IRQ_HANDLED;
+	return ret;
 }
 
 static void
@@ -1843,7 +1845,6 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
 	chip->received_irqs = 0;
 	chip->handled_irqs = 0;
 	chip->cif = NULL;
-	tasklet_setup(&chip->riptide_tq, riptide_handleirq);
 
 	if ((chip->res_port =
 	     request_region(chip->port, 64, "RIPTIDE")) == NULL) {
@@ -1856,8 +1857,9 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
 	hwport = (struct riptideport *)chip->port;
 	UNSET_AIE(hwport);
 
-	if (request_irq(pci->irq, snd_riptide_interrupt, IRQF_SHARED,
-			KBUILD_MODNAME, chip)) {
+	if (request_threaded_irq(pci->irq, snd_riptide_interrupt,
+				 riptide_handleirq, IRQF_SHARED,
+				 KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n",
 			   pci->irq);
 		snd_riptide_free(chip);

From ce4f25759372bbf6c37b92712b00c1b1b9b4b48e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:29 +0200
Subject: [PATCH 51/70] ALSA: asihpi: Replace tasklet with threaded irq

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In ASIHPI driver, a tasklet is
still used for offloading the PCM IRQ handling.  It can be achieved
gracefully with a threaded IRQ, too.

This patch replaces the tasklet usage in asihpi driver with a threaded
IRQ.  It also simplified some call patterns.

Link: https://lore.kernel.org/r/20200903104131.21097-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c  | 30 ++++--------------------------
 sound/pci/asihpi/hpioctl.c | 16 +++++++++++++---
 2 files changed, 17 insertions(+), 29 deletions(-)

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 35e76480306e..46d8166ceaeb 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -117,7 +117,6 @@ struct snd_card_asihpi {
 	 * snd_card_asihpi_timer_function().
 	 */
 	struct snd_card_asihpi_pcm *llmode_streampriv;
-	struct tasklet_struct t;
 	void (*pcm_start)(struct snd_pcm_substream *substream);
 	void (*pcm_stop)(struct snd_pcm_substream *substream);
 
@@ -547,9 +546,7 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream)
 	card = snd_pcm_substream_chip(substream);
 
 	WARN_ON(in_interrupt());
-	tasklet_disable(&card->t);
 	card->llmode_streampriv = dpcm;
-	tasklet_enable(&card->t);
 
 	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
 		HPI_ADAPTER_PROPERTY_IRQ_RATE,
@@ -565,13 +562,7 @@ static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream)
 	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
 		HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
 
-	if (in_interrupt())
-		card->llmode_streampriv = NULL;
-	else {
-		tasklet_disable(&card->t);
-		card->llmode_streampriv = NULL;
-		tasklet_enable(&card->t);
-	}
+	card->llmode_streampriv = NULL;
 }
 
 static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
@@ -921,25 +912,15 @@ static void snd_card_asihpi_timer_function(struct timer_list *t)
 		add_timer(&dpcm->timer);
 }
 
-static void snd_card_asihpi_int_task(struct tasklet_struct *t)
-{
-	struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t);
-	struct hpi_adapter *a = asihpi->hpi;
-
-	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
-	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
-	if (asihpi->llmode_streampriv)
-		snd_card_asihpi_timer_function(
-			&asihpi->llmode_streampriv->timer);
-}
-
 static void snd_card_asihpi_isr(struct hpi_adapter *a)
 {
 	struct snd_card_asihpi *asihpi;
 
 	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
 	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
-	tasklet_schedule(&asihpi->t);
+	if (asihpi->llmode_streampriv)
+		snd_card_asihpi_timer_function(
+			&asihpi->llmode_streampriv->timer);
 }
 
 /***************************** PLAYBACK OPS ****************/
@@ -2871,7 +2852,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	if (hpi->interrupt_mode) {
 		asihpi->pcm_start = snd_card_asihpi_pcm_int_start;
 		asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop;
-		tasklet_setup(&asihpi->t, snd_card_asihpi_int_task);
 		hpi->interrupt_callback = snd_card_asihpi_isr;
 	} else {
 		asihpi->pcm_start = snd_card_asihpi_pcm_timer_start;
@@ -2960,14 +2940,12 @@ __nodev:
 static void snd_asihpi_remove(struct pci_dev *pci_dev)
 {
 	struct hpi_adapter *hpi = pci_get_drvdata(pci_dev);
-	struct snd_card_asihpi *asihpi = hpi->snd_card->private_data;
 
 	/* Stop interrupts */
 	if (hpi->interrupt_mode) {
 		hpi->interrupt_callback = NULL;
 		hpi_handle_error(hpi_adapter_set_property(hpi->adapter->index,
 			HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
-		tasklet_kill(&asihpi->t);
 	}
 
 	snd_card_free(hpi->snd_card);
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 496dcde9715d..6cc2b6964bb5 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -329,11 +329,20 @@ static irqreturn_t asihpi_isr(int irq, void *dev_id)
 	   asihpi_irq_count, a->adapter->type, a->adapter->index); */
 
 	if (a->interrupt_callback)
-		a->interrupt_callback(a);
+		return IRQ_WAKE_THREAD;
 
 	return IRQ_HANDLED;
 }
 
+static irqreturn_t asihpi_isr_thread(int irq, void *dev_id)
+{
+	struct hpi_adapter *a = dev_id;
+
+	if (a->interrupt_callback)
+		a->interrupt_callback(a);
+	return IRQ_HANDLED;
+}
+
 int asihpi_adapter_probe(struct pci_dev *pci_dev,
 			 const struct pci_device_id *pci_id)
 {
@@ -478,8 +487,9 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 		}
 
 		/* Note: request_irq calls asihpi_isr here */
-		if (request_irq(pci_dev->irq, asihpi_isr, IRQF_SHARED,
-				"asihpi", &adapters[adapter_index])) {
+		if (request_threaded_irq(pci_dev->irq, asihpi_isr,
+					 asihpi_isr_thread, IRQF_SHARED,
+					 "asihpi", &adapters[adapter_index])) {
 			dev_err(&pci_dev->dev, "request_irq(%d) failed\n",
 				pci_dev->irq);
 			goto err;

From f2a852d36711bd7c7c717abc66c07751f3e1d482 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 Sep 2020 12:41:31 +0200
Subject: [PATCH 52/70] ALSA: mixart: Correct comment wrt obsoleted tasklet
 usage

The miXart driver has been already converted to use the threaded IRQ
instead of tasklet while there is a remaining comment still mentioning
a tasklet.  Update the comment appropriately.

Fixes: 8d3a8b5cb57d ("ALSA: mixart: Use nonatomic PCM ops")
Link: https://lore.kernel.org/r/20200903104131.21097-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h
index 42111562e9bc..cbed6d9a9f2e 100644
--- a/sound/pci/mixart/mixart.h
+++ b/sound/pci/mixart/mixart.h
@@ -69,7 +69,7 @@ struct mixart_mgr {
 	u32 msg_fifo[MSG_FIFO_SIZE];
 	int msg_fifo_readptr;
 	int msg_fifo_writeptr;
-	atomic_t msg_processed;       /* number of messages to be processed in tasklet */
+	atomic_t msg_processed;       /* number of messages to be processed in irq thread */
 
 	struct mutex lock;              /* interrupt lock */
 	struct mutex msg_lock;		/* mailbox lock */

From 175860c50a800fb46f5f3cb44ed3db0344c0fcc5 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 9 Sep 2020 21:49:27 +0800
Subject: [PATCH 53/70] ALSA: pci/asihpi: Remove unused function
 hpi_stream_group_get_map()

There is no caller in tree, so can remove it.

Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200909134927.33964-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 9 ---------
 1 file changed, 9 deletions(-)

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 46d8166ceaeb..5e1f9f10051b 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -257,15 +257,6 @@ static inline u16 hpi_stream_group_reset(u32 h_stream)
 		return hpi_instream_group_reset(h_stream);
 }
 
-static inline u16 hpi_stream_group_get_map(
-				u32 h_stream, u32 *mo, u32 *mi)
-{
-	if (hpi_handle_object(h_stream) ==  HPI_OBJ_OSTREAM)
-		return hpi_outstream_group_get_map(h_stream, mo, mi);
-	else
-		return hpi_instream_group_get_map(h_stream, mo, mi);
-}
-
 static u16 handle_error(u16 err, int line, char *filename)
 {
 	if (err)

From 5554743d88483934cabff45943913be0830e66bc Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 9 Sep 2020 21:57:44 +0800
Subject: [PATCH 54/70] ALSA: pcm: Remove unused inline function
 snd_mask_sizeof

There is no caller in tree, so can remove it.

Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200909135744.33464-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/pcm_params.h | 5 -----
 1 file changed, 5 deletions(-)

diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 36f94735d23d..ba184f49f7e1 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -23,11 +23,6 @@ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
 #define MASK_OFS(i)	((i) >> 5)
 #define MASK_BIT(i)	(1U << ((i) & 31))
 
-static inline size_t snd_mask_sizeof(void)
-{
-	return sizeof(struct snd_mask);
-}
-
 static inline void snd_mask_none(struct snd_mask *mask)
 {
 	memset(mask, 0, sizeof(*mask));

From 2b3d2987d800cc6dd8f6459971bc332354bd6a31 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 9 Sep 2020 18:36:59 +0200
Subject: [PATCH 55/70] ALSA: firewire: Replace tasklet with work

The tasklet is an old API that should be deprecated, usually can be
converted to another decent API.  In FireWire driver, a tasklet is
still used for offloading the AMDTP PCM stream handling.  It can be
achieved gracefully with a work queued, too.

This patch replaces the tasklet usage in firewire-lib driver with a
simple work.  The conversion is fairly straightforward but for the
in_interrupt() checks that are replaced with the check using the
current_work().

Note that in_interrupt() in amdtp_packet tracepoint is still kept as
is.  This is the place that is probed by both softirq of 1394 OHCI and
a user task of a PCM application, and the work handling is already
filtered in amdtp_domain_stream_pcm_pointer().

Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200909163659.21708-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 25 +++++++++++++------------
 sound/firewire/amdtp-stream.h |  2 +-
 2 files changed, 14 insertions(+), 13 deletions(-)

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index ee1c428b1fd3..4e2f2bb7879f 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -64,7 +64,7 @@
 #define IT_PKT_HEADER_SIZE_CIP		8 // For 2 CIP header.
 #define IT_PKT_HEADER_SIZE_NO_CIP	0 // Nothing.
 
-static void pcm_period_tasklet(struct tasklet_struct *t);
+static void pcm_period_work(struct work_struct *work);
 
 /**
  * amdtp_stream_init - initialize an AMDTP stream structure
@@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
-	tasklet_setup(&s->period_tasklet, pcm_period_tasklet);
+	INIT_WORK(&s->period_work, pcm_period_work);
 	s->packet_index = 0;
 
 	init_waitqueue_head(&s->callback_wait);
@@ -203,7 +203,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 
 	// Linux driver for 1394 OHCI controller voluntarily flushes isoc
 	// context when total size of accumulated context header reaches
-	// PAGE_SIZE. This kicks tasklet for the isoc context and brings
+	// PAGE_SIZE. This kicks work for the isoc context and brings
 	// callback in the middle of scheduled interrupts.
 	// Although AMDTP streams in the same domain use the same events per
 	// IRQ, use the largest size of context header between IT/IR contexts.
@@ -333,7 +333,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload);
  */
 void amdtp_stream_pcm_prepare(struct amdtp_stream *s)
 {
-	tasklet_kill(&s->period_tasklet);
+	cancel_work_sync(&s->period_work);
 	s->pcm_buffer_pointer = 0;
 	s->pcm_period_pointer = 0;
 }
@@ -437,13 +437,14 @@ static void update_pcm_pointers(struct amdtp_stream *s,
 	s->pcm_period_pointer += frames;
 	if (s->pcm_period_pointer >= pcm->runtime->period_size) {
 		s->pcm_period_pointer -= pcm->runtime->period_size;
-		tasklet_hi_schedule(&s->period_tasklet);
+		queue_work(system_highpri_wq, &s->period_work);
 	}
 }
 
-static void pcm_period_tasklet(struct tasklet_struct *t)
+static void pcm_period_work(struct work_struct *work)
 {
-	struct amdtp_stream *s = from_tasklet(s, t, period_tasklet);
+	struct amdtp_stream *s = container_of(work, struct amdtp_stream,
+					      period_work);
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 
 	if (pcm)
@@ -794,7 +795,7 @@ 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 (in_interrupt())
+	if (current_work() == &s->period_work)
 		amdtp_stream_pcm_abort(s);
 	WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN);
 }
@@ -1184,7 +1185,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d,
 
 	if (irq_target && amdtp_stream_running(irq_target)) {
 		// This function is called in software IRQ context of
-		// period_tasklet or process context.
+		// period_work or process context.
 		//
 		// When the software IRQ context was scheduled by software IRQ
 		// context of IT contexts, queued packets were already handled.
@@ -1195,9 +1196,9 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d,
 		// immediately to keep better granularity of PCM pointer.
 		//
 		// Later, the process context will sometimes schedules software
-		// IRQ context of the period_tasklet. Then, no need to flush the
+		// IRQ context of the period_work. Then, no need to flush the
 		// queue by the same reason as described in the above
-		if (!in_interrupt()) {
+		if (current_work() != &s->period_work) {
 			// Queued packet should be processed without any kernel
 			// preemption to keep latency against bus cycle.
 			preempt_disable();
@@ -1263,7 +1264,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s)
 		return;
 	}
 
-	tasklet_kill(&s->period_tasklet);
+	cancel_work_sync(&s->period_work);
 	fw_iso_context_stop(s->context);
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 703b710aaf7f..2ceb57d1d58e 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -163,7 +163,7 @@ struct amdtp_stream {
 
 	/* For a PCM substream processing. */
 	struct snd_pcm_substream *pcm;
-	struct tasklet_struct period_tasklet;
+	struct work_struct period_work;
 	snd_pcm_uframes_t pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
 

From 2b987515e1d18836d9df96a5876290c8c8208348 Mon Sep 17 00:00:00 2001
From: Wang Qing <wangqing@vivo.com>
Date: Thu, 17 Sep 2020 15:55:52 +0800
Subject: [PATCH 56/70] ALSA: asihpi: fix spellint typo in comments

Change the comment typo: "ununsed" -> "unused".

Signed-off-by: Wang Qing <wangqing@vivo.com>
Link: https://lore.kernel.org/r/1600329372-2266-1-git-send-email-wangqing@vivo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpios.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index 26f7cf455a1e..9e551bc46264 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -67,7 +67,7 @@ struct hpi_ioctl_linux {
 };
 
 /* Conflict?: H is already used by a number of drivers hid, bluetooth hci,
-   and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is ununsed command
+   and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is unused command
 */
 #define HPI_IOCTL_LINUX _IOWR('H', 0xFC, struct hpi_ioctl_linux)
 

From 18d122c0287b29e70bc312a994c7ee79738cec77 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Fri, 18 Sep 2020 11:56:19 +0200
Subject: [PATCH 57/70] ALSA: compat_ioctl: avoid compat_alloc_user_space

Using compat_alloc_user_space() tends to add complexity
to the ioctl handling, so I am trying to remove it everywhere.

The two callers in sound/core can rewritten to just call
the same code that operates on a kernel pointer as the
native handler.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200918095642.1446243-1-arnd@arndb.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/control.c        | 38 ++++++++++++++++++++++++-------------
 sound/core/control_compat.c | 14 ++++++--------
 sound/core/hwdep.c          | 27 ++++++++++++++++----------
 sound/core/hwdep_compat.c   | 25 ++++++++----------------
 4 files changed, 56 insertions(+), 48 deletions(-)

diff --git a/sound/core/control.c b/sound/core/control.c
index aa0c0cf182af..e014598142df 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -717,22 +717,19 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl,
 }
 
 static int snd_ctl_elem_list(struct snd_card *card,
-			     struct snd_ctl_elem_list __user *_list)
+			     struct snd_ctl_elem_list *list)
 {
-	struct snd_ctl_elem_list list;
 	struct snd_kcontrol *kctl;
 	struct snd_ctl_elem_id id;
 	unsigned int offset, space, jidx;
 	int err = 0;
 
-	if (copy_from_user(&list, _list, sizeof(list)))
-		return -EFAULT;
-	offset = list.offset;
-	space = list.space;
+	offset = list->offset;
+	space = list->space;
 
 	down_read(&card->controls_rwsem);
-	list.count = card->controls_count;
-	list.used = 0;
+	list->count = card->controls_count;
+	list->used = 0;
 	if (space > 0) {
 		list_for_each_entry(kctl, &card->controls, list) {
 			if (offset >= kctl->count) {
@@ -741,12 +738,12 @@ static int snd_ctl_elem_list(struct snd_card *card,
 			}
 			for (jidx = offset; jidx < kctl->count; jidx++) {
 				snd_ctl_build_ioff(&id, kctl, jidx);
-				if (copy_to_user(list.pids + list.used, &id,
+				if (copy_to_user(list->pids + list->used, &id,
 						 sizeof(id))) {
 					err = -EFAULT;
 					goto out;
 				}
-				list.used++;
+				list->used++;
 				if (!--space)
 					goto out;
 			}
@@ -755,11 +752,26 @@ static int snd_ctl_elem_list(struct snd_card *card,
 	}
  out:
 	up_read(&card->controls_rwsem);
-	if (!err && copy_to_user(_list, &list, sizeof(list)))
-		err = -EFAULT;
 	return err;
 }
 
+static int snd_ctl_elem_list_user(struct snd_card *card,
+				  struct snd_ctl_elem_list __user *_list)
+{
+	struct snd_ctl_elem_list list;
+	int err;
+
+	if (copy_from_user(&list, _list, sizeof(list)))
+		return -EFAULT;
+	err = snd_ctl_elem_list(card, &list);
+	if (err)
+		return err;
+	if (copy_to_user(_list, &list, sizeof(list)))
+		return -EFAULT;
+
+	return 0;
+}
+
 /* Check whether the given kctl info is valid */
 static int snd_ctl_check_elem_info(struct snd_card *card,
 				   const struct snd_ctl_elem_info *info)
@@ -1703,7 +1715,7 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg
 	case SNDRV_CTL_IOCTL_CARD_INFO:
 		return snd_ctl_card_info(card, ctl, cmd, argp);
 	case SNDRV_CTL_IOCTL_ELEM_LIST:
-		return snd_ctl_elem_list(card, argp);
+		return snd_ctl_elem_list_user(card, argp);
 	case SNDRV_CTL_IOCTL_ELEM_INFO:
 		return snd_ctl_elem_info_user(ctl, argp);
 	case SNDRV_CTL_IOCTL_ELEM_READ:
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index 02df1d7db9a1..1d708aab9c98 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -22,24 +22,22 @@ struct snd_ctl_elem_list32 {
 static int snd_ctl_elem_list_compat(struct snd_card *card,
 				    struct snd_ctl_elem_list32 __user *data32)
 {
-	struct snd_ctl_elem_list __user *data;
+	struct snd_ctl_elem_list data = {};
 	compat_caddr_t ptr;
 	int err;
 
-	data = compat_alloc_user_space(sizeof(*data));
-
 	/* offset, space, used, count */
-	if (copy_in_user(data, data32, 4 * sizeof(u32)))
+	if (copy_from_user(&data, data32, 4 * sizeof(u32)))
 		return -EFAULT;
 	/* pids */
-	if (get_user(ptr, &data32->pids) ||
-	    put_user(compat_ptr(ptr), &data->pids))
+	if (get_user(ptr, &data32->pids))
 		return -EFAULT;
-	err = snd_ctl_elem_list(card, data);
+	data.pids = compat_ptr(ptr);
+	err = snd_ctl_elem_list(card, &data);
 	if (err < 0)
 		return err;
 	/* copy the result */
-	if (copy_in_user(data32, data, 4 * sizeof(u32)))
+	if (copy_to_user(data32, &data, 4 * sizeof(u32)))
 		return -EFAULT;
 	return 0;
 }
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 21edb8ac95eb..0c029892880a 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -203,28 +203,35 @@ static int snd_hwdep_dsp_status(struct snd_hwdep *hw,
 }
 
 static int snd_hwdep_dsp_load(struct snd_hwdep *hw,
-			      struct snd_hwdep_dsp_image __user *_info)
+			      struct snd_hwdep_dsp_image *info)
 {
-	struct snd_hwdep_dsp_image info;
 	int err;
 	
 	if (! hw->ops.dsp_load)
 		return -ENXIO;
-	memset(&info, 0, sizeof(info));
-	if (copy_from_user(&info, _info, sizeof(info)))
-		return -EFAULT;
-	if (info.index >= 32)
+	if (info->index >= 32)
 		return -EINVAL;
 	/* check whether the dsp was already loaded */
-	if (hw->dsp_loaded & (1u << info.index))
+	if (hw->dsp_loaded & (1u << info->index))
 		return -EBUSY;
-	err = hw->ops.dsp_load(hw, &info);
+	err = hw->ops.dsp_load(hw, info);
 	if (err < 0)
 		return err;
-	hw->dsp_loaded |= (1u << info.index);
+	hw->dsp_loaded |= (1u << info->index);
 	return 0;
 }
 
+static int snd_hwdep_dsp_load_user(struct snd_hwdep *hw,
+				   struct snd_hwdep_dsp_image __user *_info)
+{
+	struct snd_hwdep_dsp_image info = {};
+
+	if (copy_from_user(&info, _info, sizeof(info)))
+		return -EFAULT;
+	return snd_hwdep_dsp_load(hw, &info);
+}
+
+
 static long snd_hwdep_ioctl(struct file * file, unsigned int cmd,
 			    unsigned long arg)
 {
@@ -238,7 +245,7 @@ static long snd_hwdep_ioctl(struct file * file, unsigned int cmd,
 	case SNDRV_HWDEP_IOCTL_DSP_STATUS:
 		return snd_hwdep_dsp_status(hw, argp);
 	case SNDRV_HWDEP_IOCTL_DSP_LOAD:
-		return snd_hwdep_dsp_load(hw, argp);
+		return snd_hwdep_dsp_load_user(hw, argp);
 	}
 	if (hw->ops.ioctl)
 		return hw->ops.ioctl(hw, file, cmd, arg);
diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c
index bc81db9cb3d4..a0b76706c083 100644
--- a/sound/core/hwdep_compat.c
+++ b/sound/core/hwdep_compat.c
@@ -19,26 +19,17 @@ struct snd_hwdep_dsp_image32 {
 static int snd_hwdep_dsp_load_compat(struct snd_hwdep *hw,
 				     struct snd_hwdep_dsp_image32 __user *src)
 {
-	struct snd_hwdep_dsp_image __user *dst;
+	struct snd_hwdep_dsp_image info = {};
 	compat_caddr_t ptr;
-	u32 val;
 
-	dst = compat_alloc_user_space(sizeof(*dst));
+	if (copy_from_user(&info, src, 4 + 64) ||
+	    get_user(ptr, &src->image) ||
+	    get_user(info.length, &src->length) ||
+	    get_user(info.driver_data, &src->driver_data))
+		return -EFAULT;
+	info.image = compat_ptr(ptr);
 
-	/* index and name */
-	if (copy_in_user(dst, src, 4 + 64))
-		return -EFAULT;
-	if (get_user(ptr, &src->image) ||
-	    put_user(compat_ptr(ptr), &dst->image))
-		return -EFAULT;
-	if (get_user(val, &src->length) ||
-	    put_user(val, &dst->length))
-		return -EFAULT;
-	if (get_user(val, &src->driver_data) ||
-	    put_user(val, &dst->driver_data))
-		return -EFAULT;
-
-	return snd_hwdep_dsp_load(hw, dst);
+	return snd_hwdep_dsp_load(hw, &info);
 }
 
 enum {

From 69b08bdfa8181bc7babd7d81c93dc60142c4bfd3 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:38 +0300
Subject: [PATCH 58/70] ALSA: hda - add Intel DG1 PCI and HDMI ids

Add Intel DG1 PCI id to list of supported HDA controllers and
add its HDMI id as well.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c  | 3 +++
 sound/pci/hda/patch_hdmi.c | 1 +
 2 files changed, 4 insertions(+)

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 36a9dbc33aa0..904d3af5b99e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2493,6 +2493,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake-H */
 	{ PCI_DEVICE(0x8086, 0x43c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* DG1 */
+	{ PCI_DEVICE(0x8086, 0x490d),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 402050088090..055440740184 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI",	patch_i915_icl_hdmi),

From 1bee263dfda57e45ad39c59a663c123a357ce38b Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:39 +0300
Subject: [PATCH 59/70] ALSA: hda - controller is in GPU on the DG1

Add Intel DG1 to the CONTROLLER_IN_GPU list to ensure audio power is
requested whenever programming the controller.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-3-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 904d3af5b99e..61e495187b1a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -368,7 +368,8 @@ enum {
 #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
 					((pci)->device == 0x0c0c) || \
 					((pci)->device == 0x0d0c) || \
-					((pci)->device == 0x160c))
+					((pci)->device == 0x160c) || \
+					((pci)->device == 0x490d))
 
 #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
 

From 7b882fe3e3e8bfd6427bf61780e59d5bed6699b4 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:40 +0300
Subject: [PATCH 60/70] ALSA: hda - handle multiple i915 device instances

Currently i915_component_master_match() will return the first matching
i915 instance. This does not work in case system has multiple i915
and HDA audio controller instances.

Add a new connectivity check that handles following cases:
 - i915 and HDA controller on same PCI bus
 - discrete GPU with embedded HDA audio controller connected
   via PCI bridge

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_i915.c | 44 +++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 42 insertions(+), 2 deletions(-)

diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 3c2db3816029..50b2c1db429b 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -73,11 +73,51 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
 
+/**
+ * Returns true if the devices can be connected for audio.
+ */
+static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac)
+{
+	struct pci_bus *bus_a = i915->bus, *bus_b = hdac->bus;
+
+	/* directly connected on the same bus */
+	if (bus_a == bus_b)
+		return true;
+
+	/*
+	 * on i915 discrete GPUs with embedded HDA audio, the two
+	 * devices are connected via 2nd level PCI bridge
+	 */
+	bus_a = bus_a->parent;
+	bus_b = bus_b->parent;
+	if (!bus_a || !bus_b)
+		return false;
+	bus_a = bus_a->parent;
+	bus_b = bus_b->parent;
+	if (bus_a && bus_a == bus_b)
+		return true;
+
+	return false;
+}
+
 static int i915_component_master_match(struct device *dev, int subcomponent,
 				       void *data)
 {
-	return !strcmp(dev->driver->name, "i915") &&
-	       subcomponent == I915_COMPONENT_AUDIO;
+	struct pci_dev *hdac_pci, *i915_pci;
+	struct hdac_bus *bus = data;
+
+	if (!dev_is_pci(dev))
+		return 0;
+
+	hdac_pci = to_pci_dev(bus->dev);
+	i915_pci = to_pci_dev(dev);
+
+	if (!strcmp(dev->driver->name, "i915") &&
+	    subcomponent == I915_COMPONENT_AUDIO &&
+	    connectivity_check(i915_pci, hdac_pci))
+		return 1;
+
+	return 0;
 }
 
 /* check whether intel graphics is present */

From 534ad9afb19b5ac810a5e1c638f371e81e0c1ed6 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:41 +0300
Subject: [PATCH 61/70] ALSA: hda - fix CONTROLLER_IN_GPU macro name

The CONTROLLER_IN_GPU() macro has different semantics than
the similarly named macro in hda_intel.c. The name is also
misleading as the macro is used to apply a Intel HSW/BDW
programming logic for HDA controller clock configuration.
Rename macro to reflect the actual implementation.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-5-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_i915.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 50b2c1db429b..d236e497435d 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -13,7 +13,7 @@
 
 static struct completion bind_complete;
 
-#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
+#define IS_HSW_CONTROLLER(pci) (((pci)->device == 0x0a0c) || \
 				((pci)->device == 0x0c0c) || \
 				((pci)->device == 0x0d0c) || \
 				((pci)->device == 0x160c))
@@ -41,7 +41,7 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 
 	if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq)
 		return; /* only for i915 binding */
-	if (!CONTROLLER_IN_GPU(pci))
+	if (!IS_HSW_CONTROLLER(pci))
 		return; /* only HSW/BDW */
 
 	cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev);

From 6564d0ad67efb2d977e130e7448505ee538af016 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 22 Sep 2020 10:49:53 +0200
Subject: [PATCH 62/70] ALSA: ctl: Workaround for lockdep warning wrt
 card->ctl_files_rwlock

The recent change in lockdep for read lock caused the deadlock
warnings in ALSA control code which uses the read_lock() for
notification and else while write_lock_irqsave() is used for adding
and removing the list entry.  Although a deadlock would practically
never hit in a real usage (the addition and the deletion can't happen
with the notification), it's better to fix the read_lock() usage in a
semantically correct way.

This patch replaces the read_lock() calls with read_lock_irqsave()
version for avoiding a reported deadlock.  The notification code path
takes the irq disablement in anyway, and other code paths are very
short execution, hence there shouldn't be any big performance hit by
this change.

Fixes: e918188611f0 ("locking: More accurate annotations for read_lock()")
Reported-by: syzbot+561a74f84100162990b2@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/20200922084953.29018-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/control.c | 18 ++++++++++--------
 1 file changed, 10 insertions(+), 8 deletions(-)

diff --git a/sound/core/control.c b/sound/core/control.c
index e014598142df..421ddc76f264 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -150,14 +150,14 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
 		return;
 	if (card->shutdown)
 		return;
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 #if IS_ENABLED(CONFIG_SND_MIXER_OSS)
 	card->mixer_oss_change_count++;
 #endif
 	list_for_each_entry(ctl, &card->ctl_files, list) {
 		if (!ctl->subscribed)
 			continue;
-		spin_lock_irqsave(&ctl->read_lock, flags);
+		spin_lock(&ctl->read_lock);
 		list_for_each_entry(ev, &ctl->events, list) {
 			if (ev->id.numid == id->numid) {
 				ev->mask |= mask;
@@ -174,10 +174,10 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
 		}
 	_found:
 		wake_up(&ctl->change_sleep);
-		spin_unlock_irqrestore(&ctl->read_lock, flags);
+		spin_unlock(&ctl->read_lock);
 		kill_fasync(&ctl->fasync, SIGIO, POLL_IN);
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 }
 EXPORT_SYMBOL(snd_ctl_notify);
 
@@ -1951,8 +1951,9 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type)
 {
 	struct snd_ctl_file *kctl;
 	int subdevice = -1;
+	unsigned long flags;
 
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 	list_for_each_entry(kctl, &card->ctl_files, list) {
 		if (kctl->pid == task_pid(current)) {
 			subdevice = kctl->preferred_subdevice[type];
@@ -1960,7 +1961,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type)
 				break;
 		}
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 	return subdevice;
 }
 EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice);
@@ -2009,13 +2010,14 @@ static int snd_ctl_dev_disconnect(struct snd_device *device)
 {
 	struct snd_card *card = device->device_data;
 	struct snd_ctl_file *ctl;
+	unsigned long flags;
 
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 	list_for_each_entry(ctl, &card->ctl_files, list) {
 		wake_up(&ctl->change_sleep);
 		kill_fasync(&ctl->fasync, SIGIO, POLL_ERR);
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 
 	return snd_unregister_device(&card->ctl_dev);
 }

From cdc01a1558dedcee3daee7e1802d0349a07edb87 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Franti=C5=A1ek=20Ku=C4=8Dera?= <franta-linux@frantovo.cz>
Date: Tue, 22 Sep 2020 16:42:06 +0200
Subject: [PATCH 63/70] ALSA: usb-audio: Add mixer support for Pioneer DJ
 DJM-250MK2
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This patch extends support for DJM-250MK2 and allows mapping
playback and capture channels to available sources.
Configures the card through USB commands.

Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200922144206.10472-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c | 213 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 213 insertions(+)

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 199cdbfdc761..df036a359f2f 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2602,6 +2602,216 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+/*
+ * Pioneer DJ DJM-250MK2 and maybe other DJM models
+ *
+ * For playback, no duplicate mapping should be set.
+ * There are three mixer stereo channels (CH1, CH2, AUX)
+ * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6).
+ * Each channel should be mapped just once to one source.
+ * If mapped multiple times, only one source will play on given channel
+ * (sources are not mixed together).
+ *
+ * For recording, duplicate mapping is OK. We will get the same signal multiple times.
+ *
+ * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN.
+ *
+ * See also notes in the quirks-table.h file.
+ */
+
+struct snd_pioneer_djm_option {
+	const u16 wIndex;
+	const u16 wValue;
+	const char *name;
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level[] = {
+	{ .name =  "-5 dB",                  .wValue = 0x0300, .wIndex = 0x8003 },
+	{ .name = "-10 dB",                  .wValue = 0x0200, .wIndex = 0x8003 },
+	{ .name = "-15 dB",                  .wValue = 0x0100, .wIndex = 0x8003 },
+	{ .name = "-19 dB",                  .wValue = 0x0000, .wIndex = 0x8003 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch12[] = {
+	{ .name =  "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 },
+	{ .name =  "CH1 Control Tone LINE",  .wValue = 0x0100, .wIndex = 0x8002 },
+	{ .name =  "Post CH1 Fader",         .wValue = 0x0106, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0107, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0108, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0109, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x010d, .wIndex = 0x8002 },
+	{ .name =  "REC OUT",                .wValue = 0x010a, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch34[] = {
+	{ .name =  "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 },
+	{ .name =  "CH2 Control Tone LINE",  .wValue = 0x0200, .wIndex = 0x8002 },
+	{ .name =  "Post CH2 Fader",         .wValue = 0x0206, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0207, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0208, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0209, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x020d, .wIndex = 0x8002 },
+	{ .name =  "REC OUT",                .wValue = 0x020a, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch56[] = {
+	{ .name =  "REC OUT",                .wValue = 0x030a, .wIndex = 0x8002 },
+	{ .name =  "Post CH1 Fader",         .wValue = 0x0311, .wIndex = 0x8002 },
+	{ .name =  "Post CH2 Fader",         .wValue = 0x0312, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0307, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0308, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0309, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x030d, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_12[] = {
+	{ .name =  "CH1",                    .wValue = 0x0100, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0101, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0104, .wIndex = 0x8016 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_34[] = {
+	{ .name =  "CH1",                    .wValue = 0x0200, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0201, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0204, .wIndex = 0x8016 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_56[] = {
+	{ .name =  "CH1",                    .wValue = 0x0300, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0301, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0304, .wIndex = 0x8016 }
+};
+
+struct snd_pioneer_djm_option_group {
+	const char *name;
+	const struct snd_pioneer_djm_option *options;
+	const size_t count;
+	const u16 default_value;
+};
+
+#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \
+	.name = _name, \
+	.options = snd_pioneer_djm_options_##suffix, \
+	.count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \
+	.default_value = _default_value }
+
+static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] = {
+	snd_pioneer_djm_option_group_item("Master Capture Level Capture Switch", capture_level, 0),
+	snd_pioneer_djm_option_group_item("Capture 1-2 Capture Switch",          capture_ch12,  2),
+	snd_pioneer_djm_option_group_item("Capture 3-4 Capture Switch",          capture_ch34,  2),
+	snd_pioneer_djm_option_group_item("Capture 5-6 Capture Switch",          capture_ch56,  0),
+	snd_pioneer_djm_option_group_item("Playback 1-2 Playback Switch",        playback_12,   0),
+	snd_pioneer_djm_option_group_item("Playback 3-4 Playback Switch",        playback_34,   1),
+	snd_pioneer_djm_option_group_item("Playback 5-6 Playback Switch",        playback_56,   2)
+};
+
+// layout of the kcontrol->private_value:
+#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff
+#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000
+#define SND_PIONEER_DJM_GROUP_SHIFT 16
+
+static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info)
+{
+	u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT;
+	size_t count;
+	const char *name;
+	const struct snd_pioneer_djm_option_group *group;
+
+	if (group_index >= ARRAY_SIZE(snd_pioneer_djm_option_groups))
+		return -EINVAL;
+
+	group = &snd_pioneer_djm_option_groups[group_index];
+	count = group->count;
+	if (info->value.enumerated.item >= count)
+		info->value.enumerated.item = count - 1;
+	name = group->options[info->value.enumerated.item].name;
+	strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name));
+	info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	info->count = 1;
+	info->value.enumerated.items = count;
+	return 0;
+}
+
+static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value)
+{
+	int err;
+
+	if (group >= ARRAY_SIZE(snd_pioneer_djm_option_groups)
+			|| value >= snd_pioneer_djm_option_groups[group].count)
+		return -EINVAL;
+
+	err = snd_usb_lock_shutdown(mixer->chip);
+	if (err)
+		return err;
+
+	err = snd_usb_ctl_msg(
+		mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0),
+		USB_REQ_SET_FEATURE,
+		USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+		snd_pioneer_djm_option_groups[group].options[value].wValue,
+		snd_pioneer_djm_option_groups[group].options[value].wIndex,
+		NULL, 0);
+
+	snd_usb_unlock_shutdown(mixer->chip);
+	return err;
+}
+
+static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem)
+{
+	elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK;
+	return 0;
+}
+
+static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	unsigned long private_value = kctl->private_value;
+	u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT;
+	u16 value = elem->value.enumerated.item[0];
+
+	kctl->private_value = (group << SND_PIONEER_DJM_GROUP_SHIFT) | value;
+
+	return snd_pioneer_djm_controls_update(mixer, group, value);
+}
+
+static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list)
+{
+	unsigned long private_value = list->kctl->private_value;
+	u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT;
+	u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK);
+
+	return snd_pioneer_djm_controls_update(list->mixer, group, value);
+}
+
+static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer)
+{
+	int err, i;
+	const struct snd_pioneer_djm_option_group *group;
+	struct snd_kcontrol_new knew = {
+		.iface  = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.index = 0,
+		.info = snd_pioneer_djm_controls_info,
+		.get  = snd_pioneer_djm_controls_get,
+		.put  = snd_pioneer_djm_controls_put
+	};
+
+	for (i = 0; i < ARRAY_SIZE(snd_pioneer_djm_option_groups); i++) {
+		group = &snd_pioneer_djm_option_groups[i];
+		knew.name = group->name;
+		knew.private_value = (i << SND_PIONEER_DJM_GROUP_SHIFT) | group->default_value;
+		err = snd_pioneer_djm_controls_update(mixer, i, group->default_value);
+		if (err)
+			return err;
+		err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume,
+						 &knew, NULL);
+		if (err)
+			return err;
+	}
+	return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -2706,6 +2916,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */
 		err = snd_bbfpro_controls_create(mixer);
 		break;
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
+		err = snd_pioneer_djm_controls_create(mixer);
+		break;
 	}
 
 	return err;

From 2759caad2600d503c3b0ed800e7e03d2cd7a4c05 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 22 Sep 2020 10:38:56 +0200
Subject: [PATCH 64/70] ALSA: seq: oss: Avoid mutex lock for a long-time ioctl

Recently we applied a fix to cover the whole OSS sequencer ioctls with
the mutex for dealing with the possible races.  This works fine in
general, but in theory, this may lead to unexpectedly long stall if an
ioctl like SNDCTL_SEQ_SYNC is issued and an event with the far future
timestamp was queued.

For fixing such a potential stall, this patch changes the mutex lock
applied conditionally excluding such an ioctl command.  Also, change
the mutex_lock() with the interruptible version for user to allow
escaping from the big-hammer mutex.

Fixes: 80982c7e834e ("ALSA: seq: oss: Serialize ioctls")
Suggested-by: Pavel Machek <pavel@ucw.cz>
Link: https://lore.kernel.org/r/20200922083856.28572-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/seq/oss/seq_oss.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index c8b9c0b315d8..250a92b18726 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -174,9 +174,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 	if (snd_BUG_ON(!dp))
 		return -ENXIO;
 
-	mutex_lock(&register_mutex);
+	if (cmd != SNDCTL_SEQ_SYNC &&
+	    mutex_lock_interruptible(&register_mutex))
+		return -ERESTARTSYS;
 	rc = snd_seq_oss_ioctl(dp, cmd, arg);
-	mutex_unlock(&register_mutex);
+	if (cmd != SNDCTL_SEQ_SYNC)
+		mutex_unlock(&register_mutex);
 	return rc;
 }
 

From 502f389a0fd2d987ea7405cb57c02817328b6f7a Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 24 Sep 2020 19:10:27 +0300
Subject: [PATCH 65/70] ALSA: hda - remove kerneldoc for internal hdac_i915
 function

Drop the kerneldoc markup for connectivity_check() as it's an
static helper function. Fixes the following make W=1 warning:

sound/hda/hdac_i915.c:80: warning: Function parameter or member 'i915' not described in 'connectivity_check'
sound/hda/hdac_i915.c:80: warning: Function parameter or member 'hdac' not described in 'connectivity_check'

Fixes: 7b882fe3e3e8 ('ALSA: hda - handle multiple i915 device instances')
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200924161027.3402260-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_i915.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index d236e497435d..5f0a1aa6ad84 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -73,9 +73,7 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
 
-/**
- * Returns true if the devices can be connected for audio.
- */
+/* returns true if the devices can be connected for audio */
 static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac)
 {
 	struct pci_bus *bus_a = i915->bus, *bus_b = hdac->bus;

From 716a0c2881938d222bde67d7edf630ea0648b8f7 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Sun, 25 Nov 2018 23:12:08 +0000
Subject: [PATCH 66/70] ALSA: usb-audio: fix spelling mistake "Frequence" ->
 "Frequency"

There are spelling mistakes in equalizer name fields, fix them.

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20181125231208.14350-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_us16x08.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
index a4d4d71db55b..92b1a6d9c931 100644
--- a/sound/usb/mixer_us16x08.c
+++ b/sound/usb/mixer_us16x08.c
@@ -1109,7 +1109,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQLOWFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ Low Frequence",
+		.name = "EQ Low Frequency",
 	},
 	{ /* EQ mid low gain */
 		.kcontrol_new = &snd_us16x08_eq_gain_ctl,
@@ -1123,7 +1123,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQLOWMIDFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ MidLow Frequence",
+		.name = "EQ MidLow Frequency",
 	},
 	{ /* EQ mid low Q */
 		.kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
@@ -1144,7 +1144,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQHIGHMIDFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ MidHigh Frequence",
+		.name = "EQ MidHigh Frequency",
 	},
 	{ /* EQ mid high Q */
 		.kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
@@ -1165,7 +1165,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQHIGHFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ High Frequence",
+		.name = "EQ High Frequency",
 	},
 };
 

From 86b9c4cdd76fcfc98c5967f63b4ca09620e67968 Mon Sep 17 00:00:00 2001
From: Randy Dunlap <rdunlap@infradead.org>
Date: Mon, 5 Oct 2020 12:12:23 -0700
Subject: [PATCH 67/70] ALSA: portman2x4: fix repeated word 'if'

Correct duplicated word "if" to "if it".

Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20201005191223.21514-1-rdunlap@infradead.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/portman2x4.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 38603cb2bd5b..c876cf9b5005 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -467,7 +467,7 @@ static int portman_probe(struct parport *p)
 	parport_write_control(p, 0);	/* Reset Strobe=0. */
 
 	/* Check if Tx circuitry is functioning properly.  If initialized 
-	 * unit TxEmpty is false, send out char and see if if goes true.
+	 * unit TxEmpty is false, send out char and see if it goes true.
 	 */
 	/* 8 */
 	parport_write_control(p, TXDATA0);	/* Tx channel 0, strobe off. */

From 0569b3d8ae17c29f8e09db2ba683be3915fe8b47 Mon Sep 17 00:00:00 2001
From: Randy Dunlap <rdunlap@infradead.org>
Date: Mon, 5 Oct 2020 12:12:44 -0700
Subject: [PATCH 68/70] ALSA: usb-audio: endpoint.c: fix repeated word 'there'

Drop the duplicated word "there".

Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20201005191244.23902-1-rdunlap@infradead.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/endpoint.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 5fbc8dd2f409..e2f9ce2f5b8b 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -318,7 +318,7 @@ static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep,
 
 /*
  * Send output urbs that have been prepared previously. URBs are dequeued
- * from ep->ready_playback_urbs and in case there there aren't any available
+ * from ep->ready_playback_urbs and in case there aren't any available
  * or there are no packets that have been prepared, this function does
  * nothing.
  *

From 7dcd56123e312e8b17f85a597b33552a704ce45f Mon Sep 17 00:00:00 2001
From: Naoki Hayama <naoki.hayama@lineo.co.jp>
Date: Thu, 8 Oct 2020 17:47:44 +0900
Subject: [PATCH 69/70] ALSA: hdspm: Fix typo arbitary

Fix comment typo.
s/arbitary/arbitrary/

Signed-off-by: Naoki Hayama <naoki.hayama@lineo.co.jp>
Link: https://lore.kernel.org/r/e04a8c5b-8c59-3f02-34d3-c1a871d08cc2@lineo.co.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index e532312a5e1c..4a1f576dd9cf 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1217,7 +1217,7 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm)
 	return ret;
 }
 
-/* round arbitary sample rates to commonly known rates */
+/* round arbitrary sample rates to commonly known rates */
 static int hdspm_round_frequency(int rate)
 {
 	if (rate < 38050)

From 96e503f9000f2ad17d550cd884a5e386eb7f532f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Oct 2020 19:17:22 +0300
Subject: [PATCH 70/70] ALSA: hda/i915 - fix list corruption with concurrent
 probes

Current hdac_i915 uses a static completion instance to wait
for i915 driver to complete the component bind.

This design is not safe if multiple HDA controllers are active and
communicating with different i915 instances, and can lead to list
corruption and failed audio driver probe.

Fix the design by moving completion mechanism to common acomp
code and remove the related code from hdac_i915.

Fixes: 7b882fe3e3e8 ("ALSA: hda - handle multiple i915 device instances")
Co-developed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20201006161722.500256-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/drm/drm_audio_component.h |  4 ++++
 sound/hda/hdac_component.c        |  3 +++
 sound/hda/hdac_i915.c             | 23 +++--------------------
 3 files changed, 10 insertions(+), 20 deletions(-)

diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h
index a45f93487039..0d36bfd1a4cd 100644
--- a/include/drm/drm_audio_component.h
+++ b/include/drm/drm_audio_component.h
@@ -117,6 +117,10 @@ struct drm_audio_component {
 	 * @audio_ops: Ops implemented by hda driver, called by DRM driver
 	 */
 	const struct drm_audio_component_audio_ops *audio_ops;
+	/**
+	 * @master_bind_complete: completion held during component master binding
+	 */
+	struct completion master_bind_complete;
 };
 
 #endif /* _DRM_AUDIO_COMPONENT_H_ */
diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c
index 89126c6fd216..bb37e7e0bd79 100644
--- a/sound/hda/hdac_component.c
+++ b/sound/hda/hdac_component.c
@@ -210,12 +210,14 @@ static int hdac_component_master_bind(struct device *dev)
 			goto module_put;
 	}
 
+	complete_all(&acomp->master_bind_complete);
 	return 0;
 
  module_put:
 	module_put(acomp->ops->owner);
 out_unbind:
 	component_unbind_all(dev, acomp);
+	complete_all(&acomp->master_bind_complete);
 
 	return ret;
 }
@@ -296,6 +298,7 @@ int snd_hdac_acomp_init(struct hdac_bus *bus,
 	if (!acomp)
 		return -ENOMEM;
 	acomp->audio_ops = aops;
+	init_completion(&acomp->master_bind_complete);
 	bus->audio_component = acomp;
 	devres_add(dev, acomp);
 
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 5f0a1aa6ad84..454474ac5716 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -11,8 +11,6 @@
 #include <sound/hda_i915.h>
 #include <sound/hda_register.h>
 
-static struct completion bind_complete;
-
 #define IS_HSW_CONTROLLER(pci) (((pci)->device == 0x0a0c) || \
 				((pci)->device == 0x0c0c) || \
 				((pci)->device == 0x0d0c) || \
@@ -130,19 +128,6 @@ static bool i915_gfx_present(void)
 	return pci_dev_present(ids);
 }
 
-static int i915_master_bind(struct device *dev,
-			    struct drm_audio_component *acomp)
-{
-	complete_all(&bind_complete);
-	/* clear audio_ops here as it was needed only for completion call */
-	acomp->audio_ops = NULL;
-	return 0;
-}
-
-static const struct drm_audio_component_audio_ops i915_init_ops = {
-	.master_bind = i915_master_bind
-};
-
 /**
  * snd_hdac_i915_init - Initialize i915 audio component
  * @bus: HDA core bus
@@ -163,9 +148,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 	if (!i915_gfx_present())
 		return -ENODEV;
 
-	init_completion(&bind_complete);
-
-	err = snd_hdac_acomp_init(bus, &i915_init_ops,
+	err = snd_hdac_acomp_init(bus, NULL,
 				  i915_component_master_match,
 				  sizeof(struct i915_audio_component) - sizeof(*acomp));
 	if (err < 0)
@@ -177,8 +160,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 		if (!IS_ENABLED(CONFIG_MODULES) ||
 		    !request_module("i915")) {
 			/* 60s timeout */
-			wait_for_completion_timeout(&bind_complete,
-						   msecs_to_jiffies(60 * 1000));
+			wait_for_completion_timeout(&acomp->master_bind_complete,
+						    msecs_to_jiffies(60 * 1000));
 		}
 	}
 	if (!acomp->ops) {