From c1c6c877b0c79fd7e05c931435aa42211eaeebaf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Aug 2019 14:03:56 +0200
Subject: [PATCH 01/22] ALSA: hda - Don't override global PCM hw info flag

The commit bfcba288b97f ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object.  This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.

Fix the bug by setting the PCM hw info flag locally.

Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index c8d1b4316245..2fbdde239936 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	}
 	runtime->private_data = azx_dev;
 
-	if (chip->gts_present)
-		azx_pcm_hw.info = azx_pcm_hw.info |
-			SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
-
 	runtime->hw = azx_pcm_hw;
+	if (chip->gts_present)
+		runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;

From 3d92aa45fbfd7319e3a19f4ec59fd32b3862b723 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Wed, 7 Aug 2019 04:08:51 -0500
Subject: [PATCH 02/22] ALSA: hiface: fix multiple memory leak bugs

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc().  However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.

To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.

Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/hiface/pcm.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c
index 14fc1e1d5d13..c406497c5919 100644
--- a/sound/usb/hiface/pcm.c
+++ b/sound/usb/hiface/pcm.c
@@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 		ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP,
 				    hiface_pcm_out_urb_handler);
 		if (ret < 0)
-			return ret;
+			goto error;
 	}
 
 	ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm);
 	if (ret < 0) {
-		kfree(rt);
 		dev_err(&chip->dev->dev, "Cannot create pcm instance\n");
-		return ret;
+		goto error;
 	}
 
 	pcm->private_data = rt;
@@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 
 	chip->pcm = rt;
 	return 0;
+
+error:
+	for (i = 0; i < PCM_N_URBS; i++)
+		kfree(rt->out_urbs[i].buffer);
+	kfree(rt);
+	return ret;
 }

From c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Aug 2019 17:31:48 +0200
Subject: [PATCH 03/22] ALSA: hda - Workaround for crackled sound on AMD
 controller (1022:1457)

A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:

- Set up the proper driver caps for this controller, similar as the
  other AMD controller.

- Correct the DMA position reporting with the fixed FIFO size, which
  is similar like as workaround used for VIA chip set.

- Even after the position correction, PulseAudio still shows
  mysterious stalls of playback streams when a capture is triggered in
  timer-scheduled mode.  Since we have no clear way to eliminate the
  stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
  a temporary workaround.

This patch implements the workarounds.  For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB.  It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.

Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c |  7 ++++
 sound/pci/hda/hda_controller.h |  2 +-
 sound/pci/hda/hda_intel.c      | 63 +++++++++++++++++++++++++++++++++-
 3 files changed, 70 insertions(+), 2 deletions(-)

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2fbdde239936..48d863736b3c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -613,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
+	/* by some reason, the playback stream stalls on PulseAudio with
+	 * tsched=1 when a capture stream triggers.  Until we figure out the
+	 * real cause, disable tsched mode by telling the PCM info flag.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
+		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index baa15374fbcb..f2a6df5e6bcb 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -31,7 +31,7 @@
 /* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-/* 17 unused */
+#define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1e14d7270adf..a6d8c0d77b84 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -64,6 +64,7 @@ enum {
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
 	POS_FIX_SKL,
+	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -332,6 +333,11 @@ enum {
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
 	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
+/* quirks for AMD SB */
+#define AZX_DCAPS_PRESET_AMD_SB \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\
@@ -841,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+#define AMD_FIFO_SIZE	32
+
+/* get the current DMA position with FIFO size correction */
+static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int pos, delay;
+
+	pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev));
+	if (!runtime)
+		return pos;
+
+	runtime->delay = AMD_FIFO_SIZE;
+	delay = frames_to_bytes(runtime, AMD_FIFO_SIZE);
+	if (azx_dev->insufficient) {
+		if (pos < delay) {
+			delay = pos;
+			runtime->delay = bytes_to_frames(runtime, pos);
+		} else {
+			azx_dev->insufficient = 0;
+		}
+	}
+
+	/* correct the DMA position for capture stream */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (pos < delay)
+			pos += azx_dev->core.bufsize;
+		pos -= delay;
+	}
+
+	return pos;
+}
+
+static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev,
+				   unsigned int pos)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+
+	/* just read back the calculated value in the above */
+	return substream->runtime->delay;
+}
+
 static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
 					 struct azx_dev *azx_dev)
 {
@@ -1417,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
 	case POS_FIX_SKL:
+	case POS_FIX_FIFO:
 		return fix;
 	}
 
@@ -1433,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) {
+		dev_dbg(chip->card->dev, "Using FIFO position fix\n");
+		return POS_FIX_FIFO;
+	}
 	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
@@ -1453,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
 		[POS_FIX_SKL] = azx_get_pos_skl,
+		[POS_FIX_FIFO] = azx_get_pos_fifo,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1467,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix)
 			azx_get_delay_from_lpib;
 	}
 
+	if (fix == POS_FIX_FIFO)
+		chip->get_delay[0] = chip->get_delay[1] =
+			azx_get_delay_from_fifo;
 }
 
 /*
@@ -2447,6 +2505,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD, X370 & co */
+	{ PCI_DEVICE(0x1022, 0x1457),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |

From c7cd7c748a3250ca33509f9235efab9c803aca09 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Thu, 8 Aug 2019 00:15:21 -0500
Subject: [PATCH 04/22] sound: fix a memory leak bug

In sound_insert_unit(), the controlling structure 's' is allocated through
kmalloc(). Then it is added to the sound driver list by invoking
__sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is
removed from the list through __sound_remove_unit(). If 'index' is not less
than 0, -EBUSY is returned to indicate the error. However, 's' is not
deallocated on this execution path, leading to a memory leak bug.

To fix the above issue, free 's' before -EBUSY is returned.

Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/sound_core.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/sound_core.c b/sound/sound_core.c
index b730d97c4de6..90d118cd9164 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -275,7 +275,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati
 				goto retry;
 			}
 			spin_unlock(&sound_loader_lock);
-			return -EBUSY;
+			r = -EBUSY;
+			goto fail;
 		}
 	}
 

From 1be3c1fae6c1e1f5bb982b255d2034034454527a Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Thu, 8 Aug 2019 00:50:58 -0500
Subject: [PATCH 05/22] ALSA: firewire: fix a memory leak bug

In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.

To fix the above issue, free 'b->packets' before returning the error code.

Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v2.6.39+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/packets-buffer.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
index 0d35359d25cd..0ecafd0c6722 100644
--- a/sound/firewire/packets-buffer.c
+++ b/sound/firewire/packets-buffer.c
@@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
 	packets_per_page = PAGE_SIZE / packet_size;
 	if (WARN_ON(!packets_per_page)) {
 		err = -EINVAL;
-		goto error;
+		goto err_packets;
 	}
 	pages = DIV_ROUND_UP(count, packets_per_page);
 

From de768ce45466f3009809719eb7b1f6f5277d9373 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Aug 2019 11:23:00 +0200
Subject: [PATCH 06/22] ALSA: hda - Apply workaround for another AMD chip
 1022:1487

MSI MPG X570 board is with another AMD HD-audio controller (PCI ID
1022:1487) and it requires the same workaround applied for X370, etc
(PCI ID 1022:1457).

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a6d8c0d77b84..99fc0917339b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,6 +2508,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD, X370 & co */
 	{ PCI_DEVICE(0x1022, 0x1457),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
+	/* AMD, X570 & co */
+	{ PCI_DEVICE(0x1022, 0x1487),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |

From cfef67f016e4c00a2f423256fc678a6967a9fc09 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Fri, 9 Aug 2019 23:29:48 -0500
Subject: [PATCH 07/22] ALSA: hda - Fix a memory leak bug

In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc().
Then, the pin widgets in 'codec' are parsed. However, if the parsing
process fails, 'spec' is not deallocated, leading to a memory leak.

To fix the above issue, free 'spec' before returning the error.

Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 485edaba0037..8f2beb1f3ae4 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6100,7 +6100,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec)
 
 	err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0);
 	if (err < 0)
-		return err;
+		goto error;
 
 	err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg);
 	if (err < 0)

From 190d03814eb3b49d4f87ff38fef26d36f3568a60 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Aug 2019 17:39:56 +0200
Subject: [PATCH 08/22] ALSA: hda/realtek - Add quirk for HP Envy x360

HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same
quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index de224cbea7a0..8aaf1d9c55cf 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6987,6 +6987,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),

From 401714d9534aad8c24196b32600da683116bbe09 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 14 Aug 2019 12:09:07 +0800
Subject: [PATCH 09/22] ALSA: hda - Let all conexant codec enter D3 when
 rebooting

We have 3 new lenovo laptops which have conexant codec 0x14f11f86,
these 3 laptops also have the noise issue when rebooting, after
letting the codec enter D3 before rebooting or poweroff, the noise
disappers.

Instead of adding a new ID again in the reboot_notify(), let us make
this function apply to all conexant codec. In theory make codec enter
D3 before rebooting or poweroff is harmless, and I tested this change
on a couple of other Lenovo laptops which have different conexant
codecs, there is no side effect so far.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 9 ---------
 1 file changed, 9 deletions(-)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f299f137eaea..93a303676aea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -163,15 +163,6 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	switch (codec->core.vendor_id) {
-	case 0x14f12008: /* CX8200 */
-	case 0x14f150f2: /* CX20722 */
-	case 0x14f150f4: /* CX20724 */
-		break;
-	default:
-		return;
-	}
-
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);

From 871b9066027702e6e6589da0e1edd3b7dede7205 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 14 Aug 2019 12:09:08 +0800
Subject: [PATCH 10/22] ALSA: hda - Add a generic reboot_notify

Make codec enter D3 before rebooting or poweroff can fix the noise
issue on some laptops. And in theory it is harmless for all codecs
to enter D3 before rebooting or poweroff, let us add a generic
reboot_notify, then realtek and conexant drivers can call this
function.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c    | 19 +++++++++++++++++++
 sound/pci/hda/hda_generic.h    |  1 +
 sound/pci/hda/patch_conexant.c |  6 +-----
 sound/pci/hda/patch_realtek.c  | 11 +----------
 4 files changed, 22 insertions(+), 15 deletions(-)

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8f2beb1f3ae4..5bf24fb819d2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6051,6 +6051,24 @@ void snd_hda_gen_free(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
+/**
+ * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops reboot_notify function.
+ */
+void snd_hda_gen_reboot_notify(struct hda_codec *codec)
+{
+	/* Make the codec enter D3 to avoid spurious noises from the internal
+	 * speaker during (and after) reboot
+	 */
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify);
+
 #ifdef CONFIG_PM
 /**
  * snd_hda_gen_check_power_status - check the loopback power save state
@@ -6078,6 +6096,7 @@ static const struct hda_codec_ops generic_patch_ops = {
 	.init = snd_hda_gen_init,
 	.free = snd_hda_gen_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+	.reboot_notify = snd_hda_gen_reboot_notify,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
 #endif
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 35a670a71c42..5f199dcb0d18 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -332,6 +332,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 				  struct auto_pin_cfg *cfg);
 int snd_hda_gen_build_controls(struct hda_codec *codec);
 int snd_hda_gen_build_pcms(struct hda_codec *codec);
+void snd_hda_gen_reboot_notify(struct hda_codec *codec);
 
 /* standard jack event callbacks */
 void snd_hda_gen_hp_automute(struct hda_codec *codec,
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 93a303676aea..14298ef45b21 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -166,11 +166,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
-
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
+	snd_hda_gen_reboot_notify(codec);
 }
 
 static void cx_auto_free(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8aaf1d9c55cf..e333b3e30e31 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -869,15 +869,6 @@ static void alc_reboot_notify(struct hda_codec *codec)
 		alc_shutup(codec);
 }
 
-/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
-static void alc_d3_at_reboot(struct hda_codec *codec)
-{
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
-}
-
 #define alc_free	snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -5152,7 +5143,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
+		spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		codec->power_save_node = 0; /* avoid click noises */
 		snd_hda_apply_pincfgs(codec, pincfgs);

From daac07156b330b18eb5071aec4b3ddca1c377f2c Mon Sep 17 00:00:00 2001
From: Hui Peng <benquike@gmail.com>
Date: Tue, 13 Aug 2019 22:34:04 -0400
Subject: [PATCH 11/22] ALSA: usb-audio: Fix an OOB bug in
 parse_audio_mixer_unit

The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length  of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.

```
struct uac_mixer_unit_descriptor {
	__u8 bLength;
	__u8 bDescriptorType;
	__u8 bDescriptorSubtype;
	__u8 bUnitID;
	__u8 bNrInPins;
	__u8 baSourceID[];
}
```

This patch fixes the bug by add a sanity check on the length of
the descriptor.

Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7498b5191b68..ea487378be17 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -744,6 +744,8 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 		return -EINVAL;
 	if (!desc->bNrInPins)
 		return -EINVAL;
+	if (desc->bLength < sizeof(*desc) + desc->bNrInPins)
+		return -EINVAL;
 
 	switch (state->mixer->protocol) {
 	case UAC_VERSION_1:

From 19bce474c45be69a284ecee660aa12d8f1e88f18 Mon Sep 17 00:00:00 2001
From: Hui Peng <benquike@gmail.com>
Date: Thu, 15 Aug 2019 00:31:34 -0400
Subject: [PATCH 12/22] ALSA: usb-audio: Fix a stack buffer overflow bug in
 check_input_term

`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.

This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).

Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 35 +++++++++++++++++++++++++++--------
 1 file changed, 27 insertions(+), 8 deletions(-)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ea487378be17..b5927c3d5bc0 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -68,6 +68,7 @@ struct mixer_build {
 	unsigned char *buffer;
 	unsigned int buflen;
 	DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS);
+	DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS);
 	struct usb_audio_term oterm;
 	const struct usbmix_name_map *map;
 	const struct usbmix_selector_map *selector_map;
@@ -775,16 +776,25 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
  * parse the source unit recursively until it reaches to a terminal
  * or a branched unit.
  */
-static int check_input_term(struct mixer_build *state, int id,
+static int __check_input_term(struct mixer_build *state, int id,
 			    struct usb_audio_term *term)
 {
 	int protocol = state->mixer->protocol;
 	int err;
 	void *p1;
+	unsigned char *hdr;
 
 	memset(term, 0, sizeof(*term));
-	while ((p1 = find_audio_control_unit(state, id)) != NULL) {
-		unsigned char *hdr = p1;
+	for (;;) {
+		/* a loop in the terminal chain? */
+		if (test_and_set_bit(id, state->termbitmap))
+			return -EINVAL;
+
+		p1 = find_audio_control_unit(state, id);
+		if (!p1)
+			break;
+
+		hdr = p1;
 		term->id = id;
 
 		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
@@ -802,7 +812,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 					/* call recursively to verify that the
 					 * referenced clock entity is valid */
-					err = check_input_term(state, d->bCSourceID, term);
+					err = __check_input_term(state, d->bCSourceID, term);
 					if (err < 0)
 						return err;
 
@@ -836,7 +846,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC2_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -899,7 +909,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 				/* call recursively to verify that the
 				 * referenced clock entity is valid */
-				err = check_input_term(state, d->bCSourceID, term);
+				err = __check_input_term(state, d->bCSourceID, term);
 				if (err < 0)
 					return err;
 
@@ -950,7 +960,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC3_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -966,7 +976,7 @@ static int check_input_term(struct mixer_build *state, int id,
 					return -EINVAL;
 
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 
@@ -984,6 +994,15 @@ static int check_input_term(struct mixer_build *state, int id,
 	return -ENODEV;
 }
 
+
+static int check_input_term(struct mixer_build *state, int id,
+			    struct usb_audio_term *term)
+{
+	memset(term, 0, sizeof(*term));
+	memset(state->termbitmap, 0, sizeof(state->termbitmap));
+	return __check_input_term(state, id, term);
+}
+
 /*
  * Feature Unit
  */

From f9ef724d4896763479f3921afd1ee61552fc9836 Mon Sep 17 00:00:00 2001
From: Jeronimo Borque <jeronimo@borque.com.ar>
Date: Sun, 18 Aug 2019 22:35:38 -0300
Subject: [PATCH 13/22] ALSA: hda - Fixes inverted Conexant GPIO mic mute led

"enabled" parameter historically referred to the device input or
output, not to the led indicator. After the changes added with the led
helper functions the mic mute led logic refers to the led and not to
the mic input which caused led indicator to be negated.
Fixing logic in cxt_update_gpio_led and updated
cxt_fixup_gpio_mute_hook
Also updated debug messages to ease further debugging if necessary.

Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 17 +++++++++--------
 1 file changed, 9 insertions(+), 8 deletions(-)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 14298ef45b21..968d3caab6ac 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -611,18 +611,20 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec,
 
 /* update LED status via GPIO */
 static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask,
-				bool enabled)
+				bool led_on)
 {
 	struct conexant_spec *spec = codec->spec;
 	unsigned int oldval = spec->gpio_led;
 
 	if (spec->mute_led_polarity)
-		enabled = !enabled;
+		led_on = !led_on;
 
-	if (enabled)
-		spec->gpio_led &= ~mask;
-	else
+	if (led_on)
 		spec->gpio_led |= mask;
+	else
+		spec->gpio_led &= ~mask;
+	codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n",
+			mask, led_on, spec->gpio_led);
 	if (spec->gpio_led != oldval)
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 				    spec->gpio_led);
@@ -633,8 +635,8 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
 {
 	struct hda_codec *codec = private_data;
 	struct conexant_spec *spec = codec->spec;
-
-	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+	/* muted -> LED on */
+	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled);
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
@@ -656,7 +658,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
 		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 },
 		{}
 	};
-	codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led);
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;

From 1a15718b41df026cffd0e42cfdc38a1384ce19f9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Aug 2019 08:58:12 +0200
Subject: [PATCH 14/22] ALSA: usb-audio: Add implicit fb quirk for Behringer
 UFX1604

Behringer UFX1604 requires the similar quirk to apply implicit fb like
another Behringer model UFX1204 in order to fix the noisy playback.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 75b96929f76c..e4bbf79de956 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */
 	case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
 		ep = 0x81;
 		ifnum = 1;

From 2ca371d847511f97ef991ef612a2ce805489840e Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Pawe=C5=82=20Rekowski?= <p.rekowski@gmail.com>
Date: Mon, 19 Aug 2019 22:40:07 +0200
Subject: [PATCH 15/22] ALSA: hda/ca0132 - Add new SBZ quirk
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This patch adds a new PCI subsys ID for the SBZ, as found and tested by
me and some reddit users.

Link: https://lore.kernel.org/lkml/20190819204008.14426-1-p.rekowski@gmail.com
Signed-off-by: Paweł Rekowski <p.rekowski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0d51823d7270..6d1fb7c11f17 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE),
 	SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ),
+	SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),

From 6de3c9e3f6b3eaf66859e1379b3f35dda781416b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 15 Aug 2019 11:41:06 +0200
Subject: [PATCH 16/22] ALSA: usb-audio: Fix invalid NULL check in
 snd_emuusb_set_samplerate()

The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code.  It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.

This patch fixes the check before the conversion.

While we're at it, correct a typo in the comment in the function,
too.

Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 199fa157a411..27dcb3743690 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 {
 	struct usb_mixer_interface *mixer;
 	struct usb_mixer_elem_info *cval;
-	int unitid = 12; /* SamleRate ExtensionUnit ID */
+	int unitid = 12; /* SampleRate ExtensionUnit ID */
 
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
-		cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
-		if (cval) {
+		if (mixer->id_elems[unitid]) {
+			cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
 			snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR,
 						    cval->control << 8,
 						    samplerate_id);
 			snd_usb_mixer_notify_id(mixer, unitid);
+			break;
 		}
-		break;
 	}
 }
 

From 1bc8d18c75fef3b478dbdfef722aae09e2a9fde7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 21 Aug 2019 20:00:02 +0200
Subject: [PATCH 17/22] ALSA: line6: Fix memory leak at line6_init_pcm() error
 path

I forgot to release the allocated object at the early error path in
line6_init_pcm().  For addressing it, slightly shuffle the code so
that the PCM destructor (pcm->private_free) is assigned properly
before all error paths.

Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/line6/pcm.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c
index 2c03e0f6bf72..f70211e6b174 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6,
 	line6pcm->volume_monitor = 255;
 	line6pcm->line6 = line6;
 
+	spin_lock_init(&line6pcm->out.lock);
+	spin_lock_init(&line6pcm->in.lock);
+	line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD;
+
+	line6->line6pcm = line6pcm;
+
+	pcm->private_data = line6pcm;
+	pcm->private_free = line6_cleanup_pcm;
+
 	line6pcm->max_packet_size_in =
 		usb_maxpacket(line6->usbdev,
 			usb_rcvisocpipe(line6->usbdev, ep_read), 0);
@@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6,
 		return -EINVAL;
 	}
 
-	spin_lock_init(&line6pcm->out.lock);
-	spin_lock_init(&line6pcm->in.lock);
-	line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD;
-
-	line6->line6pcm = line6pcm;
-
-	pcm->private_data = line6pcm;
-	pcm->private_free = line6_cleanup_pcm;
-
 	err = line6_create_audio_out_urbs(line6pcm);
 	if (err < 0)
 		return err;

From f9f0e9ed350e15d51ad07364b4cf910de50c472a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Aug 2019 21:43:42 +0200
Subject: [PATCH 18/22] ALSA: usb-audio: Check mixer unit bitmap yet more
 strictly

The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a
variable size depending on both input and output pins.  Its size is to
fit with input * output bits.  The problem is that the input size
can't be determined simply from the unit descriptor itself but it
needs to parse the whole connected sources.  Although the
uac_mixer_unit_get_channels() tries to check some possible overflow of
this bitmap, it's incomplete due to the lack of the  evaluation of
input pins.

For covering possible overflows, this patch adds the bitmap overflow
check in the loop of input pins in parse_audio_mixer_unit().

Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 36 ++++++++++++++++++++++++++++--------
 1 file changed, 28 insertions(+), 8 deletions(-)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index b5927c3d5bc0..eceab19766db 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -739,7 +739,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 				       struct uac_mixer_unit_descriptor *desc)
 {
 	int mu_channels;
-	void *c;
 
 	if (desc->bLength < sizeof(*desc))
 		return -EINVAL;
@@ -762,13 +761,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 		break;
 	}
 
-	if (!mu_channels)
-		return 0;
-
-	c = uac_mixer_unit_bmControls(desc, state->mixer->protocol);
-	if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength)
-		return 0; /* no bmControls -> skip */
-
 	return mu_channels;
 }
 
@@ -2009,6 +2001,31 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid,
  * Mixer Unit
  */
 
+/* check whether the given in/out overflows bmMixerControls matrix */
+static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc,
+				  int protocol, int num_ins, int num_outs)
+{
+	u8 *hdr = (u8 *)desc;
+	u8 *c = uac_mixer_unit_bmControls(desc, protocol);
+	size_t rest; /* remaining bytes after bmMixerControls */
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	default:
+		rest = 1; /* iMixer */
+		break;
+	case UAC_VERSION_2:
+		rest = 2; /* bmControls + iMixer */
+		break;
+	case UAC_VERSION_3:
+		rest = 6; /* bmControls + wMixerDescrStr */
+		break;
+	}
+
+	/* overflow? */
+	return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0];
+}
+
 /*
  * build a mixer unit control
  *
@@ -2137,6 +2154,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid,
 		if (err < 0)
 			return err;
 		num_ins += iterm.channels;
+		if (mixer_bitmap_overflow(desc, state->mixer->protocol,
+					  num_ins, num_outs))
+			break;
 		for (; ich < num_ins; ich++) {
 			int och, ich_has_controls = 0;
 

From 57f8770620e9b51c61089751f0b5ad3dbe376ff2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Aug 2019 17:17:09 +0200
Subject: [PATCH 19/22] ALSA: usb-audio: More validations of descriptor units

Introduce a new helper to validate each audio descriptor unit before
and check the unit before actually accessing it.  This should harden
against the OOB access cases with malformed descriptors that have been
recently frequently reported by fuzzers.

The existing descriptor checks are still kept although they become
superfluous after this patch.  They'll be cleaned up eventually
later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/Makefile   |   3 +-
 sound/usb/helper.h   |   4 +
 sound/usb/mixer.c    |  10 ++
 sound/usb/power.c    |   2 +
 sound/usb/quirks.c   |   3 +
 sound/usb/stream.c   |  25 ++--
 sound/usb/validate.c | 332 +++++++++++++++++++++++++++++++++++++++++++
 7 files changed, 366 insertions(+), 13 deletions(-)
 create mode 100644 sound/usb/validate.c

diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index e1ce257ab705..d27a21b0ff9c 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -16,7 +16,8 @@ snd-usb-audio-objs := 	card.o \
 			power.o \
 			proc.o \
 			quirks.o \
-			stream.o
+			stream.o \
+			validate.o
 
 snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o
 
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index 6afb70156ec4..5e8a18b4e7b9 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -31,4 +31,8 @@ static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
 	return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
 }
 
+/* in validate.c */
+bool snd_usb_validate_audio_desc(void *p, int protocol);
+bool snd_usb_validate_midi_desc(void *p);
+
 #endif /* __USBAUDIO_HELPER_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index eceab19766db..a1093fb9bf09 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -785,6 +785,8 @@ static int __check_input_term(struct mixer_build *state, int id,
 		p1 = find_audio_control_unit(state, id);
 		if (!p1)
 			break;
+		if (!snd_usb_validate_audio_desc(p1, protocol))
+			break; /* bad descriptor */
 
 		hdr = p1;
 		term->id = id;
@@ -2775,6 +2777,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 		return -EINVAL;
 	}
 
+	if (!snd_usb_validate_audio_desc(p1, protocol)) {
+		usb_audio_dbg(state->chip, "invalid unit %d\n", unitid);
+		return 0; /* skip invalid unit */
+	}
+
 	if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
 		switch (p1[2]) {
 		case UAC_INPUT_TERMINAL:
@@ -3145,6 +3152,9 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	while ((p = snd_usb_find_csint_desc(mixer->hostif->extra,
 					    mixer->hostif->extralen,
 					    p, UAC_OUTPUT_TERMINAL)) != NULL) {
+		if (!snd_usb_validate_audio_desc(p, mixer->protocol))
+			continue; /* skip invalid descriptor */
+
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
diff --git a/sound/usb/power.c b/sound/usb/power.c
index bd303a1ba1b7..606a2cb23eab 100644
--- a/sound/usb/power.c
+++ b/sound/usb/power.c
@@ -31,6 +31,8 @@ snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface,
 		struct uac3_power_domain_descriptor *pd_desc = p;
 		int i;
 
+		if (!snd_usb_validate_audio_desc(p, UAC_VERSION_3))
+			continue;
 		for (i = 0; i < pd_desc->bNrEntities; i++) {
 			if (pd_desc->baEntityID[i] == id) {
 				pd->pd_id = pd_desc->bPowerDomainID;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 78858918cbc1..7e9735aa7ac9 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -248,6 +248,9 @@ static int create_yamaha_midi_quirk(struct snd_usb_audio *chip,
 					NULL, USB_MS_MIDI_OUT_JACK);
 	if (!injd && !outjd)
 		return -ENODEV;
+	if (!snd_usb_validate_midi_desc(injd) ||
+	    !snd_usb_validate_midi_desc(outjd))
+		return -ENODEV;
 	if (injd && (injd->bLength < 5 ||
 		     (injd->bJackType != USB_MS_EMBEDDED &&
 		      injd->bJackType != USB_MS_EXTERNAL)))
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index e852c7fd6109..a0649c8ae460 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -627,16 +627,14 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
  */
 static void *
 snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-				       int terminal_id, bool uac23)
+				       int terminal_id, int protocol)
 {
 	struct uac2_input_terminal_descriptor *term = NULL;
-	size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) :
-		sizeof(struct uac_input_terminal_descriptor);
 
 	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
 					       ctrl_iface->extralen,
 					       term, UAC_INPUT_TERMINAL))) {
-		if (term->bLength < minlen)
+		if (!snd_usb_validate_audio_desc(term, protocol))
 			continue;
 		if (term->bTerminalID == terminal_id)
 			return term;
@@ -647,7 +645,7 @@ snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
 
 static void *
 snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-					int terminal_id)
+					int terminal_id, int protocol)
 {
 	/* OK to use with both UAC2 and UAC3 */
 	struct uac2_output_terminal_descriptor *term = NULL;
@@ -655,8 +653,9 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
 	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
 					       ctrl_iface->extralen,
 					       term, UAC_OUTPUT_TERMINAL))) {
-		if (term->bLength >= sizeof(*term) &&
-		    term->bTerminalID == terminal_id)
+		if (!snd_usb_validate_audio_desc(term, protocol))
+			continue;
+		if (term->bTerminalID == terminal_id)
 			return term;
 	}
 
@@ -731,7 +730,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 
 		iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 							       as->bTerminalLink,
-							       false);
+							       protocol);
 		if (iterm) {
 			num_channels = iterm->bNrChannels;
 			chconfig = le16_to_cpu(iterm->wChannelConfig);
@@ -767,7 +766,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 		 */
 		input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 								    as->bTerminalLink,
-								    true);
+								    protocol);
 		if (input_term) {
 			clock = input_term->bCSourceID;
 			if (!chconfig && (num_channels == input_term->bNrChannels))
@@ -776,7 +775,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 		}
 
 		output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-								      as->bTerminalLink);
+								      as->bTerminalLink,
+								      protocol);
 		if (output_term) {
 			clock = output_term->bCSourceID;
 			goto found_clock;
@@ -1002,14 +1002,15 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
 	 */
 	input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 							    as->bTerminalLink,
-							    true);
+							    UAC_VERSION_3);
 	if (input_term) {
 		clock = input_term->bCSourceID;
 		goto found_clock;
 	}
 
 	output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-							     as->bTerminalLink);
+							      as->bTerminalLink,
+							      UAC_VERSION_3);
 	if (output_term) {
 		clock = output_term->bCSourceID;
 		goto found_clock;
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
new file mode 100644
index 000000000000..3c8f73a0eb12
--- /dev/null
+++ b/sound/usb/validate.c
@@ -0,0 +1,332 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+//
+// Validation of USB-audio class descriptors
+//
+
+#include <linux/init.h>
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/usb/audio-v3.h>
+#include <linux/usb/midi.h>
+#include "usbaudio.h"
+#include "helper.h"
+
+struct usb_desc_validator {
+	unsigned char protocol;
+	unsigned char type;
+	bool (*func)(const void *p, const struct usb_desc_validator *v);
+	size_t size;
+};
+
+#define UAC_VERSION_ALL		(unsigned char)(-1)
+
+/* UAC1 only */
+static bool validate_uac1_header(const void *p,
+				 const struct usb_desc_validator *v)
+{
+	const struct uac1_ac_header_descriptor *d = p;
+
+	return d->bLength >= sizeof(*d) &&
+		d->bLength >= sizeof(*d) + d->bInCollection;
+}
+
+/* for mixer unit; covering all UACs */
+static bool validate_mixer_unit(const void *p,
+				const struct usb_desc_validator *v)
+{
+	const struct uac_mixer_unit_descriptor *d = p;
+	size_t len;
+
+	if (d->bLength < sizeof(*d) || !d->bNrInPins)
+		return false;
+	len = sizeof(*d) + d->bNrInPins;
+	/* We can't determine the bitmap size only from this unit descriptor,
+	 * so just check with the remaining length.
+	 * The actual bitmap is checked at mixer unit parser.
+	 */
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		len += 2 + 1; /* wChannelConfig, iChannelNames */
+		/* bmControls[n*m] */
+		len += 1; /* iMixer */
+		break;
+	case UAC_VERSION_2:
+		len += 4 + 1; /* bmChannelConfig, iChannelNames */
+		/* bmMixerControls[n*m] */
+		len += 1 + 1; /* bmControls, iMixer */
+		break;
+	case UAC_VERSION_3:
+		len += 2; /* wClusterDescrID */
+		/* bmMixerControls[n*m] */
+		break;
+	}
+	return d->bLength >= len;
+}
+
+/* both for processing and extension units; covering all UACs */
+static bool validate_processing_unit(const void *p,
+				     const struct usb_desc_validator *v)
+{
+	const struct uac_processing_unit_descriptor *d = p;
+	const unsigned char *hdr = p;
+	size_t len, m;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	len = d->bLength < sizeof(*d) + d->bNrInPins;
+	if (d->bLength < len)
+		return false;
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		/* bNrChannels, wChannelConfig, iChannelNames, bControlSize */
+		len += 1 + 2 + 1 + 1;
+		if (d->bLength < len) /* bControlSize */
+			return false;
+		m = hdr[len];
+		len += 1 + m + 1; /* bControlSize, bmControls, iProcessing */
+		break;
+	case UAC_VERSION_2:
+		/* bNrChannels, bmChannelConfig, iChannelNames */
+		len += 1 + 4 + 1;
+		if (v->type == UAC2_PROCESSING_UNIT_V2)
+			len += 2; /* bmControls -- 2 bytes for PU */
+		else
+			len += 1; /* bmControls -- 1 byte for EU */
+		len += 1; /* iProcessing */
+		break;
+	case UAC_VERSION_3:
+		/* wProcessingDescrStr, bmControls */
+		len += 2 + 4;
+		break;
+	}
+	if (d->bLength < len)
+		return false;
+
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		if (v->type == UAC1_EXTENSION_UNIT)
+			return true; /* OK */
+		switch (d->wProcessType) {
+		case UAC_PROCESS_UP_DOWNMIX:
+		case UAC_PROCESS_DOLBY_PROLOGIC:
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 2; /* bNrModes, waModes(n) */
+			break;
+		default:
+			break;
+		}
+		break;
+	case UAC_VERSION_2:
+		if (v->type == UAC2_EXTENSION_UNIT_V2)
+			return true; /* OK */
+		switch (d->wProcessType) {
+		case UAC2_PROCESS_UP_DOWNMIX:
+		case UAC2_PROCESS_DOLBY_PROLOCIC: /* SiC! */
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 4; /* bNrModes, daModes(n) */
+			break;
+		default:
+			break;
+		}
+		break;
+	case UAC_VERSION_3:
+		if (v->type == UAC3_EXTENSION_UNIT) {
+			len += 2; /* wClusterDescrID */
+			break;
+		}
+		switch (d->wProcessType) {
+		case UAC3_PROCESS_UP_DOWNMIX:
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 2; /* bNrModes, waClusterDescrID(n) */
+			break;
+		case UAC3_PROCESS_MULTI_FUNCTION:
+			len += 2 + 4; /* wClusterDescrID, bmAlgorighms */
+			break;
+		default:
+			break;
+		}
+		break;
+	}
+	if (d->bLength < len)
+		return false;
+
+	return true;
+}
+
+/* both for selector and clock selector units; covering all UACs */
+static bool validate_selector_unit(const void *p,
+				   const struct usb_desc_validator *v)
+{
+	const struct uac_selector_unit_descriptor *d = p;
+	size_t len;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	len = sizeof(*d) + d->bNrInPins;
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		len += 1; /* iSelector */
+		break;
+	case UAC_VERSION_2:
+		len += 1 + 1; /* bmControls, iSelector */
+		break;
+	case UAC_VERSION_3:
+		len += 4 + 2; /* bmControls, wSelectorDescrStr */
+		break;
+	}
+	return d->bLength >= len;
+}
+
+static bool validate_uac1_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d) || !d->bControlSize)
+		return false;
+	/* at least bmaControls(0) for master channel + iFeature */
+	return d->bLength >= sizeof(*d) + d->bControlSize + 1;
+}
+
+static bool validate_uac2_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac2_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	/* at least bmaControls(0) for master channel + iFeature */
+	return d->bLength >= sizeof(*d) + 4 + 1;
+}
+
+static bool validate_uac3_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac3_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	/* at least bmaControls(0) for master channel + wFeatureDescrStr */
+	return d->bLength >= sizeof(*d) + 4 + 2;
+}
+
+static bool validate_midi_out_jack(const void *p,
+				   const struct usb_desc_validator *v)
+{
+	const struct usb_midi_out_jack_descriptor *d = p;
+
+	return d->bLength >= sizeof(*d) &&
+		d->bLength >= sizeof(*d) + d->bNrInputPins * 2;
+}
+
+#define FIXED(p, t, s) { .protocol = (p), .type = (t), .size = sizeof(s) }
+#define FUNC(p, t, f) { .protocol = (p), .type = (t), .func = (f) }
+
+static struct usb_desc_validator audio_validators[] = {
+	/* UAC1 */
+	FUNC(UAC_VERSION_1, UAC_HEADER, validate_uac1_header),
+	FIXED(UAC_VERSION_1, UAC_INPUT_TERMINAL,
+	      struct uac_input_terminal_descriptor),
+	FIXED(UAC_VERSION_1, UAC_OUTPUT_TERMINAL,
+	      struct uac1_output_terminal_descriptor),
+	FUNC(UAC_VERSION_1, UAC_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_1, UAC_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_1, UAC_FEATURE_UNIT, validate_uac1_feature_unit),
+	FUNC(UAC_VERSION_1, UAC1_PROCESSING_UNIT, validate_processing_unit),
+	FUNC(UAC_VERSION_1, UAC1_EXTENSION_UNIT, validate_processing_unit),
+
+	/* UAC2 */
+	FIXED(UAC_VERSION_2, UAC_HEADER, struct uac2_ac_header_descriptor),
+	FIXED(UAC_VERSION_2, UAC_INPUT_TERMINAL,
+	      struct uac2_input_terminal_descriptor),
+	FIXED(UAC_VERSION_2, UAC_OUTPUT_TERMINAL,
+	      struct uac2_output_terminal_descriptor),
+	FUNC(UAC_VERSION_2, UAC_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_2, UAC_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_2, UAC_FEATURE_UNIT, validate_uac2_feature_unit),
+	/* UAC_VERSION_2, UAC2_EFFECT_UNIT: not implemented yet */
+	FUNC(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2, validate_processing_unit),
+	FUNC(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2, validate_processing_unit),
+	FIXED(UAC_VERSION_2, UAC2_CLOCK_SOURCE,
+	      struct uac_clock_source_descriptor),
+	FUNC(UAC_VERSION_2, UAC2_CLOCK_SELECTOR, validate_selector_unit),
+	FIXED(UAC_VERSION_2, UAC2_CLOCK_MULTIPLIER,
+	      struct uac_clock_multiplier_descriptor),
+	/* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */
+
+	/* UAC3 */
+	FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor),
+	FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL,
+	      struct uac3_input_terminal_descriptor),
+	FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL,
+	      struct uac3_output_terminal_descriptor),
+	/* UAC_VERSION_3, UAC3_EXTENDED_TERMINAL: not implemented yet */
+	FUNC(UAC_VERSION_3, UAC3_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_3, UAC3_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_3, UAC_FEATURE_UNIT, validate_uac3_feature_unit),
+	/*  UAC_VERSION_3, UAC3_EFFECT_UNIT: not implemented yet */
+	FUNC(UAC_VERSION_3, UAC3_PROCESSING_UNIT, validate_processing_unit),
+	FUNC(UAC_VERSION_3, UAC3_EXTENSION_UNIT, validate_processing_unit),
+	FIXED(UAC_VERSION_3, UAC3_CLOCK_SOURCE,
+	      struct uac3_clock_source_descriptor),
+	FUNC(UAC_VERSION_3, UAC3_CLOCK_SELECTOR, validate_selector_unit),
+	FIXED(UAC_VERSION_3, UAC3_CLOCK_MULTIPLIER,
+	      struct uac3_clock_multiplier_descriptor),
+	/* UAC_VERSION_3, UAC3_SAMPLE_RATE_CONVERTER: not implemented yet */
+	/* UAC_VERSION_3, UAC3_CONNECTORS: not implemented yet */
+	{ } /* terminator */
+};
+
+static struct usb_desc_validator midi_validators[] = {
+	FIXED(UAC_VERSION_ALL, USB_MS_HEADER,
+	      struct usb_ms_header_descriptor),
+	FIXED(UAC_VERSION_ALL, USB_MS_MIDI_IN_JACK,
+	      struct usb_midi_in_jack_descriptor),
+	FUNC(UAC_VERSION_ALL, USB_MS_MIDI_OUT_JACK,
+	     validate_midi_out_jack),
+	{ } /* terminator */
+};
+
+
+/* Validate the given unit descriptor, return true if it's OK */
+static bool validate_desc(unsigned char *hdr, int protocol,
+			  const struct usb_desc_validator *v)
+{
+	if (hdr[1] != USB_DT_CS_INTERFACE)
+		return true; /* don't care */
+
+	for (; v->type; v++) {
+		if (v->type == hdr[2] &&
+		    (v->protocol == UAC_VERSION_ALL ||
+		     v->protocol == protocol)) {
+			if (v->func)
+				return v->func(hdr, v);
+			/* check for the fixed size */
+			return hdr[0] >= v->size;
+		}
+	}
+
+	return true; /* not matching, skip validation */
+}
+
+bool snd_usb_validate_audio_desc(void *p, int protocol)
+{
+	return validate_desc(p, protocol, audio_validators);
+}
+
+bool snd_usb_validate_midi_desc(void *p)
+{
+	return validate_desc(p, UAC_VERSION_1, midi_validators);
+}
+

From 68e9fde245591d18200f8a9054cac22339437adb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 15 Aug 2019 16:30:39 +0200
Subject: [PATCH 20/22] ALSA: usb-audio: Simplify parse_audio_unit()

Minor code refactoring by combining the UAC version and the type in
the switch-case flow, so that we reduce the indentation and
redundancy.  One good bonus is that the duplicated definition of the
same type value (e.g. UAC2_EFFECT_UNIT) can be handled more cleanly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 95 +++++++++++++++++++----------------------------
 1 file changed, 39 insertions(+), 56 deletions(-)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index a1093fb9bf09..34def5911af2 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2782,62 +2782,45 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 		return 0; /* skip invalid unit */
 	}
 
-	if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
-		switch (p1[2]) {
-		case UAC_INPUT_TERMINAL:
-			return parse_audio_input_terminal(state, unitid, p1);
-		case UAC_MIXER_UNIT:
-			return parse_audio_mixer_unit(state, unitid, p1);
-		case UAC2_CLOCK_SOURCE:
-			return parse_clock_source_unit(state, unitid, p1);
-		case UAC_SELECTOR_UNIT:
-		case UAC2_CLOCK_SELECTOR:
-			return parse_audio_selector_unit(state, unitid, p1);
-		case UAC_FEATURE_UNIT:
-			return parse_audio_feature_unit(state, unitid, p1);
-		case UAC1_PROCESSING_UNIT:
-		/*   UAC2_EFFECT_UNIT has the same value */
-			if (protocol == UAC_VERSION_1)
-				return parse_audio_processing_unit(state, unitid, p1);
-			else
-				return 0; /* FIXME - effect units not implemented yet */
-		case UAC1_EXTENSION_UNIT:
-		/*   UAC2_PROCESSING_UNIT_V2 has the same value */
-			if (protocol == UAC_VERSION_1)
-				return parse_audio_extension_unit(state, unitid, p1);
-			else /* UAC_VERSION_2 */
-				return parse_audio_processing_unit(state, unitid, p1);
-		case UAC2_EXTENSION_UNIT_V2:
-			return parse_audio_extension_unit(state, unitid, p1);
-		default:
-			usb_audio_err(state->chip,
-				"unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
-			return -EINVAL;
-		}
-	} else { /* UAC_VERSION_3 */
-		switch (p1[2]) {
-		case UAC_INPUT_TERMINAL:
-			return parse_audio_input_terminal(state, unitid, p1);
-		case UAC3_MIXER_UNIT:
-			return parse_audio_mixer_unit(state, unitid, p1);
-		case UAC3_CLOCK_SOURCE:
-			return parse_clock_source_unit(state, unitid, p1);
-		case UAC3_SELECTOR_UNIT:
-		case UAC3_CLOCK_SELECTOR:
-			return parse_audio_selector_unit(state, unitid, p1);
-		case UAC3_FEATURE_UNIT:
-			return parse_audio_feature_unit(state, unitid, p1);
-		case UAC3_EFFECT_UNIT:
-			return 0; /* FIXME - effect units not implemented yet */
-		case UAC3_PROCESSING_UNIT:
-			return parse_audio_processing_unit(state, unitid, p1);
-		case UAC3_EXTENSION_UNIT:
-			return parse_audio_extension_unit(state, unitid, p1);
-		default:
-			usb_audio_err(state->chip,
-				"unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
-			return -EINVAL;
-		}
+#define PTYPE(a, b)	((a) << 8 | (b))
+	switch (PTYPE(protocol, p1[2])) {
+	case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL):
+	case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL):
+	case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL):
+		return parse_audio_input_terminal(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT):
+		return parse_audio_mixer_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE):
+	case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE):
+		return parse_clock_source_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR):
+	case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR):
+		return parse_audio_selector_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT):
+		return parse_audio_feature_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2):
+	case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT):
+		return parse_audio_processing_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2):
+	case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT):
+		return parse_audio_extension_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT):
+		return 0; /* FIXME - effect units not implemented yet */
+	default:
+		usb_audio_err(state->chip,
+			      "unit %u: unexpected type 0x%02x\n",
+			      unitid, p1[2]);
+		return -EINVAL;
 	}
 }
 

From 52c3e317a857091fd746e15179a637f32be4d337 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 08:23:10 +0200
Subject: [PATCH 21/22] ALSA: usb-audio: Unify the release of
 usb_mixer_elem_info objects

Instead of the direct kfree() calls, introduce a new local helper to
release the usb_mixer_elem_info object.  This will be extended to do
more than a single kfree() in the later patches.

Also, use the standard goto instead of multiple calls in
parse_audio_selector_unit() error paths.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 48 +++++++++++++++++++++++++++--------------------
 1 file changed, 28 insertions(+), 20 deletions(-)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 34def5911af2..277660fd6e0a 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1026,10 +1026,15 @@ static struct usb_feature_control_info audio_feature_info[] = {
 	{ UAC2_FU_PHASE_INVERTER,	 "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 },
 };
 
+static void usb_mixer_elem_info_free(struct usb_mixer_elem_info *cval)
+{
+	kfree(cval);
+}
+
 /* private_free callback */
 void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl)
 {
-	kfree(kctl->private_data);
+	usb_mixer_elem_info_free(kctl->private_data);
 	kctl->private_data = NULL;
 }
 
@@ -1552,7 +1557,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 
 	ctl_info = get_feature_control_info(control);
 	if (!ctl_info) {
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	if (mixer->protocol == UAC_VERSION_1)
@@ -1585,7 +1590,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 
 	if (!kctl) {
 		usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
@@ -1755,7 +1760,7 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
 	kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval);
 	if (!kctl) {
 		usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	get_connector_control_name(mixer, term, is_input, kctl->id.name,
@@ -1808,7 +1813,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
 	kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval);
 
 	if (!kctl) {
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return -ENOMEM;
 	}
 
@@ -2070,7 +2075,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
 	kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
 	if (!kctl) {
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
@@ -2468,7 +2473,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
 		if (!kctl) {
-			kfree(cval);
+			usb_mixer_elem_info_free(cval);
 			return -ENOMEM;
 		}
 		kctl->private_free = snd_usb_mixer_elem_free;
@@ -2606,7 +2611,7 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl)
 	if (kctl->private_data) {
 		struct usb_mixer_elem_info *cval = kctl->private_data;
 		num_ins = cval->max;
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		kctl->private_data = NULL;
 	}
 	if (kctl->private_value) {
@@ -2678,10 +2683,10 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		break;
 	}
 
-	namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL);
+	namelist = kcalloc(desc->bNrInPins, sizeof(char *), GFP_KERNEL);
 	if (!namelist) {
-		kfree(cval);
-		return -ENOMEM;
+		err = -ENOMEM;
+		goto error_cval;
 	}
 #define MAX_ITEM_NAME_LEN	64
 	for (i = 0; i < desc->bNrInPins; i++) {
@@ -2689,11 +2694,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		len = 0;
 		namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL);
 		if (!namelist[i]) {
-			while (i--)
-				kfree(namelist[i]);
-			kfree(namelist);
-			kfree(cval);
-			return -ENOMEM;
+			err = -ENOMEM;
+			goto error_name;
 		}
 		len = check_mapped_selector_name(state, unitid, i, namelist[i],
 						 MAX_ITEM_NAME_LEN);
@@ -2707,10 +2709,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval);
 	if (! kctl) {
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
-		for (i = 0; i < desc->bNrInPins; i++)
-			kfree(namelist[i]);
-		kfree(namelist);
-		kfree(cval);
+		err = -ENOMEM;
+		goto error_name;
 		return -ENOMEM;
 	}
 	kctl->private_value = (unsigned long)namelist;
@@ -2757,6 +2757,14 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	usb_audio_dbg(state->chip, "[%d] SU [%s] items = %d\n",
 		    cval->head.id, kctl->id.name, desc->bNrInPins);
 	return snd_usb_mixer_add_control(&cval->head, kctl);
+
+ error_name:
+	for (i = 0; i < desc->bNrInPins; i++)
+		kfree(namelist[i]);
+	kfree(namelist);
+ error_cval:
+	usb_mixer_elem_info_free(cval);
+	return err;
 }
 
 /*

From b8e4f1fdfa422398c2d6c47bfb7d1feb3046d70a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 09:25:27 +0200
Subject: [PATCH 22/22] ALSA: usb-audio: Remove superfluous bLength checks

Now that we got the more comprehensive validation code for USB-audio
descriptors, the check of overflow in each descriptor unit parser
became superfluous.  Drop some of the obvious cases.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/clock.c | 14 ++++----
 sound/usb/mixer.c | 84 -----------------------------------------------
 2 files changed, 6 insertions(+), 92 deletions(-)

diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 72e9bdf76115..6b8c14f9b5d4 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -38,39 +38,37 @@ static void *find_uac_clock_desc(struct usb_host_interface *iface, int id,
 static bool validate_clock_source_v2(void *p, int id)
 {
 	struct uac_clock_source_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_source_v3(void *p, int id)
 {
 	struct uac3_clock_source_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_selector_v2(void *p, int id)
 {
 	struct uac_clock_selector_descriptor *cs = p;
-	return cs->bLength >= sizeof(*cs) && cs->bClockID == id &&
-		cs->bLength == 7 + cs->bNrInPins;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_selector_v3(void *p, int id)
 {
 	struct uac3_clock_selector_descriptor *cs = p;
-	return cs->bLength >= sizeof(*cs) && cs->bClockID == id &&
-		cs->bLength == 11 + cs->bNrInPins;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_multiplier_v2(void *p, int id)
 {
 	struct uac_clock_multiplier_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_multiplier_v3(void *p, int id)
 {
 	struct uac3_clock_multiplier_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 #define DEFINE_FIND_HELPER(name, obj, validator, type)		\
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 277660fd6e0a..c9777d8a76f5 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -740,13 +740,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 {
 	int mu_channels;
 
-	if (desc->bLength < sizeof(*desc))
-		return -EINVAL;
-	if (!desc->bNrInPins)
-		return -EINVAL;
-	if (desc->bLength < sizeof(*desc) + desc->bNrInPins)
-		return -EINVAL;
-
 	switch (state->mixer->protocol) {
 	case UAC_VERSION_1:
 	case UAC_VERSION_2:
@@ -1781,13 +1774,6 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
 	if (state->mixer->protocol != UAC_VERSION_2)
 		return -EINVAL;
 
-	if (hdr->bLength != sizeof(*hdr)) {
-		usb_audio_dbg(state->chip,
-			      "Bogus clock source descriptor length of %d, ignoring.\n",
-			      hdr->bLength);
-		return 0;
-	}
-
 	/*
 	 * The only property of this unit we are interested in is the
 	 * clock source validity. If that isn't readable, just bail out.
@@ -1846,62 +1832,20 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid,
 	__u8 *bmaControls;
 
 	if (state->mixer->protocol == UAC_VERSION_1) {
-		if (hdr->bLength < 7) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = hdr->bControlSize;
-		if (!csize) {
-			usb_audio_dbg(state->chip,
-				      "unit %u: invalid bControlSize == 0\n",
-				      unitid);
-			return -EINVAL;
-		}
 		channels = (hdr->bLength - 7) / csize - 1;
 		bmaControls = hdr->bmaControls;
-		if (hdr->bLength < 7 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	} else if (state->mixer->protocol == UAC_VERSION_2) {
 		struct uac2_feature_unit_descriptor *ftr = _ftr;
-		if (hdr->bLength < 6) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = 4;
 		channels = (hdr->bLength - 6) / 4 - 1;
 		bmaControls = ftr->bmaControls;
-		if (hdr->bLength < 6 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	} else { /* UAC_VERSION_3 */
 		struct uac3_feature_unit_descriptor *ftr = _ftr;
 
-		if (hdr->bLength < 7) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = 4;
 		channels = (ftr->bLength - 7) / 4 - 1;
 		bmaControls = ftr->bmaControls;
-		if (hdr->bLength < 7 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	}
 
 	/* parse the source unit */
@@ -2101,15 +2045,11 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid,
 
 	if (state->mixer->protocol == UAC_VERSION_2) {
 		struct uac2_input_terminal_descriptor *d_v2 = raw_desc;
-		if (d_v2->bLength < sizeof(*d_v2))
-			return -EINVAL;
 		control = UAC2_TE_CONNECTOR;
 		term_id = d_v2->bTerminalID;
 		bmctls = le16_to_cpu(d_v2->bmControls);
 	} else if (state->mixer->protocol == UAC_VERSION_3) {
 		struct uac3_input_terminal_descriptor *d_v3 = raw_desc;
-		if (d_v3->bLength < sizeof(*d_v3))
-			return -EINVAL;
 		control = UAC3_TE_INSERTION;
 		term_id = d_v3->bTerminalID;
 		bmctls = le32_to_cpu(d_v3->bmControls);
@@ -2371,18 +2311,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 	const char *name = extension_unit ?
 		"Extension Unit" : "Processing Unit";
 
-	if (desc->bLength < 13) {
-		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
-		return -EINVAL;
-	}
-
 	num_ins = desc->bNrInPins;
-	if (desc->bLength < 13 + num_ins ||
-	    desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) {
-		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
-		return -EINVAL;
-	}
-
 	for (i = 0; i < num_ins; i++) {
 		err = parse_audio_unit(state, desc->baSourceID[i]);
 		if (err < 0)
@@ -2637,13 +2566,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	const struct usbmix_name_map *map;
 	char **namelist;
 
-	if (desc->bLength < 5 || !desc->bNrInPins ||
-	    desc->bLength < 5 + desc->bNrInPins) {
-		usb_audio_err(state->chip,
-			"invalid SELECTOR UNIT descriptor %d\n", unitid);
-		return -EINVAL;
-	}
-
 	for (i = 0; i < desc->bNrInPins; i++) {
 		err = parse_audio_unit(state, desc->baSourceID[i]);
 		if (err < 0)
@@ -3149,8 +3071,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;
@@ -3162,8 +3082,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		} else if (mixer->protocol == UAC_VERSION_2) {
 			struct uac2_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;
@@ -3189,8 +3107,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		} else {  /* UAC_VERSION_3 */
 			struct uac3_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;