From b51abed8355e5556886623b2772fa6b7598d2282 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 29 Nov 2018 08:02:49 +0100
Subject: [PATCH 01/11] ALSA: pcm: Call snd_pcm_unlink() conditionally at
closing
Currently the PCM core calls snd_pcm_unlink() always unconditionally
at closing a stream. However, since snd_pcm_unlink() invokes the
global rwsem down, the lock can be easily contended. More badly, when
a thread runs in a high priority RT-FIFO, it may stall at spinning.
Basically the call of snd_pcm_unlink() is required only for the linked
streams that are already rare occasion. For normal use cases, this
code path is fairly superfluous.
As an optimization (and also as a workaround for the RT problem
above in normal situations without linked streams), this patch adds a
check before calling snd_pcm_unlink() and calls it only when needed.
Reported-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 66c90f486af91..6afcc393113a9 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2369,7 +2369,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
static void pcm_release_private(struct snd_pcm_substream *substream)
{
- snd_pcm_unlink(substream);
+ if (snd_pcm_stream_linked(substream))
+ snd_pcm_unlink(substream);
}
void snd_pcm_release_substream(struct snd_pcm_substream *substream)
From b888a5f713e4d17faaaff24316585a4eb07f35b7 Mon Sep 17 00:00:00 2001
From: Chanho Min <chanho.min@lge.com>
Date: Mon, 26 Nov 2018 14:36:37 +0900
Subject: [PATCH 02/11] ALSA: pcm: Fix starvation on down_write_nonblock()
Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM
stream") fixes deadlock for non-atomic PCM stream. But, This patch
causes antother stuck.
If writer is RT thread and reader is a normal thread, the reader
thread will be difficult to get scheduled. It may not give chance to
release readlocks and writer gets stuck for a long time if they are
pinned to single cpu.
The deadlock described in the previous commit is because the linux
rwsem queues like a FIFO. So, we might need non-FIFO writelock, not
non-block one.
My suggestion is that the writer gives reader a chance to be scheduled
by using the minimum msleep() instaed of spinning without blocking by
writer. Also, The *_nonblock may be changed to *_nonfifo appropriately
to this concept.
In terms of performance, when trylock is failed, this minimum periodic
msleep will have the same performance as the tick-based
schedule()/wake_up_q().
[ Although this has a fairly high performance penalty, the relevant
code path became already rare due to the previous commit ("ALSA:
pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now
this unconditional msleep appears only when using linked streams,
and this must be a rare case. So we accept this as a quick
workaround until finding a more suitable one -- tiwai ]
Fixes: 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream")
Suggested-by: Wonmin Jung <wonmin.jung@lge.com>
Signed-off-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 11 ++++++-----
1 file changed, 6 insertions(+), 5 deletions(-)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 6afcc393113a9..818dff1de545f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -36,6 +36,7 @@
#include <sound/timer.h>
#include <sound/minors.h>
#include <linux/uio.h>
+#include <linux/delay.h>
#include "pcm_local.h"
@@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
* and this may lead to a deadlock when the code path takes read sem
* twice (e.g. one in snd_pcm_action_nonatomic() and another in
* snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to
- * spin until it gets the lock.
+ * sleep until all the readers are completed without blocking by writer.
*/
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
{
while (!down_write_trylock(lock))
- cond_resched();
+ msleep(1);
}
#define PCM_LOCK_DEFAULT 0
@@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
res = -ENOMEM;
goto _nolock;
}
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
substream->runtime->status->state != substream1->runtime->status->state ||
@@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
struct snd_pcm_substream *s;
int res = 0;
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (!snd_pcm_stream_linked(substream)) {
res = -EALREADY;
From 44ff57e685f96d0cb9540004cc9d1d880e7a4315 Mon Sep 17 00:00:00 2001
From: Tony Das <tdas444@gmail.com>
Date: Wed, 28 Nov 2018 20:16:37 +0000
Subject: [PATCH 03/11] ALSA: usb-audio: Add SMSL D1 to quirks for native DSD
support
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/quirks.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 8a945ece98690..6623cafc94f2c 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
+ case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */
case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */
From 3deef52ce10514ccdebba8e8ab85f9cebd0eb3f7 Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Thu, 29 Nov 2018 08:57:37 +0000
Subject: [PATCH 04/11] ALSA: hda: Add support for AMD Stoney Ridge
It's similar to other AMD audio devices, it also supports D3, which can
save some power drain.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0bbdf1a01e763..76f03abd15ab7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2498,6 +2498,10 @@ 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 Stoney */
+ { PCI_DEVICE(0x1022, 0x157a),
+ .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+ AZX_DCAPS_PM_RUNTIME },
/* AMD Raven */
{ PCI_DEVICE(0x1022, 0x15e3),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
From 5363857b916c1f48027e9b96ee8be8376bf20811 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 29 Nov 2018 12:05:19 +0100
Subject: [PATCH 05/11] ALSA: pcm: Fix interval evaluation with openmin/max
As addressed in alsa-lib (commit b420056604f0), we need to fix the
case where the evaluation of PCM interval "(x x+1]" leading to
-EINVAL. After applying rules, such an interval may be translated as
"(x x+1)".
Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/pcm_params.h | 4 +++-
1 file changed, 3 insertions(+), 1 deletion(-)
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 2dd37cada7c08..888a833d3b003 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i)
static inline int snd_interval_single(const struct snd_interval *i)
{
return (i->min == i->max ||
- (i->min + 1 == i->max && i->openmax));
+ (i->min + 1 == i->max && (i->openmin || i->openmax)));
}
static inline int snd_interval_value(const struct snd_interval *i)
{
+ if (i->openmin && !i->openmax)
+ return i->max;
return i->min;
}
From 54947cd64c1b8290f64bb2958e343c07270e3a58 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 3 Dec 2018 10:44:15 +0100
Subject: [PATCH 06/11] ALSA: hda/realtek - Fix speaker output regression on
Thinkpad T570
We've got a regression report for some Thinkpad models (at least
T570s) which shows the too low speaker output volume. The bisection
leaded to the commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad
Dock device for ALC298 platform"), and it's basically adding the two
pin configurations for the dock, and looks harmless.
The real culprit seems, though, that the DAC assignment for the
speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14
to be coupled with DAC NID 0x03. When more pins are configured by the
commit above, the auto-parser changes the DAC assignment, and this
resulted in the regression.
As a workaround, just provide the fixed pin / DAC mapping table for
this Thinkpad fixup function. It's no generic solution, but the
problem itself is pretty much device-specific, so must be good
enough.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304
Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 9 +++++++++
1 file changed, 9 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 06f93032d0ccf..802f1f1b3a193 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4988,9 +4988,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec,
{ 0x19, 0x21a11010 }, /* dock mic */
{ }
};
+ /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+ * the speaker output becomes too low by some reason on Thinkpads with
+ * ALC298 codec
+ */
+ static hda_nid_t preferred_pairs[] = {
+ 0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+ 0
+ };
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.preferred_dacs = preferred_pairs;
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
snd_hda_apply_pincfgs(codec, pincfgs);
} else if (action == HDA_FIXUP_ACT_INIT) {
From 5f8cf712582617d523120df67d392059eaf2fc4b Mon Sep 17 00:00:00 2001
From: Hui Peng <benquike@gmail.com>
Date: Mon, 3 Dec 2018 16:09:34 +0100
Subject: [PATCH 07/11] ALSA: usb-audio: Fix UAF decrement if card has no live
interfaces in card.c
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49abe5 ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/card.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2bfe4e80a6b92..a105947eaf55c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf,
__error:
if (chip) {
+ /* chip->active is inside the chip->card object,
+ * decrement before memory is possibly returned.
+ */
+ atomic_dec(&chip->active);
if (!chip->num_interfaces)
snd_card_free(chip->card);
- atomic_dec(&chip->active);
}
mutex_unlock(®ister_mutex);
return err;
From 33aaebd48ae2d2c78fef5063a0381e17db19b060 Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Wed, 5 Dec 2018 14:48:53 +0800
Subject: [PATCH 08/11] ALSA: hda/realtek: ALC286 mic and headset-mode fixups
for Acer Aspire U27-880
Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic
and internal mic not working either. It needs the similar quirk like
Sony laptops to fix headphone jack sensing and enables use of the
internal microphone.
Unfortunately jack sensing for the headset mic is still not working.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 14 ++++++++++++++
1 file changed, 14 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 802f1f1b3a193..0693dbb2c1674 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5519,6 +5519,7 @@ enum {
ALC221_FIXUP_HP_HEADSET_MIC,
ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
ALC295_FIXUP_HP_AUTO_MUTE,
+ ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -6396,6 +6397,15 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_auto_mute_via_amp,
},
+ [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7074,6 +7084,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x14, 0x90170110},
{0x19, 0x04a11040},
{0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
+ {0x12, 0x90a60130},
+ {0x17, 0x90170110},
+ {0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0x90a60120},
{0x14, 0x90170110},
From 705b65f107470499442240ff7afee5021a7002a6 Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Wed, 5 Dec 2018 14:48:54 +0800
Subject: [PATCH 09/11] ALSA: hda/realtek - Add support for Acer Aspire C24-860
headset mic
The Acer AIO Aspire C24-860 with ALC286 can't detect the headset
microphone. Just like another Acer AIO U27-880, it needs a different
pin value for 0x18 and the headset fixup to make headset mic work.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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 0693dbb2c1674..91e1487b25e29 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6420,6 +6420,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+ SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
From 9f8aefed9623a91dec54eab8908f3810b7f8d73a Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Wed, 5 Dec 2018 14:48:55 +0800
Subject: [PATCH 10/11] ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton
Z4660G
Acer AIO Veriton Z4660G with ALC286 codec has issue with the input
from external microphones connecting via 'Front Mic' jack. The fixup
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of
the headset and fix the audio input issue of external microphone.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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 91e1487b25e29..2a5ecdf261484 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6422,6 +6422,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
From b72f936f6b325f4fde06b02e4b6ab682f6f2e73f Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Wed, 5 Dec 2018 14:48:56 +0800
Subject: [PATCH 11/11] ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton
Z4860G/Z6860G
Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues
with the input from external microphone. The issue can be fixed by
the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2a5ecdf261484..8d75597028eeb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6422,6 +6422,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),