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[ Upstream commit 712a8038cc24dba668afe82f0413714ca87184e0 ]
When the ssm4567 is powered up the driver calles regcache_sync() to restore
the register map content. regcache_sync() assumes that the device is in its
power-on reset state. Make sure that this is the case by explicitly
resetting the ssm4567 register map before calling regcache_sync() otherwise
we might end up with a incorrect register map which leads to undefined
behaviour.
One such undefined behaviour was observed when returning from system
suspend while a playback stream is active, in that case the ssm4567 was
kept muted after resume.
Fixes: 1ee44ce03011 ("ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier")
Reported-by: Harsha Priya <harshapriya.n@intel.com>
Tested-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit ba4bc32eaa39ba7687f0958ae90eec94da613b46 ]
An older patch to convert the API in the s3c i2s driver
ended up passing a const pointer into a function that takes
a non-const pointer, so we now get a warning:
sound/soc/samsung/s3c2412-i2s.c: In function 's3c2412_iis_dev_probe':
sound/soc/samsung/s3c2412-i2s.c:172:9: error: passing argument 3 of 's3c_i2sv2_register_component' discards 'const' qualifier from pointer target type [-Werror=discarded-qualifiers]
However, the s3c_i2sv2_register_component() function again
passes the pointer into another function taking a const, so
we just need to change its prototype.
Fixes: eca3b01d0885 ("ASoC: switch over to use snd_soc_register_component() on s3c i2s")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Krzysztof Kozlowski <k.kozlowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 4a07083ed613644c96c34a7dd2853dc5d7c70902 ]
ALSA system timer backend stops the timer via del_timer() without sync
and leaves del_timer_sync() at the close instead. This is because of
the restriction by the design of ALSA timer: namely, the stop callback
may be called from the timer handler, and calling the sync shall lead
to a hangup. However, this also triggers a kernel BUG() when the
timer is rearmed immediately after stopping without sync:
kernel BUG at kernel/time/timer.c:966!
Call Trace:
<IRQ>
[<ffffffff8239c94e>] snd_timer_s_start+0x13e/0x1a0
[<ffffffff8239e1f4>] snd_timer_interrupt+0x504/0xec0
[<ffffffff8122fca0>] ? debug_check_no_locks_freed+0x290/0x290
[<ffffffff8239ec64>] snd_timer_s_function+0xb4/0x120
[<ffffffff81296b72>] call_timer_fn+0x162/0x520
[<ffffffff81296add>] ? call_timer_fn+0xcd/0x520
[<ffffffff8239ebb0>] ? snd_timer_interrupt+0xec0/0xec0
....
It's the place where add_timer() checks the pending timer. It's clear
that this may happen after the immediate restart without sync in our
cases.
So, the workaround here is just to use mod_timer() instead of
add_timer(). This looks like a band-aid fix, but it's a right move,
as snd_timer_interrupt() takes care of the continuous rearm of timer.
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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snd_usb_add_audio_stream() call
[ Upstream commit 836b34a935abc91e13e63053d0a83b24dfb5ea78 ]
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e4a ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 902eb7fd1e4af3ac69b9b30f8373f118c92b9729 ]
Just a minor code cleanup: unify the error paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit b9a1a743818ea3265abf98f9431623afa8c50c86 ]
ARM64 allmodconfig produces a bunch of warnings when building the
samsung ASoC code:
sound/soc/samsung/dmaengine.c: In function 'samsung_asoc_init_dma_data':
sound/soc/samsung/dmaengine.c:53:32: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
playback_data->filter_data = (void *)playback->channel;
sound/soc/samsung/dmaengine.c:60:31: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
capture_data->filter_data = (void *)capture->channel;
We could easily shut up the warning by adding an intermediate cast,
but there is a bigger underlying problem: The use of IORESOURCE_DMA
to pass data from platform code to device drivers is dubious to start
with, as what we really want is a pointer that can be passed into
a filter function.
Note that on s3c64xx, the pl08x DMA data is already a pointer, but
gets cast to resource_size_t so we can pass it as a resource, and it
then gets converted back to a pointer. In contrast, the data we pass
for s3c24xx is an index into a device specific table, and we artificially
convert that into a pointer for the filter function.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Krzysztof Kozlowski <k.kozlowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit d5dbbe6569481bf12dcbe3e12cff72c5f78d272c ]
syzkaller fuzzer spotted a potential use-after-free case in snd-dummy
driver when hrtimer is used as backend:
> ==================================================================
> BUG: KASAN: use-after-free in rb_erase+0x1b17/0x2010 at addr ffff88005e5b6f68
> Read of size 8 by task syz-executor/8984
> =============================================================================
> BUG kmalloc-192 (Not tainted): kasan: bad access detected
> -----------------------------------------------------------------------------
>
> Disabling lock debugging due to kernel taint
> INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446705582212484632
> ....
> [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464
> ....
> INFO: Freed in 0xfffd8e09 age=18446705496313138713 cpu=2164287125 pid=-1
> [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481
> ....
> Call Trace:
> [<ffffffff8179e59e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:333
> [< inline >] rb_set_parent include/linux/rbtree_augmented.h:111
> [< inline >] __rb_erase_augmented include/linux/rbtree_augmented.h:218
> [<ffffffff82ca5787>] rb_erase+0x1b17/0x2010 lib/rbtree.c:427
> [<ffffffff82cb02e8>] timerqueue_del+0x78/0x170 lib/timerqueue.c:86
> [<ffffffff814d0c80>] __remove_hrtimer+0x90/0x220 kernel/time/hrtimer.c:903
> [< inline >] remove_hrtimer kernel/time/hrtimer.c:945
> [<ffffffff814d23da>] hrtimer_try_to_cancel+0x22a/0x570 kernel/time/hrtimer.c:1046
> [<ffffffff814d2742>] hrtimer_cancel+0x22/0x40 kernel/time/hrtimer.c:1066
> [<ffffffff85420531>] dummy_hrtimer_stop+0x91/0xb0 sound/drivers/dummy.c:417
> [<ffffffff854228bf>] dummy_pcm_trigger+0x17f/0x1e0 sound/drivers/dummy.c:507
> [<ffffffff85392170>] snd_pcm_do_stop+0x160/0x1b0 sound/core/pcm_native.c:1106
> [<ffffffff85391b26>] snd_pcm_action_single+0x76/0x120 sound/core/pcm_native.c:956
> [<ffffffff85391e01>] snd_pcm_action+0x231/0x290 sound/core/pcm_native.c:974
> [< inline >] snd_pcm_stop sound/core/pcm_native.c:1139
> [<ffffffff8539754d>] snd_pcm_drop+0x12d/0x1d0 sound/core/pcm_native.c:1784
> [<ffffffff8539d3be>] snd_pcm_common_ioctl1+0xfae/0x2150 sound/core/pcm_native.c:2805
> [<ffffffff8539ee91>] snd_pcm_capture_ioctl1+0x2a1/0x5e0 sound/core/pcm_native.c:2976
> [<ffffffff8539f2ec>] snd_pcm_kernel_ioctl+0x11c/0x160 sound/core/pcm_native.c:3020
> [<ffffffff853d9a44>] snd_pcm_oss_sync+0x3a4/0xa30 sound/core/oss/pcm_oss.c:1693
> [<ffffffff853da27d>] snd_pcm_oss_release+0x1ad/0x280 sound/core/oss/pcm_oss.c:2483
> .....
A workaround is to call hrtimer_cancel() in dummy_hrtimer_sync() which
is called certainly before other blocking ops.
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Tested-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 21e9d017b88ea0baa367ef0b6516d794fa23e85e ]
One more Dell laptop with alc293 codec needs
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, but the pin 0x1e does not match
the corresponding one in the ALC292_STANDARD_PINS. To use this macro
for this machine, we need to remove pin 0x1e from it.
BugLink: https://bugs.launchpad.net/bugs/1476888
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit f90d83b301701026b2e4c437a3613f377f63290e ]
Add the pin configuration value of this machine into the pin_quirk
table to make DELL1_MIC_NO_PRESENCE apply to this machine.
Signed-off-by: AceLan Kao <acelan.kao@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit d3030d11961a8c103cf07aed59905276ddfc06c2 ]
The ak4642 driver is using a regmap cache sync to restore the
configuration of the chip on resume but (as Peter observed) does not
actually define a register cache which means that the resume is never
going to work and we trigger asserts in regmap. Fix this by enabling
caching.
Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Reported-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit f8ea6cebcfa6499949392da71fc427567c9e5a0e ]
The max_register setting for ak4642, ak4643 and ak4648 are wrong, fix it.
According to the datasheet:
the maximum valid register for ak4642 is 0x1f
the maximum valid register for ak4643 is 0x24
the maximum valid register for ak4648 is 0x27
The default settings for ak4642 and ak4643 are the same for 0x0 ~ 0x1f
registers, so it's fine to use the same reg_default table with differnt
num_reg_defaults setting.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 84add303ef950b8d85f54bc2248c2bc73467c329 ]
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 3231e2053eaeee70bdfb216a78a30f11e88e2243 ]
Subwoofer does not work out of the box on ASUS N751/N551 laptops. This
patch fixes it. Patch tested on N751 laptop. N551 part is not tested,
but according to [1] and [2] this laptop requires similar changes, so I
included them in the patch.
1. https://github.com/honsiorovskyi/asus-n551-hda-fix
2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781
Signed-off-by: Yura Pakhuchiy <pakhuchiy@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 9d4dc5840f93bcb002fa311693349deae7702bc5 ]
For reducing the noise from the headphone output on ASUS N750JV,
call the existing fixup, alc_fixup_auto_mute_via_amp(), additionally.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115181
Signed-off-by: Bobi Mihalca <bobbymihalca@touchtech.ro>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 70cf2cbd685e218c3ffd105d9fb6cf0f8d767481 ]
ASUS N750JV needs the same fixup as N550 for enabling its subwoofer.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115181
Signed-off-by: Bobi Mihalca <bobbymihalca@touchtech.ro>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 2da2dc9ead232f25601404335cca13c0f722d41b ]
For reducing the noise from the headset output on ASUS UX501VW,
call the existing fixup, alc_fixup_headset_mode_alc668(), additionally.
Thread: https://bbs.archlinux.org/viewtopic.php?id=209554
Signed-off-by: Kaho Ng <ngkaho1234@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 83a9efb5b8170b7cffef4f62656656e1d8ad2ccd ]
Apply the new fixup that is used for ASUS N750JV to another similar
model, N500JV, too, for reducing the headphone noise.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115181
Signed-off-by: Bobi Mihalca <bobbymihalca@touchtech.ro>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit db8948e653e12b218058bb6696f4a33fa7845f64 ]
ASUS N550JX (PCI SSID 1043:13df) requires the same fixup for a bass
speaker output pin as other N550 models.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110001
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 2d2c038a9999f423e820d89db2b5d7774b67ba49 ]
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 037e119738120c1cdc460c6ae33871c3000531f3 ]
Fixes audio output on a ThinkPad X260, when using Lenovo CES 2013
docking station series (basic, pro, ultra).
Signed-off-by: Conrad Kostecki <ck+linuxkernel@bl4ckb0x.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 653aa4645244042826f105aab1be3d01b3d493ca ]
this patch corrects the interface adc/dac control register definition
according to datasheet.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 67f3754b51f22b18c4820fb84062f658c30e8644 ]
The commit [9bef72bdb26e: ALSA: pcxhr: Use nonatomic PCM ops]
converted to non-atomic PCM ops, but shamelessly with an unbalanced
mutex locking, which leads to the hangup easily. Fix it.
Fixes: 9bef72bdb26e ('ALSA: pcxhr: Use nonatomic PCM ops')
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116441
Cc: <stable@vger.kernel.org> # 3.18+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit adcdd0d5a1cb779f6d455ae70882c19c527627a8 ]
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.
[Some background: the patch essentially skips the controls of a couple
of FU volumes. Although the firmware exposes the dB and the value
information via the usb descriptor, changing the values (we set the
min volume as default) screws up the device. Although this has been
fixed in the newer firmware, the devices are shipped with the old
firmware, thus we need the workaround in the driver side. -- tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit e549d190f7b5f94e9ab36bd965028112914d010d ]
The front mic jack (pink color) can't detect any plug or unplug. After
applying this fix, both detecting function and recording function
work well.
BugLink: https://bugs.launchpad.net/bugs/1564712
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit b4203ff5464da00b7812e7b480192745b0d66bbf ]
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit f03b24a851d32ca85dacab01785b24a7ee717d37 ]
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
usb 2-1.3: 2:1: cannot get freq at ep 0x85
usb 2-1.3: 1:1: cannot get freq at ep 0x2
....
Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 12a6116e66695a728bcb9616416c508ce9c051a1 ]
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 2f80b2958abe5658000d5ad9b45a36ecf879666e ]
This quirk allows us to avoid the noisy:
current rate 0 is different from the runtime rate
message every time playback starts. While USB DAC in the RR2150
supports reading the sample rate, it never returns a sample rate
other than zero in my observation with common sample rates.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 023d8218ec0dfc30e11d4ec54f640e8f127d1fbe ]
The commit [bd48128539ab: ALSA: hda - Fix forgotten HDMI
monitor_present update] covered the missing update of monitor_present
flag, but this caused a regression for devices without the i915 eld
notifier. Since the old code supposed that pin_eld->monitor_present
was updated by the caller side, the hdmi_present_sense_via_verbs()
doesn't update the temporary eld->monitor_present but only
pin_eld->monitor_present, which is now overridden in update_eld().
The fix is to update pin_eld->monitor_present as well before calling
update_eld().
Note that this may still leave monitor_present flag in an inconsistent
state when the driver repolls, but this is at least the old behavior.
More proper fix will follow in the later patch.
Fixes: bd48128539ab ('ALSA: hda - Fix forgotten HDMI monitor_present update')
Signed-off-by: Hyungwon Hwang <hyungwon.hwang7@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 0ef21100ae912f76ed89f76ecd894f4ffb3689c1 ]
The Microsoft HD-5001 webcam microphone does not support sample rate
reading as the HD-5000 one.
This results in dmesg errors and sound hanging with pulseaudio.
Signed-off-by: Victor Clément <victor.clement@openmailbox.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 0f886ca12765d20124bd06291c82951fd49a33be ]
create_fixed_stream_quirk() may cause a NULL-pointer dereference by
accessing the non-existing endpoint when a USB device with a malformed
USB descriptor is used.
This patch avoids it simply by adding a sanity check of bNumEndpoints
before the accesses.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 1f7c6658962fa1260c1658d681bd6bb0c746b99a ]
Cirrus HD-audio driver may adjust GPIO pins for EAPD dynamically
depending on the jack plug state. This works fine for the auto-mute
mode where the speaker gets muted upon the HP jack plug. OTOH, when
the auto-mute mode is off, this turns off the EAPD unexpectedly
depending on the jack state, which results in the silent speaker
output.
This patch fixes the silent speaker output issue by setting GPIO bits
constantly when the auto-mute mode is off.
Reported-and-tested-by: moosotc@gmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 4061db03dd71d195b9973ee466f6ed32f6a3fc16 ]
The clock measurement on the AC'97 audio card found in the IBM ThinkPad X41
will often fail, so add a quirk entry to fix it.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=441087
Cc: <stable@vger.kernel.org>
Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 3ec622f40913ae036f218e5e7e92df9c1f1753d9 ]
Vendor ID 0x10de0083 is used by a yet-to-be-named GPU chip.
This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is
appropriate here.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 2d369c748c2ecc2a012ee85412a04007e67913ec ]
Vendor ID 0x10de0082 is used by a yet-to-be-named GPU chip.
This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is
appropriate here.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 6c3d91193d829bf58a35a10650415b05a736ca6c ]
Vendor ID 0x10de007d is used by a yet-to-be-named GPU chip.
This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is
appropriate here.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 6ef2f68fa38bf415830f67903d87180d933e0f47 ]
This Lenovo ThinkCentre AIO also uses Line2 as mic mute button and
uses GPIO2 to control the mic mute led, so applying this quirk can
make both the button and led work.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1555912
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit e7fdd52779a6c2b49d457f452296a77c8cffef6a ]
Many codecs, typically found on Realtek codecs, have the analog
loopback path merged to the secondary input of the middle of the
output paths. Currently, we don't offer the dynamic switching in such
configuration but let each loopback path mute by itself.
This should work well in theory, but in reality, we often see that
such a dead loopback path causes some background noises even if all
the elements get muted. Such a problem has been fixed by adding the
quirk accordingly to disable aamix, and it's the right fix, per se.
The only problem is that it's not so trivial to achieve it; user needs
to pass a hint string via patch module option or sysfs.
This patch gives a bit improvement on the situation: it adds "Loopback
Mixing" control element for such codecs like other codecs (e.g. IDT or
VIA codecs) with the individual loopback paths. User can turn on/off
the loopback path simply via a mixer app.
For keeping the compatibility, the loopback is still enabled on these
codecs. But user can try to turn it off if experiencing a suspicious
background or click noise on the fly, then build a static fixup later
once after the problem is addressed.
Other than the addition of the loopback enable/disablement control,
there should be no changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 27c41dad3a012c5acead1d903d1743297457b69c ]
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 56f27013482c0803d978b667fe85de04ce9357cd ]
Inform userspace that one channel of the internal mic has reversed
polarity, so it does not attempt to add both channels together and
end up with silence.
Cc: stable@vger.kernel.org
Reported-by: Andrzej Mendel <andrzej.mendel@gmail.com>
Alsa-info: http://www.alsa-project.org/db/?f=3088f82a0cf977855f92af9db8ad406c04f71efa
BugLink: https://bugs.launchpad.net/bugs/1529624
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 15c665700bf6f4543f003ac0fbb1e9ec692e93f2 ]
The firmware ctls like "DSP1 Firmware" in wm_adsp codec driver are
enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 8019c0b37cd5a87107808300a496388b777225bf ]
The DRC Mode like "AIF1DRC1 Mode" and EQ Mode like "AIF1.1 EQ Mode" in
wm8994 codec driver are enum ctls, while the current driver accesses
wrongly via value.integer.value[]. They have to be via
value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit d0784829ae3b0beeb69b476f017d5c8a2eb95198 ]
"MBC Mode", "VSS Mode", "VSS HPF Mode" and "Enhanced EQ Mode" ctls in
wm8958 codec driver are enum, while the current driver accesses
wrongly via value.integer.value[]. They have to be via
value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 02322ac9dee9aff8d8862e8d6660ebe102f492ea ]
This patch applies the microphone-related fix created for the Acer
Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282
CODEC and demonstrates the same issues.
This patch allows an external, headset microphone to be used and limits
the gain on the (quite noisy) internal microphone.
Signed-off-by: Simon South <simon@simonsouth.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 197b958c1e76a575d77038cc98b4bebc2134279f ]
The OSS sequencer client tries to drain the pending events at
releasing. Unfortunately, as spotted by syzkaller fuzzer, this may
lead to an unkillable process state when the event has been queued at
the far future. Since the process being released can't be signaled
any longer, it remains and waits for the echo-back event in that far
future.
Back to history, the draining feature was implemented at the time we
misinterpreted POSIX definition for blocking file operation.
Actually, such a behavior is superfluous at release, and we should
just release the device as is instead of keeping it up forever.
This patch just removes the draining call that may block the release
for too long time unexpectedly.
BugLink: http://lkml.kernel.org/r/CACT4Y+Y4kD-aBGj37rf-xBw9bH3GMU6P+MYg4W1e-s-paVD2pg@mail.gmail.com
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 17e2df4613be57d0fab68df749f6b8114e453152 ]
Plantronics DA45 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x4" and "cannot get freq at
ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit eab3c4db193f5fcccf70e884de9a922ca2c63d80 ]
snd-hdsp driver accesses enum item values (int) instead of boolean
values (long) wrongly for some ctl elements. This patch fixes them.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit c1099c3294c2344110085a38c50e478a5992b368 ]
HDSPM driver contains a code issuing zero-division potentially in
system sample rate ctl code. This patch fixes it by not processing
a zero or invalid rate value as a divisor, as well as excluding the
invalid value to be passed via the given ctl element.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit 537e48136295c5860a92138c5ea3959b9542868b ]
snd-hdspm driver accesses enum item values (int) instead of boolean
values (long) wrongly for some ctl elements. This patch fixes them.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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[ Upstream commit b24e7ad1fdc22177eb3e51584e1cfcb45d818488 ]
X32 ABI takes the 64bit timespec, thus the timer user status ioctl becomes
incompatible with IA32. This results in NOTTY error when the ioctl is
issued.
Meanwhile, this struct in X32 is essentially identical with the one in
X86-64, so we can just bypassing to the existing code for this
specific compat ioctl.
Cc: <stable@vger.kernel.org> # v3.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sasha.levin@oracle.com>
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