commit a523ef731ac6674dc07574f31bf44cc5bfa14e4d upstream.
kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Based on similar fix in kbl_rt5663_rt5514_max98927.c
from Harsha Priya <harshapriya.n@intel.com> and
Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Cc: <stable@vger.kernel.org> # 5.4+
Signed-off-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210415124347.475432-1-lma@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d58970da324732686529655c21791cef0ee547c4 upstream.
cppcheck warning:
sound/soc/samsung/tm2_wm5110.c:605:6: style: Variable 'ret' is
reassigned a value before the old one has been
used. [redundantAssignment]
ret = devm_snd_soc_register_component(dev, &tm2_component,
^
sound/soc/samsung/tm2_wm5110.c:554:7: note: ret is assigned
ret = of_parse_phandle_with_args(dev->of_node, "i2s-controller",
^
sound/soc/samsung/tm2_wm5110.c:605:6: note: ret is overwritten
ret = devm_snd_soc_register_component(dev, &tm2_component,
^
The args is a stack variable, so it could have junk (uninitialized)
therefore args.np could have a non-NULL and random value even though
property was missing. Later could trigger invalid pointer dereference.
There's no need to check for args.np because args.np won't be
initialized on errors.
Fixes: 8d1513cef5 ("ASoC: samsung: Add support for HDMI audio on TM2 board")
Cc: <stable@vger.kernel.org>
Suggested-by: Krzysztof Kozlowski <krzk@kernel.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210312180231.2741-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 970e3012c04c96351c413f193a9c909e6d871ce2 upstream.
This applies a SND_PCI_QUIRK(...) to the Clevo PCx0Dx barebones. This
fix enables audio output over the headset jack and ensures that a
microphone connected via the headset combo jack is correctly recognized
when pluged in.
[ Rearranged the list entries in a sorted order -- tiwai ]
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Co-developed-by: Werner Sembach <wse@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210427153025.451118-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9bbb94e57df135ef61bef075d9c99b8d9e89e246 upstream.
Remove a duplicate vendor+subvendor pin fixup entry as one is masking
the other and making it unreachable. Consider the more specific newcomer
as a second chance instead.
The generic entry is made less strict to also match for laptops with
slightly different 0x12 pin configuration. Tested on Lenovo Yoga 6 (AMD)
where 0x12 is 0x40000000.
Fixes: 607184cb1635 ("ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button")
Signed-off-by: Sami Loone <sami@loone.fi>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/YIXS+GT/dGI/LtK6@yoga
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1c9d9dfd2d254211cb37b1513b1da3e6835b8f00 upstream.
Boot with plugged headset, the Headset Mic will be gone.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/207eecfc3189466a820720bc0c409ea9@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d1ee66c5d3c5a0498dd5e3f2af5b8c219a98bba5 upstream.
Fix two bugs with the Intel HDA Realtek ALC233 sound codec
present in Intel NUC NUC8i7BEH and probably a few other similar
NUC models.
These codecs advertise a 4-level microphone input boost amplifier on
pin 0x19, but the highest two boost settings do not work correctly,
and produce only low analog noise that does not seem to contain any
discernible signal. There is an existing fixup for this exact problem
but for a different PCI subsystem ID, so we re-use that logic.
Changing the boost level also triggers a DC spike in the input signal
that bleeds off over about a second and overwhelms any input during
that time. Thankfully, the existing fixup has the side effect of
making the boost control show up in userspace as a mute/unmute switch,
and this keeps (e.g.) PulseAudio from fiddling with it during normal
input volume adjustments.
Finally, the NUC hardware has built-in inverted stereo mics. This
patch also enables the usual fixup for this so the two channels cancel
noise instead of the actual signal.
[ Re-ordered the quirk entry point by tiwai ]
Signed-off-by: Phil Calvin <phil@philcalvin.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/80dc5663-7734-e7e5-25ef-15b5df24511a@philcalvin.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 76fae6185f5456865ff1bcb647709d44fd987eb6 upstream.
The GA503 has almost exactly the same default setup as the GA401
model with the same issues. The GA401 quirks solve all the issues
so we will use the full quirk chain.
Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210419030411.28304-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 75b62ab65d2715ce6ff0794033d61ab9dc4a2dfc upstream.
The HP ProBook 445 G7 (17T32ES) uses ALC236. Like ALC236_FIXUP_HP_GPIO_LED,
COEF index 0x34 bit 5 is used to control the playback mute LED, but the
microphone mute LED is controlled using pin VREF instead of a COEF index.
AlsaInfo: https://alsa-project.org/db/?f=0d3f4d1af39cc359f9fea9b550727ee87e5cf45a
Signed-off-by: Jonas Witschel <diabonas@archlinux.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210416105852.52588-1-diabonas@archlinux.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ab2165e2e6ed17345ffa8ee88ca764e8788ebcd7 upstream.
The decibel volume range contains a negative maximum value resulting in
pipewire complaining about the device and effectivly having no sound
output. The wrong values also resulted in the headset sounding muted
already at a mixer level of about ~25%.
PipeWire BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1049
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212897
Signed-off-by: Timo Gurr <timo.gurr@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210503110822.10222-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 upstream.
In the current code, we have some assumption that the audio clock
selector has been set up implicitly and don't want to touch it unless
it's really needed for the fallback autoclock setup. This works for
most devices but some seem having a problem. Partially this was
covered for the devices with a single connector at the initialization
phase (commit 086b957cc17f "ALSA: usb-audio: Skip the clock selector
inquiry for single connections"), but also there are cases where the
wrong clock set up is kept silently. The latter seems to be the cause
of the noises on Behringer devices.
In this patch, we explicitly set up the audio clock selector whenever
the appropriate node is found.
Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327
Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4fb44dd2c1dda18606348acdfdb97e8759dde9df upstream.
In snd_sb_qsound_build, snd_ctl_add(..,p->qsound_switch...) and
snd_ctl_add(..,p->qsound_space..) are called. But the second
arguments of snd_ctl_add() could be freed via snd_ctl_add_replace()
->snd_ctl_free_one(). After the error code is returned,
snd_sb_qsound_destroy(p) is called in __error branch.
But in snd_sb_qsound_destroy(), the freed p->qsound_switch and
p->qsound_space are still used by snd_ctl_remove().
My patch set p->qsound_switch and p->qsound_space to NULL if
snd_ctl_add() failed to avoid the uaf bugs. But these codes need
to further be improved with the code style.
Signed-off-by: Lv Yunlong <lyl2019@mail.ustc.edu.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426145541.8070-1-lyl2019@mail.ustc.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2e6a731296be9d356fdccee9fb6ae345dad96438 upstream.
Just re-order the cx5066_fixups[] entries for HP devices for avoiding
the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210428112704.23967-14-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1c98f574403dbcf2eb832d5535a10d967333ef2d upstream.
Our code analyzer reported a uaf.
In snd_emu8000_create_mixer, the callee snd_ctl_add(..,emu->controls[i])
calls snd_ctl_add_replace(.., kcontrol,..). Inside snd_ctl_add_replace(),
if error happens, kcontrol will be freed by snd_ctl_free_one(kcontrol).
Then emu->controls[i] points to a freed memory, and the execution comes
to __error branch of snd_emu8000_create_mixer. The freed emu->controls[i]
is used in snd_ctl_remove(card, emu->controls[i]).
My patch set emu->controls[i] to NULL if snd_ctl_add() failed to avoid
the uaf.
Signed-off-by: Lv Yunlong <lyl2019@mail.ustc.edu.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426131129.4796-1-lyl2019@mail.ustc.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ]
When using the driver in I2S TDM mode, the fsl_esai_startup()
function rewrites the number of slots previously set by the
fsl_esai_set_dai_tdm_slot() function to 2.
To fix this, let's use the saved slot count value or, if TDM
is not used and the number of slots is not set, the driver will use
the default value (2), which is set by fsl_esai_probe().
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3a27875e91fb9c29de436199d20b33f9413aea77 ]
Amp requires 10 ~ 30ms for the power ON and OFF.
Added 30ms delay for stability.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a23f9099ff1541f15704e96b784d3846d2a4483d ]
0x20FF(amp global enable) register was defined as non-volatile,
but it is not. Overheating, overcurrent can cause amp shutdown
in hardware.
'regmap_write' compare register readback value before writing
to avoid same value writing. 'regmap_read' just read cache
not actual hardware value for the non-volatile register.
When amp is internally shutdown by some reason, next 'AMP ON'
command can be ignored because regmap think amp is already ON.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ]
card->owner is a required property and since commit 81033c6b58 ("ALSA:
core: Warn on empty module") a warning is issued if it is empty. Add it.
This fixes following warning observed on Lamobo R1:
WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd]
Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ...
CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1
Hardware name: Allwinner sun7i (A20) Family
Call trace:
(snd_card_new [snd])
(snd_soc_bind_card [snd_soc_core])
(snd_soc_register_card [snd_soc_core])
(sun4i_codec_probe [sun4i_codec])
Fixes: 45fb6b6f2a ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs")
Related: commit 3c27ea23ff ("ASoC: qcom: Set card->owner to avoid warnings")
Related: commit ec653df2a0 ("drm/vc4/vc4_hdmi: fill ASoC card owner")
Cc: linux-arm-kernel@lists.infradead.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Bastian Germann <bage@linutronix.de>
Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 16b82e75c15a7dbd564ea3654f3feb61df9e1e6f ]
The input MCLK is 12.288MHz, the desired output sysclk is 11.2896MHz
and sample rate is 44100Hz, with the configuration pllprescale=2,
postscale=sysclkdiv=1, some chip may have wrong bclk
and lrclk output with pll enabled in master mode, but with the
configuration pllprescale=1, postscale=2, the output clock is correct.
>From Datasheet, the PLL performs best when f2 is between
90MHz and 100MHz when the desired sysclk output is 11.2896MHz
or 12.288MHz, so sysclkdiv = 2 (f2/8) is the best choice.
So search available sysclk_divs from 2 to 1 other than from 1 to 2.
Fixes: 84fdc00d51 ("ASoC: codec: wm9860: Refactor PLL out freq search")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1616150926-22892-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 927280909fa7d8e61596800d82f18047c6cfbbe4 ]
When checking for enabled cores it isn't enough to check that
some of the requested cores are running, we have to check that
all of them are.
Fixes: 747503b181 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations")
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210322163728.16616-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit aa65bacdb70e549a81de03ec72338e1047842883 upstream.
The SST firmware's media and deep-buffer inputs are hardcoded to
S16LE, the corresponding DAIs don't have a hw_params callback and
their prepare callback also does not take the format into account.
So far the advertising of non working S24LE support has not caused
issues because pulseaudio defaults to S16LE, but changing pulse-audio's
config to use S24LE will result in broken sound.
Pipewire is replacing pulse now and pipewire prefers S24LE over S16LE
when available, causing the problem of the broken S24LE support to
come to the surface now.
Cc: stable@vger.kernel.org
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/866
Fixes: 098c2cd281 ("ASoC: Intel: Atom: add 24-bit support for media playback and capture")
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210324132711.216152-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c6423ed2da6214a68527446b5f8e09cf7162b2ce upstream.
There is another HP ZBook G5 model with the PCI SSID 103c:844f that
requires the same quirk for controlling the mute LED. Add the
corresponding entry to the quirk table.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212407
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210401171314.667-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c8426b2700b57d2760ff335840a02f66a64b6044 upstream.
We've got a report about Acer Aspire E1 (PCI SSID 1025:0840) that
loses the speaker output after resume. With the comparison of COEF
dumps, it was identified that the COEF 0x0d bits 0x6000 corresponds to
the speaker amp.
This patch adds the specific quirk for the device to restore the COEF
bits at the codec (re-)initialization.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1183869
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210407095730.12560-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 168632a495f49f33a18c2d502fc249d7610375e9 upstream.
Add a control to the card before copying the id so that the numid field
is initialized in the copy. Otherwise the numid field of active_id,
format_id, rate_id and channels_id will be the same (0) and
snd_ctl_notify() will not queue the events properly.
Signed-off-by: Jonas Holmberg <jonashg@axis.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210407075428.2666787-1-jonashg@axis.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 417eadfdd9e25188465280edf3668ed163fda2d0 upstream.
The HP EliteBook 640 G8 Notebook PC is using ALC236 codec which is
using 0x02 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210330114428.40490-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e54f30befa7990b897189b44a56c1138c6bfdbb5 upstream.
We found the alc_update_headset_mode() is not called on some machines
when unplugging the headset, as a result, the mode of the
ALC_HEADSET_MODE_UNPLUGGED can't be set, then the current_headset_type
is not cleared, if users plug a differnt type of headset next time,
the determine_headset_type() will not be called and the audio jack is
set to the headset type of previous time.
On the Dell machines which connect the dmic to the PCH, if we open
the gnome-sound-setting and unplug the headset, this issue will
happen. Those machines disable the auto-mute by ucm and has no
internal mic in the input source, so the update_headset_mode() will
not be called by cap_sync_hook or automute_hook when unplugging, and
because the gnome-sound-setting is opened, the codec will not enter
the runtime_suspend state, so the update_headset_mode() will not be
called by alc_resume when unplugging. In this case the
hp_automute_hook is called when unplugging, so add
update_headset_mode() calling to this function.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210320091542.6748-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit febf22565549ea7111e7d45e8f2d64373cc66b11 upstream.
We found a recording issue on a Dell AIO, users plug a headset-mic and
select headset-mic from UI, but can't record any sound from
headset-mic. The root cause is the determine_headset_type() returns a
wrong type, e.g. users plug a ctia type headset, but that function
returns omtp type.
On this machine, the internal mic is not connected to the codec, the
"Input Source" is headset mic by default. And when users plug a
headset, the determine_headset_type() will be called immediately, the
codec on this AIO is alc274, the delay time for this codec in the
determine_headset_type() is only 80ms, the delay is too short to
correctly determine the headset type, the fail rate is nearly 99% when
users plug the headset with the normal speed.
Other codecs set several hundred ms delay time, so here I change the
delay time to 850ms for alc2x4 series, after this change, the fail
rate is zero unless users plug the headset slowly on purpose.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210320091542.6748-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 66affb7bb0dc0905155a1b2475261aa704d1ddb5 upstream.
The recently added PM prepare and complete callbacks don't have the
sanity check whether the card instance has been properly initialized,
which may potentially lead to Oops.
This patch adds the azx_is_pm_ready() call in each place
appropriately like other PM callbacks.
Fixes: f5dac54d9d ("ALSA: hda: Separate runtime and system suspend")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210329113059.25035-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c8f79808cd8eb5bc8d14de129bd6d586d3fce0aa upstream.
The card power state change via snd_power_change_state() at the system
suspend/resume seems dropped mistakenly during the PM code rewrite.
The card power state doesn't play much role nowadays but it's still
referred in a few places such as the HDMI codec driver.
This patch restores them, but in a more appropriate place now in the
prepare and complete callbacks.
Fixes: f5dac54d9d ("ALSA: hda: Separate runtime and system suspend")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210329113059.25035-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 625bd5a616ceda4840cd28f82e957c8ced394b6a upstream.
Logitech ConferenceCam Connect is a compound USB device with UVC and
UAC. Not 100% reproducible but sometimes it keeps responding STALL to
every control transfer once it receives get_freq request.
This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419
Signed-off-by: Ikjoon Jang <ikjn@chromium.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 899b12542b0897f92de9ba30944937c39ebb246d ]
We do some IO operations in the snd_soc_component_set_jack callback
function and snd_soc_component_set_jack() will be called when soc
component is removed. However, we should not access SoundWire registers
when the bus is suspended.
So set regcache_cache_only(regmap, true) to avoid accessing in the
soc component removal process.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ]
Simple-card/audio-graph-card drivers do not handle MCLK clock when it
is specified in the codec device node. The expectation here is that,
the codec should actually own up the MCLK clock and do necessary setup
in the driver.
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ]
This delay is part of the power-up sequence defined in the datasheet.
A runtime_resume is a power-up so must also include the delay.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ]
The minimum value is 0x3f (-63dB), which also is mute
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ]
Remove the hard coded 32 bits width and replace with the correct width
calculated by params_width.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ]
The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd ]
Many systems do not use ACPI and hence do not provide a DMI table. On
non-ACPI systems a warning, such as the following, is printed on boot.
WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name!
The variable 'dmi_available' is not exported and so currently cannot be
used by kernel modules without adding an accessor. However, it is
possible to use the function is_acpi_device_node() to determine if the
sound card is an ACPI device and hence indicate if we expect a DMI table
to be present. Therefore, call is_acpi_device_node() to see if we are
using ACPI and only parse the DMI table if we are booting with ACPI.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ]
Most steps in this table are steps of 3dB (300 centi-dB), so we can
simplify the table.
This not only reduces the amount of space it takes inside the kernel,
this also makes alsa-lib's mixer code actually accept the table, where
as before this change alsa-lib saw the "ADC PGA Gain" control as a
control without a dB scale.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ]
According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has
the following bit field definitions:
| BITS | FIELD | RW | RESET | DEFINITION |
| 15 | RSVD | RO | 0x0 | Reserved |
| 14 | RSVD | RW | 0x1 | Reserved |
| 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode |
| 11:10 | RSVD | RO | 0x0 | Reserved |
| 9:8 | LBI_RESP | RW | 0x1 | Integrator Response |
| 7:6 | RSVD | RO | 0x0 | Reserved |
| 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode |
| 4:1 | RSVD | RO | 0x0 | Reserved |
| 0 | EN | RW | 0x0 | Enable/Disable AVC |
The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED. It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0. While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.
The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'. Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.
Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.
[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf
Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ]
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b5 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ]
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b5 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a14a6219996ee6f6e858d83b11affc7907633687 ]
On some Lenovo systems if the microphone is disabled in the BIOS
only the NHLT table header is created, with no data. This means
the endpoints field is not correctly set to zero - leading to an
unintialised variable and hence invalid descriptors are parsed
leading to page faults.
The Lenovo firmware team is addressing this, but adding a check
preventing invalid tables being parsed is worthwhile.
Tested on a Lenovo T14.
Tested-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Reported-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Signed-off-by: Mark Pearson <markpearson@lenovo.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 50b1affc891cbc103a2334ce909a026e25f4c84d upstream.
The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.
Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c7990 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8ca88d53351cc58d535b2bfc7386835378fb0db2 upstream.
This reverts commit 1e30f642cf29 ("ASoC: simple-card-utils: Fix device
module clock"). The original patch ended up breaking following platform,
which depends on set_sysclk() to configure internal PLL on wm8904 codec
and expects simple-card-utils to not update the MCLK rate.
- "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"
It would be best if codec takes care of setting MCLK clock via DAI
set_sysclk() callback.
Reported-by: Michael Walle <michael@walle.cc>
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Fixes: 1e30f642cf29 ("ASoC: simple-card-utils: Fix device module clock")
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9922f50f7178496e709d3d064920b5031f0d9061 upstream.
The max boundary check while parsing dai ids makes
sound card registration fail after common up dai ids.
Fixes: cd3484f7f138 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY")
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3bb4852d598f0275ed5996a059df55be7318ac2f upstream.
set channel map can be passed with a channel maps, however if
the number of channels that are passed are more than the actual
supported channels then we would be accessing array out of bounds.
So add a sanity check to validate these numbers!
Fixes: a61f3b4f47 ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4800fe6ea1022eb240215b1743d2541adad8efc7 upstream.
WCD934x has only 13 RX SLIM ports however we are setting it as 16
in set_channel_map, this will lead to array out of bounds error!
Orignally caught by enabling USBAN array out of bounds check:
Fixes: 5caf64c633 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1c668e1c0a0f74472469cd514f40c9012b324c31 upstream.
Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.
Fix this by properly using AFE_PORT_MAX macro.
Fixes: 1b93a88431 ("ASoC: qcom: sdm845: handle soundwire stream")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fd8299181995093948ec6ca75432e797b4a39143 upstream.
The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered
down, we need to check ADSPCS_CPA, the Current Power Active bit.
Fixes: 747503b181 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations")
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
