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commit 45461e3b554c75ddff9703539f3711cc3dfb0422 upstream.
Just re-order the alc269_fixup_tbl[] entries for HP devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Formerly, some entries were grouped for the actual codec, but this
doesn't seem reasonable to keep in that way. So now we simply keep
the PCI SSID order for the whole.
Also Cc-to-stable for the further patch applications.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210428112704.23967-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 13e1a4cd490b959a4c72c9f4fb502ef56b190062 upstream.
Just re-order the alc882_fixup_tbl[] entries for Clevo devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also, user lower hex letters in the entry.
Also Cc-to-stable for the further patch applications.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210428112704.23967-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b7529c18feecb1af92f9db08c8e7fe446a82d96d upstream.
Just re-order the alc882_fixup_tbl[] entries for Sony 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/20210428112704.23967-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b265047ac56bad8c4f3d0c8bf9cb4e828ee0d28e upstream.
Just re-order the alc882_fixup_tbl[] entries for Acer 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/20210428112704.23967-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: 8d1513cef51a ("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 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 a01df925d1bbc97d6f7fe07b157aadb565315337 upstream.
Apply const prefix to the remaining places: the static table for the
unit information, the mixer maps, the validator tables, etc.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-12-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 7c0d6e482062eb5c06ecccfab340abc523bdca00 ]
card->owner is a required property and since commit 81033c6b584b ("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: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs")
Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings")
Related: commit ec653df2a0cb ("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: 84fdc00d519f ("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>
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: 098c2cd281409 ("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 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 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 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 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 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 3f31f7d9b540 ("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 3f31f7d9b540 ("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>
commit 87263968516fb9507d6215d53f44052627fae8d8 upstream.
When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt()
function rewrites the number of slots previously set by the
fsl_ssi_set_dai_tdm_slot() function to 2 by default.
To fix this, let's use the saved slot count value or, if TDM
is not used and the slot count is not set, proceed as before.
Fixes: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2bf44e0ee95f39cc54ea1b942f0a027e0181ca4e upstream.
Recently we found the micmute led init state is not correct after
freshly installing the ubuntu linux on a Lenovo AIO machine. The
internal mic is not muted, but the micmute led is on and led mode is
'follow mute'. If we mute internal mic, the led is keeping on, then
unmute the internal mic, the led is off. And from then on, the
micmute led will work correctly.
So the micmute led init state is not correct. The led is controlled
by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the
patch_realtek, the gpio data is set to 0x4 initially and the led is
on with this data. In the hda_generic, the led_value is set to
0 initially, suppose users set the 'capture switch' to on from
user space and the micmute led should change to be off with this
operation, but the check "if (val == spec->micmute_led.led_value)" in
the call_micmute_led_update() will skip the led setting.
To guarantee the led state will be set by the 1st time of changing
"Capture Switch", set -1 to the init led_value.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210312041408.3776-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 80cffd2468ddb850e678f17841fc356930b2304a upstream.
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4ec5b96775a88dd9b1c3ba1d23c43c478cab95a2 upstream.
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 06abcb18b3a021ba1a3f2020cbefb3ed04e59e72 upstream.
Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fec60c3bc5d1713db2727cdffc638d48f9c07dc3 upstream.
Dell AE515 sound bar (413c:a506) spews the error messages when the
driver tries to read the current sample frequency, hence it needs to
be on the list in snd_usb_get_sample_rate_quirk().
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211551
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304083021.2152-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5ff9dde42e8c72ed8102eb8cb62e03f9dc2103ab upstream.
When HD-audio bus receives unsolicited events during its system
suspend/resume (S3 and S4) phase, the controller driver may still try
to process events although the codec chips are already (or yet)
powered down. This might screw up the codec communication, resulting
in CORB/RIRB errors. Such events should be rather skipped, as the
codec chip status such as the jack status will be fully refreshed at
the system resume time.
Since we're tracking the system suspend/resume state in codec
power.power_state field, let's add the check in the common unsol event
handler entry point to filter out such events.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org> # 183ab39eb0ea: ALSA: hda: Initialize power_state
Link: https://lore.kernel.org/r/20210310112809.9215-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 28e96c1693ec1cdc963807611f8b5ad400431e82 upstream.
The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on
AMD controller (1022:1457)") introduced a few workarounds for the
recent AMD HD-audio controller, and one of them is the forced BATCH
PCM mode so that PulseAudio avoids the timer-based scheduling. This
was thought to cover for some badly working applications, but this
actually worsens for more others. In total, this wasn't a good idea
to enforce it.
This is a partial revert of the commit above for dropping the PCM
BATCH enforcement part to recover from the regression again.
Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210308160726.22930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit eea46a0879bcca23e15071f9968c0f6e6596e470 upstream.
The per_pin->work might be still floating at the suspend, and this may
hit the access to the hardware at an unexpected timing. Cancel the
work properly at the suspend callback for avoiding the buggy access.
Note that the bug doesn't trigger easily in the recent kernels since
the work is queued only when the repoll count is set, and usually it's
only at the resume callback, but it's still possible to hit in
theory.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fc7c5c208eb7bc2df3a9f4234f14eca250001cb6 upstream.
The microphone in the Plantronics C320-M headset will randomly
fail to initialize properly, at least when using Microsoft Teams.
Introducing a 20ms delay on the control messages appears to
resolve the issue.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065
Tested-by: Andreas Kempe <kempe@lysator.liu.se>
Signed-off-by: John Ernberg <john.ernberg@actia.se>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303181405.39835-1-john.ernberg@actia.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ]
Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing,
thus the heaset functions are not working.
Add quirk for this model to select the correct input map, jack-detect
options and channel map to enable jack sensing and headset microphone.
This device uses IN1 for its internal MIC and JD2 for jack-detect.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26a9630c72ebac7c564db305a6aee54a8edde70e ]
Currently the mask operation on variable conf is just 3 bits so
the switch statement case value of 8 is unreachable dead code.
The function daio_mgr_dao_init can be passed a 4 bit value,
function dao_rsc_init calls it with conf set to:
conf = (desc->msr & 0x7) | (desc->passthru << 3);
so clearly when desc->passthru is set to 1 then conf can be
at least 8.
Fix this by changing the mask to 0xf.
Fixes: 8cc72361481f ("ALSA: SB X-Fi driver merge")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210227001527.1077484-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 26af17722a07597d3e556eda92c6fce8d528bc9f upstream.
There is another MSI board (1462:cc34) that has dual Realtek codecs,
and we need to apply the existing quirk for fixing the conflicts of
Master control.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303142346.28182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 48698c973e6b4dde94d87cd1ded56d9436e9c97d upstream.
This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This
fixes the issue of the device not recognizing a pluged in microphone.
The device has both, a microphone only jack, and a speaker + microphone
combo jack. The combo jack already works. The microphone-only jack does
not recognize when a device is pluged in without this patch.
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/0eee6545-5169-ef08-6cfa-5def8cd48c86@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ]
The Acer One S1002 tablet is using an analog mic on IN1 and has
its jack-detect connected to JD2_IN4N, instead of using the default
IN3 for its internal mic and JD1_IN4P for jack-detect.
Note it is also using AIF2 instead of AIF1 which is somewhat unusual,
this is correctly advertised in the ACPI CHAN package, so the speakers
do work without the quirk.
Add a quirk for the mic and jack-detect settings.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ]
The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using
SSP2 (AIF1) and it mostly works with the defaults. But instead of using
DMIC1 it is using an analog mic on IN1, add a quirk for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ]
The Estar Beauty HD MID 7316R tablet almost fully works with out default
settings. The only problem is that it has only 1 speaker so any sounds
only playing on the right channel get lost.
Add a quirk for this model using the default settings + MONO_SPEAKER.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4841b8e6318a7f0ae57c4e5ec09032ea057c97a8 upstream.
Modify 0x20 index 7 bit 5 to 1, make the 0x15 EAPD the same as 0x14.
Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/e62c5058957f48d8b8953e97135ff108@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit de5bfae2fd962a9da99f56382305ec7966a604b9 ]
The correct mask is 0x1f8 (Bit 3-8), but due to missing BIT() 0xf (Bit
0-3) was set instead. This means setting of CPCAP_BIT_MIC1_RX_TIMESLOT0
(Bit 3) still worked (part of both masks). On the other hand the code
does not properly clear the other MIC timeslot bits. I think this
is not a problem, since they are probably initialized to 0 and not
touched by the driver anywhere else. But the mask also contains some
wrong bits, that will be cleared. Bit 0 (CPCAP_BIT_SMB_CDC) should be
safe, since the driver enforces it to be 0 anyways.
Bit 1-2 are CPCAP_BIT_FS_INV and CPCAP_BIT_CLK_INV. This means enabling
audio recording forces the codec into SND_SOC_DAIFMT_NB_NF mode, which
is obviously bad.
The bug probably remained undetected, because there are not many use
cases for routing microphone to the CPU on platforms using cpcap and
user base is small. I do remember having some issues with bad sound
quality when testing voice recording back when I wrote the driver.
It probably was this bug.
Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Reviewed-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20210123172945.3958622-1-sre@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 856fe64da84c95a1d415564b981ae3908eea2a76 ]
There are two issues with this code. The first error path forgot to set
the error code and instead returns success. The second error path
doesn't clean up.
Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X9NE/9nK9/TuxuL+@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
