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info.mode and info.port are indirectly controlled by user-space,
hence leading to a potential exploitation of the Spectre variant 1
vulnerability.
These issues were detected with the help of Smatch:
sound/synth/emux/emux_hwdep.c:72 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[i]->ctrls' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[info.port]->ctrls' [w] (local cap)
Fix this by sanitizing both info.mode and info.port before using them
to index emu->portptrs[i]->ctrls, emu->portptrs[info.port]->ctrls and
emu->portptrs.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull refactoring / fixes of HD-audio PM and display power management
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fireface 800 is a flagship model of RME GmbH for audio and music units
on IEEE 1394 bus, shipped 2004. This model consists of four chips:
- TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus
- TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus
- Xilinx Spartan-3 FPGA XC3S400
- Xilinx High-Performance CPLD XC9572XL
This commit adds support Fireface 800. In this time, the support is
restricted to its MIDI functionality, thus this commit adds some
condition statements to avoid touching streaming functionality.
Unlike Fireface 400, Fireface 800 has no functionality to suppress
asynchronous transactions for MIDI messages except for unregister of
listen address in controller side, thus the feature is available as is.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Content of asynchronous transaction for MIDI messages differs between
Fireface 400 and 800.
This commit adds a model-specific handler for the transaction and adds
arrangement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fireface 400 and 800 have the same mechanism to decide address to which
asynchronous transactions are sent for MIDI messages, however they use
different registers for controllers to notify higher 4 byte of the
address.
This commit adds a model-specific parameter to represent the address.
Additionally, it corrects some comments. I note that these two models have
a difference to enable/disable the transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As long as investigating packet dumps from Fireface 400/800, a register
to receive asynchronous transactions for MIDI messages is the same. For
Fireface 800, minor register is used.
This commit declares macros for the transactions and obsoletes
model-specific parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Unlike Fireface 400, Fireface 800 have two pair of optical interface
for ADAT signal and S/PDIF signal. ADAT signals for the interface
are handled for sampling clock source separately.
This commit modifies a parser for clock configuration to distinguish
these two ADAT signals.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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source
As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for a register of clock configuration to
obsolete model-specific operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for the registers to obsolete model-specific
operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As long as investigating packet dumps from Fireface 400/800, status
registers for clock synchronization is common.
This commit moves some macros for them to header file.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When building without CONFIG_PCI, we can (depending on the architecture)
get a link failure:
ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined!
Adding a compile-time check for PCI gets it to work correctly on
32-bit ARM.
Fixes: d99501b8575d ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We've excluded the display_power_control flag for Intel HSW and BDW
codecs as the HD-audio controllers of the corresponding platforms take
care of the display power as well. But the recent refactoring
separates the controller and the codec power accounting, so it's fine
to call the display PM even for HSW/BDW codecs. This is less
confusing since we can avoid this well-hidden condition.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The display power is in unbalance at removing the driver since it
misses the snd_hdac_display_power(OFF) call.
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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After the recent refactoring, snd_hdac_display_power() doesn't return
any error, hence it can be defined to return void.
This makes many error checks redundant and allows us to reduce them
gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When an error occurs in azx_probe_continue(), we should release the
display power. However, the current code ignores it and releases the
display power only for HSW/BDW cases. Fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_hdac_display_power() can be called even for a HDA controller
without DRM binding. The same is true for other helpers,
snd_hdac_i915_set_bclk() and snd_hdac_set_codec_wakeup().
So all superfluous AZX_DCAPS_I915_POWERWELL checks in hda_intel.c can
be dropped, and the definition of AZX_DCAPS_I915_POWERWELL itself can
be removed as well. This simplifies the code a lot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current HD-audio code manages the DRM audio power via too complex
redirections, and this seems even still unbalanced in a corner case as
Intel DRM CI has been intermittently reporting. This patch is a big
surgery for addressing the complexity and the possible unbalance.
Basically the patch changes the display PM in the following ways:
- Both HD-audio controller and codec drivers call a single helper,
snd_hdac_display_power(). (Formerly, the display power control from
a codec was done indirectly via link_power bus ops.)
- snd_hdac_display_power() receives the codec address index. For
turning on/off from the controller, pass HDA_CODEC_IDX_CONTROLLER.
- snd_hdac_display_power() doesn't manage refcounts any longer, but
keeps the power status in bitmap. If any of controller or codecs is
turned on, the function updates the DRM power state via get_power()
or put_power().
Also this refactor allows us more cleanup:
- The link_power bus ops is dropped, so there is no longer indirect
management, as mentioned in the above.
- hdac_device link_power_control flag is moved to hda_codec
display_power_control flag, as it's only for HDA legacy.
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=106525
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge for resolving the conflict of fixup entries added in both
branches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC
and output through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack
sensing and enable use of the internal microphone on this laptop
X542UN. However, it's ALC294 so create a new fixup named
ALC294_FIXUP_ASUS_MIC to avoid confusion.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Make unified suspend / resume helpers and call them from both the
runtime- and the system-PM callbacks for simplifying code.
There are slight changes of call orders, but there shouldn't be any
functional difference after refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In an initial commit, 'SYNC_STATUS' register is referred to get
clock configuration, however this is wrong, according to my local
note at hand for reverse-engineering about packet dump. It should
be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config()
is correctly programmed.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Users reported a mute LED regression on Lenovo X1 Carbon, the root
cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE
to this machine, then the machine can't apply the fixup of
ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup
together.
Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch will enable headset button for new Chrome platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Extend some structs to add the support for jack button changes.
Now snd_hda_jack_add_kctl() receives two more arguments: the jack type
and the jack keymaps. Both are optional, and when zero are passed,
the function behaves just like before.
For reporting button state changes, you'd need to update
jack->button_state bits accordingly, typically in the jack callback.
Then the value OR'ed with button_state and the jack plug state is
passed to snd_jack_report().
Note that currently the code assumes only the one-shot button events,
i.e. it tries to send the button release soon after sending the button
event. If a driver really supports the button release handling by
itself, we may need to introduce some flag to control this behavior in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For allowing the callee to evaluate the associated jack information
and the unsolicited event data, add the new fields to
hda_jack_callback. They can be used, for example, to retrieve the
headset button state in the callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge for applying the more HD-audio quirks on top of the latest
code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If it plugged headphone or headset into the jack, then
do the reboot, it will have a chance to cause headphone no sound.
It just need to run the headphone mode procedure after boot time.
The issue will be fixed.
It also suitable for ALC234 ALC274 and ALC294.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
A couple of open coded iterating thru the child node names are converted
to use for_each_child_of_node() instead.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert soundbus uevent and sysfs OF node name and device type usage to
use printf specifier and helper functions instead of directly accessing
the name and type pointers. This will allow the eventual removal of the
pointers.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues
with the input from external microphone. The issue can be fixed by
the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Acer AIO Veriton Z4660G with ALC286 codec has issue with the input
from external microphones connecting via 'Front Mic' jack. The fixup
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of
the headset and fix the audio input issue of external microphone.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer AIO Aspire C24-860 with ALC286 can't detect the headset
microphone. Just like another Acer AIO U27-880, it needs a different
pin value for 0x18 and the headset fixup to make headset mic work.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic
and internal mic not working either. It needs the similar quirk like
Sony laptops to fix headphone jack sensing and enables use of the
internal microphone.
Unfortunately jack sensing for the headset mic is still not working.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Tegra186 and Tegra194 contain the same codecs as earlier chips and can
be supported using the same patch function.
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recent devices support more than the 4 codecs that the AZX core will
probe by default. Probe up to 8 codecs to make sure all of them are
enumerated.
Suggested-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49abe5 ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We've got a regression report for some Thinkpad models (at least
T570s) which shows the too low speaker output volume. The bisection
leaded to the commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad
Dock device for ALC298 platform"), and it's basically adding the two
pin configurations for the dock, and looks harmless.
The real culprit seems, though, that the DAC assignment for the
speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14
to be coupled with DAC NID 0x03. When more pins are configured by the
commit above, the auto-parser changes the DAC assignment, and this
resulted in the regression.
As a workaround, just provide the fixed pin / DAC mapping table for
this Thinkpad fixup function. It's no generic solution, but the
problem itself is pretty much device-specific, so must be good
enough.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304
Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a series of patches for conversion to LEDs audio-mute
trigger. It's based on 4.20-rc3 to be an immutable branch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As addressed in alsa-lib (commit b420056604f0), we need to fix the
case where the evaluation of PCM interval "(x x+1]" leading to
-EINVAL. After applying rules, such an interval may be translated as
"(x x+1)".
Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It's similar to other AMD audio devices, it also supports D3, which can
save some power drain.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM
stream") fixes deadlock for non-atomic PCM stream. But, This patch
causes antother stuck.
If writer is RT thread and reader is a normal thread, the reader
thread will be difficult to get scheduled. It may not give chance to
release readlocks and writer gets stuck for a long time if they are
pinned to single cpu.
The deadlock described in the previous commit is because the linux
rwsem queues like a FIFO. So, we might need non-FIFO writelock, not
non-block one.
My suggestion is that the writer gives reader a chance to be scheduled
by using the minimum msleep() instaed of spinning without blocking by
writer. Also, The *_nonblock may be changed to *_nonfifo appropriately
to this concept.
In terms of performance, when trylock is failed, this minimum periodic
msleep will have the same performance as the tick-based
schedule()/wake_up_q().
[ Although this has a fairly high performance penalty, the relevant
code path became already rare due to the previous commit ("ALSA:
pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now
this unconditional msleep appears only when using linked streams,
and this must be a rare case. So we accept this as a quick
workaround until finding a more suitable one -- tiwai ]
Fixes: 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream")
Suggested-by: Wonmin Jung <wonmin.jung@lge.com>
Signed-off-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently the PCM core calls snd_pcm_unlink() always unconditionally
at closing a stream. However, since snd_pcm_unlink() invokes the
global rwsem down, the lock can be easily contended. More badly, when
a thread runs in a high priority RT-FIFO, it may stall at spinning.
Basically the call of snd_pcm_unlink() is required only for the linked
streams that are already rare occasion. For normal use cases, this
code path is fairly superfluous.
As an optimization (and also as a workaround for the RT problem
above in normal situations without linked streams), this patch adds a
check before calling snd_pcm_unlink() and calls it only when needed.
Reported-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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By default HDA sound card is registered with shortname "tegra-hda".
Same driver is used across tegra platforms and it is necessary to
distinguish between platforms to use platform specific settings from
userspace. One such example is, hdmi port on different platforms use
different alsa pcm device ID. For hdmi playback to work it should
open correct pcm device depending on the platform.
This patch applies shortname from first compatible string provided
in root node of device tree. Userspace then can use this card name
to apply specific settings.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we've switched to the LED trigger for binding with HD-audio,
we can drop the exported function as well as the whole
linux/thinkpad_acpi.h.
The own TPACPI_LED_MUTE and TPACPI_LED_MICMUTE definitions are
replaced with the identical ones for LEDS, i.e. LED_AUDIO_MUTE and
LED_AUDIO_MICMUTE, respectively. They are no longer needed as
referred only locally.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we've switched to the LED trigger for binding with HD-audio,
we can drop the exported function as well as the whole
linux/dell-led.h.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now all relevant platform drivers are providing the LED audio trigger,
we can switch the mute LED control with the LED trigger, finally.
For the mic-mute LED trigger, a common fixup function,
snd_hda_gen_fixup_micmute_led(), is provided to be called for the
corresponding quirk entries. This sets up the capture sync hook with
ledtrig_audio_set() call appropriately.
For the mute LED trigger, which is done currently only for
thinkpad_acpi, the call is replaced with ledtrig_audio_set() as well.
Overall, the beauty of the new implementation is that the whole ugly
bindings with request_symbol() are dropped, and also that it provides
more flexibility to users.
One potential behavior change by this patch is that the mute LED enum
may be created on machines that actually have no LED device. In the
former code, we did test-call and abort binding if the test failed.
But with the LED-trigger binding, this test isn't possible, and the
actual check is done in the LED class device side. So it's the
downside of simpleness.
Also, note that the HD-audio codec driver doesn't select CONFIG_LEDS
and co by itself. It's supposed to be selected by the platform
drivers instead.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In the upcoming change, the binding of audio mute / mic-mute LED
controls will be switched with LED trigger. This patch is the last
piece of preparation: adding the audio mute / mic-mute LED class
devices to thinkpad_acpi driver.
Two devices, platform::mute and platform::micmute, will be added for
controlling the mute LED and mic-mute LED, respectively. The new
prefix "platform" is the suggestion by upstream for indicating the
generic laptop attribute.
Also this selects CONFIG_LEDS_TRIGGERS and CONFIG_LEDS_TRIGGERS_AUDIO
unconditionally. Strictly speaking, these aren't 100% mandatory, but
leaving these manual selections would lead to a functional regression
easily once after converting from the dynamic symbol binding to the
LEDs trigger in a later patch.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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