The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect
the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk
applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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>
The Acer Aspire Z24-890 cannot detect the headset MIC until
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed96 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G")
Fixes: b72f936f6b ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3baffc4a84 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another machine which does not like the power saving (noise):
https://bugzilla.redhat.com/show_bug.cgi?id=1689623
Also, reorder the Lenovo C50 entry to keep the table sorted.
Reported-by: hs.guimaraes@outlook.com
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case ioremap_nocache fails, the fix releases chip and returns
an error code upstream to avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook.
There are no other users for ALC225_FIXUP_HEADSET_JACK other than
the chromebook hardware.
Fixes: 10f5b1b85e ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a port of the ASoC Icelake HDMI codec code to the legacy
HDA driver with some cleanups.
ASoC commit 019033c854:
"ASoC: Intel: hdac_hdmi: add Icelake support"
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Bard liao <bard.liao@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and
HP Z240 models.
Reported-by: Jeff Burrell <jeff.burrell@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Implement of reset combo jack JD. It will show normally.
Fixes: e854747d75 ("ALSA: hda/realtek - Enable headset button support for new codec")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC
until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the
internal DMIC uses another module - snd_soc_skl as the driver. We still
need the NID 0x1a in the quirk to enable the headset MIC.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The #ifdef protection around the PM functions is wrong, leading to
a failed reference in some configurations:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend':
sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration]
Better remove the #ifdefs entirely and rely on the compiler silently
dropping unused functions marked __maybe_unused.
Fixes: 707e0759f2 ("ALSA: hda/tegra: implement runtime suspend/resume")
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
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Merge tag 'asoc-v5.1-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More changes for v5.1
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
A platform can have multiple sound cards for different audio paths.
Following is the print seen duirng device boot for jetson-xavier,
ALSA device list:
#0: nvidia,p2972-0000 at 0x3518000 irq 17
By looking at above, it is not very clear if the sound card is for
HDA. It becomes confusing when platform has registered multiple cards,
and platform model name is used for card.
This patch uses "nvidia,model" property mentioned in hda device tree
to get the card name. Since property is optional, legacy boards will
continue to use "tegra-hda". Custom name can be passed wherever needed.
This naming convention is conistent with the way sound cards are named
in general.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is reported that there's a constant background "hum/whitenoise"
in the headset on the Lenovo X1 machines with the codec alc285, and it
is confirmed that if we run the command below, the noise will stop.
sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0
Then I consulted this issue with Kailang, he told me the pin 0x1d on
this codec is used for PC beep in, the noise probably comes from this
pin and we can also disable the PC beep in passthrough, then the PC
beep in will not affect other sound playback.
Fixes: c4cfcf6f42 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the System76 Oryx Pro (oryp5), there is a headset microphone input
attached to 0x19 that does not have a jack detect. In order to get it
working, the pin configuration needs to be set correctly, and the
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is
similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we
have a separate microphone jack that is already configured correctly.
Since the ALC1220 does not have a fixup similar to
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the
ALC269 in a way that it can be accessed from the
alc1220_fixup_system76_oryp5 function. In addition, the
alc1220_fixup_clevo_p950 needs to be applied to gain speaker output.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables mute LED support and fixes switching jacks when the laptop
is docked.
Signed-off-by: Jurica Vukadin <jurica.vukadin@rt-rk.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Again no functional changes, but only code clean up.
Use a standard macro for initializing the procfs entries, also drop
the info entries stored in dsp_spos_instance, as they are removed
recursively by a single snd_info_free_entry() calls.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls, as well as dropping the superfluous setup of
SNDRV_INFO_CONTENT_TEXT.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A call of pci_iounmap() call without CONFIG_PCI leads to a build error
on some architectures. We tried to address this and add a check of
IS_ENABLED(CONFIG_PCI), but this still doesn't seem enough for sh.
Ideally we should fix it globally, it's really a corner case, so let's
paper over it with a simpler ifdef.
Fixes: 1e73359a24 ("ALSA: hda/ca0132 - make pci_iounmap() call conditional")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, the codec registration may happen both at the
codec bind time and the end of the controller probe time. In a rare
occasion, they race with each other, leading to Oops due to the still
uninitialized card device.
This patch introduces a simple flag to prevent the codec registration
at the codec bind time as long as the controller probe is going on.
The controller probe invokes snd_card_register() that does the whole
registration task, and we don't need to register each piece
beforehand.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the open-codes in many places with a new common helper for
performing the same thing: referring to the primary headphone pin.
This eventually fixes the potentially missing headphone pin on some
weird devices, too.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When auto_mute = no or spec->suppress_auto_mute = 1, cfg->hp_pins will
lose value.
Add this patch to find hp_pins value.
I add fixed for ALC282 ALC225 ALC256 ALC294 and alc_default_init()
alc_default_shutup().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the System76 Darter Pro (darp5), there is a headset microphone
input attached to 0x1a that does not have a jack detect. In order to
get it working, the pin configuration needs to be set correctly, and
the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied.
This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops,
except we have a separate microphone jack that is already configured
correctly.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the registration and free of beep input device was done
manually from the register and the disconnect callbacks of the
assigned codec object. This seems working in most cases, but this may
be a cause of some races at probe. Moreover, due to these manual
calls, the total code became unnecessarily lengthy.
This patch rewrites the beep registration code to follow the standard
sound device object style. This allows us reducing the code, in
addition to avoiding the nested device registration calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The init sequence for ALC294 headphone stuff is needed not only for
the boot up time but also for the resume from hibernation, where the
device is switched from the boot kernel without sound driver to the
suspended image. Since we record the PM event in the device
power_state field, we can now recognize the call pattern and apply the
sequence conditionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix hp_pin always no value.
[More notes on the changes:
The hp_pin value that is referred in alc294_hp_init() is always zero
at the moment the function gets called, hence this is actually
useless as in the current code.
And, this kind of init sequence should be called from the codec init
callback, instead of the parser function. So, the first fix in this
patch to move the call call into its own init_hook.
OTOH, this function is needed to be called only once after the boot,
and it'd take too long for invoking at each resume (where the init
callback gets called). So we add a new flag and invoke this only
once as an additional fix.
The one case is still not covered, though: S4 resume. But this
change itself won't lead to any regression in that regard, so we
leave S4 issue as is for now and fix it later. -- tiwai ]
Fixes: bde1a74596 ("ALSA: hda/realtek - Fixed headphone issue for ALC700")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AZX_DCAPS_PM_RUNTIME flag is added to indicate support for runtime PM.
azx_has_pm_runtime() is used to check if above is enabled and thus
forbid runtime PM calls if needed.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves clock enable/disable from system resume/suspend to
runtime resume/suspend respectively. Along with this hda controller
chip init or stop is also moved. System resume/suspend can invoke
runtime callbacks and do necessary setup.
chip->running can be used to check for probe completion and device
access during runtime_resume or runtime_suspend can be avoided if
probe is not yet finished. This helps to avoid kernel panic during
boot where runtime PM callbacks can happen from system PM.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explicit clock enable is not required during probe, as this would be
managed by runtime PM calls. Clock can be enabled/disabled in runtime
resume/suspend. This way it is easier to balance clock enable/disable
counts.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved devm_clk_get() API calls to a separate function and the same
can be called early in the probe. This is done before runtime PM
for the device is enabled. The runtime resume/suspend callbacks can
later enable/disable clocks respectively(the support would be added
in subsequent patches). Clock handles should be available by the
time runtime suspend/resume calls can happen.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables runtime power management(runtime PM) support for
hda. pm_runtime_enable() and pm_runtime_disable() are added during
device probe and remove respectively. The runtime PM callbacks will
be forbidden if hda controller does not have support for runtime PM.
pm_runtime_get_sync() and pm_runtime_put() are added for hda register
access. The callbacks for above will be added in subsequent patches.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support speaker and mic mute LEDs on HP ProBook 470 G5.
BugLink: https://bugs.launchpad.net/bugs/1811254
Signed-off-by: Anthony Wong <anthony.wong@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unused variable was forgotten to be removed and now we get a
compiler warning:
sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend':
sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm'
Fixes: 17bc4815de ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATIIXP driver supports the full PCM resume and saves/restores the
running PCM pointer. This used to be done in the suspend and resume
callbacks together with snd_pcm_suspend() call. But since we moved
the snd_pcm_supsend*() call in PCM device PM ops, this should be moved
to a more appropriate place, i.e. the trigger callback.
Along with the movement of the PCM suspend/resume code, remove the
superfluous snd_pcm_suspend_all() call, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple
sdo lines if supported. Audio needs to be striped across number of sdo
lines for simultaneous playbacks of higher resolutions to work.
This needs to be implemented only for an Audio Output Converter and only
if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter
is 1.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix typo for model alc255-dell1 to alc225-dell1.
Enable headset mode support for new WYSE NB platform.
Fixes: a26d96c780 ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable Headset Mic VREF for headset mode of ALC225.
This will be controlled by coef bits of headset mode functions.
[ Fixed a compile warning and code simplification -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to add unplug function to unplug state of headset mode
for ALC225.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "chip->dsp_spos_instance" can be NULL on some of the ealier error
paths in snd_cs46xx_create().
Reported-by: "Yavuz, Tuba" <tuba@ece.ufl.edu>
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_i2c_sendbytes could fail. The fix checks its return value: if it
fails, issues an error message and returns with its error code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no reason for us to do that while we initialize dac_mute to
1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE
bit anyway.
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell has new platform for ALC274.
This will support to enable headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This essentially reverts the commits
c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present
and Skylake driver selected")
and
d82b51c855 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+
driver selection")
for the path of legacy HD-audio controller (snd-hda-intel).
The automatic DSP detection and skip of binding with the legacy driver
caused regressions on several machines like Dell XPS13. They give the
PCI class 0x40380 indicating the availability of DSP while they don't
work with ASoC SKL driver (yet).
As the support of ASoC driver for such devices isn't available, it's
better to revert the whole DSP-detection-and-skip behavior of the
legacy driver, so that we can get the old good driver working on such
devices.
The pci_binding option for ASoC SKL driver is still kept so that it
can work without blacklisting.
Fixes: c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Reported-by: Hans de Goede <hdegoede@redhat.com>
Reported-by: Azat Khuzhin <dohardgopro@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after disabling interrupts on the module, it could be possible
that irq handlers are still running. System hang is seen during
suspend path. It was found that, there were pending writes on the
HDA bus and clock was disabled by that time.
Above mentioned issue is fixed by clearing any pending irq handlers
before disabling clocks and returning from hda suspend.
Suggested-by: Mohan Kumar <mkumard@nvidia.com>
Suggested-by: Dara Ramesh <dramesh@nvidia.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose
CODEC is Realtek ALC294 has jack auto detection feature. This patch
enables the feature.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
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>
For HDaudio and Skylake drivers, add module parameter "pci_binding"
When pci_binding == 0 (AUTO), the PCI class/subclass info is used to
select drivers based on the presence of the DSP.
pci_binding == 1 (LEGACY) forces the use of the HDAudio legacy driver,
even if the DSP is present.
pci_binding == 2 (ASOC) forces the use of the ASOC driver. The
information on the DSP presence is bypassed.
The value for the module parameter needs to be identical for both
drivers. This parameter is intended as a back-up solution if the
automatic detection fails or when the DSP usage fails. Such cases
should be reported on the alsa-devel mailing list for analysis.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the SST/Skylake driver supports per platform selectors, we
can add logic to automatically select the right driver.
If the Skylake driver is selected for a specific platform, and the DSP
is detected at run-time based on the PCI class/subclass/prog-if
information, the legacy HDaudio driver aborts the probe. This will
result in a single driver probing and remove the need for modprobe
blacklists.
Follow-up patches will add a module parameter to bypass the logic if
this automatic detection fails, or if the Skylake driver is unable to
actually support the platform (firmware authentication, missing
topology file, hardware issue, etc).
The same mechanism will be used to conflicts generated by the same PCI
ID being registered by both legacy HDAuudio and SOF drivers for Intel
platforms. In other words SOF will not require changes to the HDaudio
legacy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default, there is no sound on Asus UX391UA on Linux.
This patch adds sound support on Asus UX391UA. Tested working by three
different users.
The problem has also been described at
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485
Signed-off-by: Wandrille RONCE <w@ndrille.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ipcm->substream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm
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
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->channel is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap)
Fix this by sanitizing info->channel before using it to index hdsp->channel_map
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].
Also, notice that I refactored the code a bit in order to get rid of the
following checkpatch warning:
ERROR: do not use assignment in if condition
FILE: sound/pci/rme9652/hdsp.c:4103:
if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested with 4.19.9.
v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because
that's what the existing fixups for EliteBooks use.
Signed-off-by: Mantas Mikulėnas <grawity@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of Huawei laptops come with a LED in the micmute key. This patch
enables the use of micmute LED for these devices:
1. Matebook X (19e5:3200), (19e5:3201)
2. Matebook X Pro (19e5:3204)
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch solves bug 200501 'Only 2 of 4 speakers playing sound.'
It enables the front speakers on Huawei Matebook X Pro laptops.
These laptops come with Dolby Atmos sound system and these pins
configuration enables the front speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200501
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: d99501b857 ("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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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: c4cfcf6f42 ("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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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 61fcf8ece9 ("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: 61fcf8ece9 ("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>
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>
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>
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>
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>
This patch will enable ALC300.
[ It's almost equivalent with other ALC269-compatible ones, and
apparently has no loopback mixer -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device makes a loud buzzing sound when a headphone is inserted while
playing audio at full volume through the speaker.
Fixes: bbf8ff6b1d ("ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers")
Signed-off-by: Girija Kumar Kasinadhuni <gkumar@neverware.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have several Lenovo laptops with the codec alc285, when playing
sound via headphone, we can hear click/pop noise in the headphone,
if we let the headphone share the DAC of NID 0x2 with the speaker,
the noise disappears.
The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon.
I have tried to set preferred_dacs and override_conn, but neither of
them worked. Thanks for Kailang, he told me to invalidate the NID 0x3
through override_wcaps.
BugLink: https://bugs.launchpad.net/bugs/1805079
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the
MS-B120, in order for the headset mic to be properly detected.
They both use a single 3-way jack for both mic and headset with an
ALC283 codec, with the same pins used.
Cc: stable@vger.kernel.org
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on the built-in audio
of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to
the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Cc: <stable@vger.kernel.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_ac97_put_spsa() gets the bit shift value from the
associated private_value, but it extracts too much; the current code
extracts 8 bit values in bits 8-15, but this is a combination of two
nibbles (bits 8-11 and bits 12-15) for left and right shifts.
Due to the incorrect bits extraction, the actual shift may go beyond
the 32bit value, as spotted recently by UBSAN check:
UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7
shift exponent 68 is too large for 32-bit type 'int'
This patch fixes the shift value extraction by masking the properly
with 0x0f instead of 0xff.
Reported-and-tested-by: Meelis Roos <mroos@linux.ee>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the pincfg assignment for the AE-5, which was
previously using the Recon3D pincfg's by mistake.
Fixes: d06feaf02f ("ALSA: hda/ca0132 - Add pincfg for AE-5")
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new PCI subsys ID for the ZxR, as found and tested by
other users. Without a way to know if any Z's use it as well, it keeps
the quirk of QUIRK_SBZ and goes through the HDA subsys test function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct declaration is not indented correctly. Fix this by replacing
spaces with a tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to call pci_iounmap() instead of iounmap() for the regions
obtained via pci_iomap() call for some archs that need special
treatment.
Fixes: aa31704fd8 ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for
the mute LED control over mic3 pin. Added the corresponding quirk
entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All the recent support of Creative boards and onboard audio depend on
PCI, but they can't be trimmed easily even if you build without
CONFIG_PCI, since the quirk is detected dynamically and the code has
many branches with the flag check like spec->quirk type or
spec->use_alt_functions.
This patch makes these checks static for CONFIG_PCI=n case so that the
compiler optimizes out. The access to flags are replaced with macros
that are replaced with a static value for CONFIG_PCI=n.
The macros look slightly ugly for avoiding compiler warnings wrt
unused variables, and some additional default-case handlings for
another compiler warnings, but the rest are very straightforward
changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit c647f806b8 ("ALSA: hda - Allow multiple ADCs for
mic mute LED controls") we allow enabling the mic mute LED with
multiple ADCs. The commit changed the function return value to be
zero or a negative error, while this change was overlooked in the
thinkpad_acpi helper code where it still expects a positive return
value for success. This eventually leads to a NULL dereference on a
system that has only a mic mute LED.
This patch corrects the return value check in the corresponding code
as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621
Fixes: c647f806b8 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Creative Audigy SE (SB0570) card currently exhibits an audible pop
whenever playback is stopped or resumed, or during silent periods of an
audio stream. Initialise the IZD bit to the 0 to eliminate these pops.
The Infinite Zero Detection (IZD) feature on the DAC causes the output
to be shunted to Vcap after 2048 samples of silence. This discharges the
AC coupling capacitor through the output and causes the aforementioned
pop/click noise.
The behaviour of the IZD bit is described on page 15 of the WM8768GEDS
datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples
will cause all outputs to be connected directly to VCAP. This also
happens if 2048 consecutive zero input samples are applied to all 6
channels, and IZD=0. It will be removed as soon as any channel receives
a non-zero input". I believe the second sentence might be referring to
IZD=1 instead of IZD=0 given the observed behaviour of the card.
This change should make the DAC initialisation consistent with
Creative's Windows driver, as this popping persists when initialising
the card in Linux and soft rebooting into Windows, but is not present on
a cold boot to Windows.
Signed-off-by: Alex Stanoev <alex@astanoev.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:
- More component transitions from Morimoto-san, I think we're about
finished with this. Thanks for all the hard work!
- Morimoto-san also added a bunch of for_each_foo macros
- A bunch of cleanups and fixes for DAPM from Charles.
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373.
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060.
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Merge tag 'asoc-v5.0' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.0/v4.20
As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:
- More component transitions from Morimoto-san, I think we're about
finished with this. Thanks for all the hard work!
- Morimoto-san also added a bunch of for_each_foo macros
- A bunch of cleanups and fixes for DAPM from Charles.
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373.
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060.
This patch fixes the microphone issue for all cards. The previous fix
worked on the ZxR, but not on the AE-5 or Z. This patch has been tested
to work for all cards.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on Dell Precision T3600
laptops and Intel DZ77BH boards, add these to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA API does its own zone decisions based on the coherent_dma_mask.
[ Note: as the driver doesn't set the DMA coherent mask, we can assume
the default 32bit DMA, hence it should be safe to drop the flag here
-- tiwai ]
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in au88xx driver.
Although the usages in both both functions vortex_adbdma_setbuffers()
and vortex_wtdma_setbuffers() look a bit suspicious, we keep the
behavior as before, just to be safer. If it were broken, we should
have already received bug reports.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo G50-30, like other G50 models, has a Conexant codec that
requires a quirk for its inverted stereo dmic.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1249364
Reported-by: Alexander Ploumistos <alex.ploumistos@gmail.com>
Tested-by: Alexander Ploumistos <alex.ploumistos@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The front MIC on the Lenovo M715 can't record sound, after applying
the ALC294_FIXUP_LENOVO_MIC_LOCATION, the problem is fixed. So add
the pin configuration of this machine to the pin quirk table.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on ASUS G751 doesn't seem to map the headphone pin (NID 0x16)
correctly. Add a quirk to address it, as well as chaining to the
previous fix for the microphone.
Reported-by: Håvard <hovardslill@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the echo cancellation control for desktop cards, and
makes use of the special 0x47 SCP command for noise reduction.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch cleans up the patch_ca0132() function with suggestions from
Takashi Sakamoto.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes microphone inconsistency issues by adding a delay to
each setup_defaults function. Without this, the microphone only works
intermittently.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a corresponding model list entry for ASUS G751 so that user can
test the quirk for another compatible machines more easily.
Reported-and-tested-by: Håvard <hovardslill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS G751 requires the extra COEF initialization to make it microphone
working properly.
Reported-and-tested-by: Håvard <hovardslill@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Elo VuPoint 15MX has two headphone jacks of which neither work by
default. Disabling automute allows ALSA to work normally with the
speakers & left headphone jack.
Future pin configuration changes may be required in the future to get
the right headphone jack working in tandem.
Signed-off-by: Michael Pobega <mpobega@neverware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The issue is the same as commit dd9aa335c8 ("ALSA: hda/realtek - Can't
adjust speaker's volume on a Dell AIO"), the output requires to connect
to a node with Amp-out capability.
Applying the same fixup ALC298_FIXUP_SPK_VOLUME can fix the issue.
BugLink: https://bugs.launchpad.net/bugs/1775068
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a control for 600 ohm gain on the Sound Blaster ZxR.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the input select control for the ZxR, as it only has
one input option, rear microphone.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds commands for selecting input and output on the Sound
Blaster ZxR. The ZxR has no front panel header, and has line-in on the
separate daughter board, so it only does rear-mic.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds commands for setting up the ZxR after the DSP is
downloaded. The ZxR already shares most of the post-download commands
from the regular Sound Blaster Z.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds separate hda_codec_ops for the DBPro daughter board, as
it behaves more like a generic HDA codec than the other ca0132 cards,
despite having a ca0132 on board.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a pincfg for the ZxR, and defines which pins are used
for both.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk ID's for the ZxR and it's daughter board, the
DBPro. It also adds a function for determining the quirk for each board
through HDA subsytem ID's instead of PCI subsystem ID's.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes an issue where if surround sound was the selected
output and output effects were enabled, the sound wasn't sent to all
channels correctly.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch corrects the control type of the additional AE-5 controls
added in a previous patch from HDA_INPUT to HDA_OUTPUT.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds controls for the AE-5's headphone gain setting, and the
DAC's interpolation filter setting.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the input selection commands for the Sound BlasterX
AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
