For making the driver behavior compatible with the earlier kernels,
use the analog beep in the loopback path instead of the digital beep.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer widget (NID 0x20) of AD1884 and AD1984 codecs isn't
connected directly to the actual I/O paths but only via another mixer
widget (NID 0x21). We need a similar fix as we did for AD1882.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When plugging a headphone or headset, lots of noise is heard from
internal speaker, after changing the automute via amp instead of
pinctl, the noise disappears.
BugLink: https://bugs.launchpad.net/bugs/1268468
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Folio 13 may have a broken BIOS that doesn't set up the mute LED
GPIO properly, and the driver guesses it wrongly, too. Add a new
fixup entry for setting the GPIO pin statically for this laptop.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The front headphone and mic jackes on a HP desktop model (Vendor Id:
0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this
machine has 8 physical ports, 6 of them are routed to rear jackes
and all of them work very well, while the remaining 2 ports are
routed to front headphone and mic jackes, but the corresponding
pin complex node are not defined correctly.
After apply this fix, the front audio jackes can work very well.
[trivial fix of enum definition by tiwai]
BugLink: https://bugs.launchpad.net/bugs/1282369
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Gerald Yang <gerald.yang@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorrect ADC is picked in ca0132_capture_pcm_prepare(),
where it assumes multiple streams while there is one stream
per ADC. Note that ca0132_capture_pcm_cleanup() already does
the right thing.
The Chromebook Pixel has a microphone under the keyboard that
is attached to node id 0x8. Before this fix, recording would
always go to the main internal mic (node id 0x7).
Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Reviewed-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a HDMI stream is opened with the same stream tag
as a following opened stream to ca0132, audio will be
heard from two ports simultaneously.
Fix this issue by change to use snd_hda_codec_setup_stream
and snd_hda_codec_cleanup_stream instead, so that an
inactive stream can be marked as 'dirty' when found
with a conflict stream tag, and then get purified.
Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Reviewed-by: Chih-Chung Chang <chihchung@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machines (Vendor ID:
0x10ec0255, Subsystem ID: 0x10280657; Vendor ID: 0x10ec0255,
Subsystem ID: 0x1028065f), the headset mic can't be
detected, after apply this patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit d3c56568f4.
The reverted commit breaks audio through headphone line out on
the Acer TravelMate B113 (Type1Sku0) Notebook, my main work
machine. I don't know much about it but this fixes my problem.
Bisected and tested.
Fixes: d3c56568f4 ('ALSA: hda/realtek - Avoid invalid COEFs for ALC271X')
Cc: <stable@vger.kernel.org>
Tested-by: Martin Kepplinger <martink@posteo.de>
Signed-off-by: Martin Kepplinger <martink@posteo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after the fix for leftover kconfig handling (commit f8f1becf),
the current code still doesn't handle properly the builtin/module
mixup case between the core snd-hda-codec and other codec drivers.
For example, when CONFIG_SND_HDA_INTEL=y and
CONFIG_SND_HDA_CODEC_HDMI=m, it'll end up with an unresolved symbol
snd_hda_parse_hdmi_codec. This patch fixes the issue.
Now codec->parser points to the parser object *only* when a module
(either generic or HDMI parser) is loaded and bound. When a builtin
symbol is used, codec->parser still points to NULL. This is the
difference from the previous versions.
Fixes: f8f1becfa4 ('ALSA: hda - Fix leftover ifdef checks after modularization')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The very same fixup is needed to make the mic on Sony VAIO Pro 11
working as well as VAIO Pro 13 model.
Reported-and-tested-by: Hendrik-Jan Heins <hjheins@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code for controlling mic mute LED in patch_sigmatel.c
blindly assumes that there is a single capture switch. But, there can
be multiple multiple ones, and each of them flips the state, ended up
in an inconsistent state.
For fixing this problem, this patch adds kcontrol to be passed to the
hook function so that the callee can check which switch is being
accessed. In stac_capture_led_hook(), the state is checked as a
bitmask, and turns on the LED when all capture switches are off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit [595fe1b702: ALSA: hda - Make
CONFIG_SND_HDA_CODEC_* tristate], the kconfig variables for the
generic parser and codec drivers can be "m" instead of boolean, but
some codes are left unchanged to check only #ifdef
CONFIG_SND_HDA_CODEC_XXX, which is no longer true for modules.
This patch fixes them by replacing with IS_ENABLED() macros.
Fixes: 595fe1b702 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate')
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1983 has flexible loopback routes and the generic parser would take
wrong path confusingly instead of taking individual paths via NID 0x0c
and 0x0d. For avoiding it, limit the connections at these widgets so
that the parser can think more straightforwardly. This fixes the
regression of the missing line-in loopback on Dell machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mac Pro 1,1 with ALC889A codec needs the VREF setup on NID 0x18 to
VREF50, in order to make the speaker working. The same fixup was
already needed for MacBook Air 1,1, so we can reuse it.
Reported-by: Nicolai Beuermann <mail@nico-beuermann.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer widget on AD1983 at NID 0x0e was missing in the commit
[f2f8be43c5: ALSA: hda - Add aamix NID to AD codecs].
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've seen often problems after suspend/resume on Acer Aspire One
AO725 with ALC271X codec as reported in kernel bugzilla, and it turned
out that some COEFs doesn't work and triggers the codec communication
stall.
Since these magic COEF setups are specific to ALC269VB for some PLL
configurations, the machine works even without these manual
adjustment. So, let's simply avoid applying them for ALC271X.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Toshiba Satellite L40 with AD1986A codec requires the EAPD of NID 0x1b
to be constantly on, otherwise the output doesn't work.
Unlike most of other AD1986A machines, EAPD is correctly implemented
in HD-audio manner (that is, bit set = amp on), so we need to clear
the inv_eapd flag in the fixup, too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=67481
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 384a48d715 "ALSA: hda: HDMI: Support codecs with fewer cvts
than pins" dynamically enabled each pin widget's PIN_OUT only when the
pin was actively in use. This was required on certain NVIDIA CODECs for
correct operation. Specifically, if multiple pin widgets each had their
mux input select the same audio converter widget and each pin widget had
PIN_OUT enabled, then only one of the pin widgets would actually receive
the audio, and often not the one the user wanted!
However, this apparently broke some Intel systems, and commit
6169b67361 "ALSA: hda - Always turn on pins for HDMI/DP" reverted the
dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA
CODECs.
This change supports either dynamic or static handling of PIN_OUT,
selected by a flag set up during CODEC initialization. This flag is
enabled for all recent NVIDIA GPUs.
Reported-by: Uosis <uosisl@gmail.com>
Cc: <stable@vger.kernel.org> # v3.13
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (Vendor ID:
0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be
detected, after apply this patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Doro Wu <fan-cheng.wu@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This completes the hardware support for the Asus Xonar DG/DGX cards,
and makes them actually usable.
This is v4 of Roman's patch set with some small formatting changes.
Remove old SPI control functions, change anti-pop init
sequence, remove some garbage from structures. The 'Apply' functions
must be called at the mixer initialization, otherwise
mixer settings sometimes will not be applied at startup.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
First of all, we should not touch the GPIOs. They are not
for selecting the capture source, but they seems just enable
the whole audio input curcuit. The 'put' function calls the
'apply' functions to change register values. Change the order
of capture sources.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the input_vol_* functions to use the new SPI routines,
There is a new applying function that will be called when
the capture source changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
I tried both variants: volume control and impedance selector.
In the first case one minus is that we can't change the
volume of multichannel output without additional software
volume control. However, I am using this variant for the
last three months and this seems good. All multichannel
speaker systems have internal amplifier with the
volume control included, but not all headphones have
this regulator. In the second case, my software volume
control does not save the value after reboot.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the order of elements in the output select control. This will
reduce the number of relay switches. Change 'put' function to call the
oxygen_update_dac_routing() function. Otherwise multichannel playback
does not work. Also there is a new function to apply settings, this
prevents from duplicating the code.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Actually CS4245 connected to the I2S channel 1 for
capture, not channel 2. Otherwise capturing and
playback does not work for CS4245.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Moving the mixer code away makes things easier. The mixer
will control the driver, so the functions of the
driver need to be non-static.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the function to read the data from the new shadow buffer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio output destinations (headphones,
FP headphones, multichannel output), the channel routing
should be changed depending on what destination selected.
Also unnecessary I2S channels are digitally muted. This
function called when the user selects the destination
in the ALSA mixer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change CS4245 initialization: different sequence and GPIO values,
according to datasheets and reverse-engineering information.
Change cleanup/resume/suspend functions, since they use
initialization.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add the new SPI write and read functions. The SPI read function
is used for creating initial registers dump and may be used for
debugging purposes. SPI operations are cached, so there is a new
function to manage the cache (shadow). I have to remove
the shift from the CS4245_SPI_* constants, since when
we are performing the reading, we need to shift by 8 instead
of 16.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add additional constants to the xonar_dg.h file:
capture and playback sources. Move GPIO_* constants and the
dg struct to the header file from the xonar_dg.c file.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add some additional information in comments and my copyright.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When the user switches the output from stereo to multichannel
or vice versa, the driver needs to update the channel routing.
Instead of creating additional subroutines, I better export existing
oxygen_update_dac_routing symbol from the oxygen mixer
and call this function. It calls model.adjust_dac_routing()
and my function does the work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the oxygen_write_spi() function to use the newly
introduced oxygen_wait_spi() function. Change return value
from void to int, so it can return error codes. Older
drivers just ignore that return value, new drivers can
check this value. We need to wait AFTER
initiating the SPI transaction, otherwise read
operation will not work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The oxygen_wait_spi() function now performs waiting when the
SPI bus completes a transaction. Introduce the timeout error
checking and increase timeout to 200 from 40.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Processing coefficients are often a vital part of the codec's configuration,
so dumping them can be important. However, because they are undocumented and
secret, we do not want to enable this for all codecs by default.
Therefore instead add this as a debugging parameter.
I have prepared for codecs that want to enable this by default by the extra
dump_coef bitfield, but unsure if we want to do that as long as the
(unlikely, but still) race remains.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly to other Apple products, MBA 1,1 needs a specific quirk.
Pin 0x18 must be set to VREF_50 to have sound output. This was no
longer done since commit 1a97b7f, resulting in a mute built-in speaker.
This patch corrects the regression by creating a fixup for the MBA 1,1.
Fixes: 1a97b7f227 ("ALSA: hda/realtek - Remove the last static quirks for ALC882")
Cc: <stable@vger.kernel.org> [v3.4+]
Tested-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Zenbook UX31A has yet another problem -- softer output level than
others. According to the measurement, the peak output difference
between 31A and 31E is 5dB. As ALC269VB has a COEF for the class-D
pre-amp, let's apply it for +5dB.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on some Dell machines, the headset
mic can't be detected, after apply this patch, the headset mic
can work well on all those machines.
On the machine with the Subsytem ID 0x10280610, if we use
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be
detected and work well, but the sound can't be outputed via
headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE
can fix this problem.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: David Chen <david.chen@canonical.com>
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Tested-by: Shawn Wang <shawn.wang@canonical.com>
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new vmaster hook, update_tpacpi_mute_led(), calls the original
vmaster hook, but I forgot to save the original hook function but keep
calling the updated one, which of course results in a stupid endless
loop. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
