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: NTakashi Iwai <tiwai@suse.de>

Since the commit [595fe1b7: 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: 595fe1b7 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate')
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161Signed-off-by: NTakashi 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: NTakashi 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: NNicolai Beuermann <mail@nico-beuermann.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The mixer widget on AD1983 at NID 0x0e was missing in the commit
[f2f8be43: 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: NTakashi 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: NTakashi 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: NTakashi Iwai <tiwai@suse.de>

Commit 384a48d7 "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
6169b673 "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: NUosis <uosisl@gmail.com>
Cc: <stable@vger.kernel.org> # v3.13
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NTakashi 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: NDoro Wu <fan-cheng.wu@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change the function to read the data from the new shadow buffer.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Add some additional information in comments and my copyright.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens 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: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi 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 1a97b7f2, resulting in a mute built-in speaker.

This patch corrects the regression by creating a fixup for the MBA 1,1.

Fixes: 1a97b7f2 ("ALSA: hda/realtek - Remove the last static quirks for ALC882")
Cc: <stable@vger.kernel.org> [v3.4+]
Tested-by: NAdrien Vergé <adrienverge@gmail.com>
Signed-off-by: NAdrien Vergé <adrienverge@gmail.com>
Signed-off-by: NTakashi 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: NTakashi Iwai <tiwai@suse.de>

The release of module object itself was forgotten.
Spotted by COVERIY CID 1162828.
Signed-off-by: NTakashi 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: NDavid Chen <david.chen@canonical.com>
Tested-by: NCyrus Lien <cyrus.lien@canonical.com>
Tested-by: NShawn Wang <shawn.wang@canonical.com>
Tested-by: NChih-Hsyuan Ho <chih.ho@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi 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: NTakashi Iwai <tiwai@suse.de>

On some AIO (All In One) models with the codec alc668
(Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack,
the system will switch the output to headphone and set the speaker to
automute as well as change the speaker Pin-ctls from 0x40 to 0x00,
this will bring loud noise to the headphone.

I tried to disable the corresponding EAPD, but it did not help to
eliminate the noise.

According to Takashi's suggestion, we use amp operation to replace the
pinctl modification for the automute, this really eliminate the noise.

BugLink: https://bugs.launchpad.net/bugs/1268468
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Apply the codec->power_filter to the FG nodes in general for reducing
hackish set_power_state ops override in patch_sigmatel.c.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some old AD codecs don't like the independent HP handling, either it
contains a single DAC (AD1981) or it mandates the mixer routing
(AD1986A).  This patch removes the indep_hp flag for such codecs.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The PCI devices with DMA masks smaller than 32bit should enable
CONFIG_ZONE_DMA.  Since the recent change of page allocator, page
allocations via dma_alloc_coherent() with the limited DMA mask bits
may fail more frequently, ended up with no available buffers, when
CONFIG_ZONE_DMA isn't enabled.  With CONFIG_ZONE_DMA, the system has
much more chance to obtain such pages.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Occasionally, on playback stream ringbuffer wraparound, the EMU20K1
hardware will momentarily return 0 instead of the proper current(loop)
address. This patch handles that case, fixing the problem of playback
position corruption and subsequent loss of buffered sound data, that
occurs with some common buffering layout patterns(e.g. multiple
simultaneous output streams with differently-sized or
non-power-of-2-sized buffers).

An alternate means of fixing the problem would be to read the ca
register continuously, until two sequential reads return the same
value; however, that would be a more invasive change, has performance
implications, and isn't necessary unless there are also issues with the
value not being updated atomically in regards to individual bits or
something similar(which I have not encountered through light testing).

I have no EMU20K2 hardware to confirm if the issue is present there,
but even if it's not, this change shouldn't break anything that's not
already broken.
Signed-off-by: NSarah Bessmer <aotos@fastmail.fm>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Nowadays we have CMA for obtaining the contiguous memory pages
efficiently.  Let's kill the old kludge for reserving the memory pages
for large buffers.  It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>