The last user of snd_hda_gen_spec_free() is patch_via.c, and we can
rewrite it safely with snd_hda_gen_free(), so that
snd_hda_gen_spec_free() can be a local function in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to alsa-info.sh outputs, all three entries with static
quirks have the correct pin configs, so it's safe to remove static
quirks. For now, turn the static quirks off via ifdef. The dead
codes will be removed in later release.
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>
Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the
new IS_ENABLED() macro.
The patch still doesn't cover all ifdefs. For example, the dependency
on CONFIG_GAMEPORT is still open-coded because this also has an extra
dependency on MODULE. Similarly, an open-coded ifdef in pcm_oss.c and
some sequencer-related stuff are left untouched.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused function in pci/lx6464es/lx_core.c.
This eliminates the following warning in pci/lx6464es/lx_core.c:
sound/pci/lx6464es/lx_core.c:144:5: warning: no previous prototype for ‘lx_plx_mbox_read’ [-Wmissing-prototypes]
sound/pci/lx6464es/lx_core.c:172:5: warning: no previous prototype for ‘lx_plx_mbox_write’ [-Wmissing-prototypes]
sound/pci/lx6464es/lx_core.c:494:5: warning: no previous prototype for ‘lx_dsp_es_check_pipeline’ [-Wmissing-prototypes]
Signed-off-by: Rashika Kheria <rashika.kheria@gmail.com>
Reviewed-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bug reporter report that the -mode4 makes the subwoofer work.
I have simplified the quirk a bit to avoid possible regressions
with the microphones.
BugLink: https://bugs.launchpad.net/bugs/871808
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just a small refactoring to make the next patch slightly simpler.
Signed-off-by: David Henningsson <david.henningsson@canonical.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>
Lenovo Ideapad with ALC272 has a mute LED that is controlled via
GPIO1. Add a simple vmaster hook for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373
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>
While looking into some spurious responses, I found that the addr value was
treated a bit inconsistent: values 8..0xf will be treated as codec 0 and
values 0..7 will be treated as no error regardless of whether there is a codec
there, or not.
With this patch, all non-existing codecs will be treated equally.
In addition, printing rp and wp could help figuring out if the wp value is
reported wrongly from the controller or if something else is wrong.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all weird setups have been converted to fixups for the generic
parser, and we can disable the static quirks. This commit just turns
the build off. The bulky static quirk code still remains for a while,
in case we get an overlooked regression. It'll be removed at the next
kernel version.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both CX20549 and CX20551 codecs have a mixer widget and it can be
connected as the ADC source. Like AD and VIA codecs, enable the
add_stereo_mix_input flag for these codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OLPC XO needs a few special handling. Now these are implemented as a
fixup to the generic parser.
Obviously, the DC BIAS mode had to be added manually. This is mainly
implemented in the mic_autoswitch hook, where the mic pins are
overwritten depending on the DC bias mode. This also required the
override of the mic boost control, since the mic boost is applied only
when the DC mode is disabled.
In addition, the mic pins must be set dynamically at recording time
because these also control the LED.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... by using snd_Hda_codec_update_cache() instead of *_write_cache().
Since all path elements should have been updated by this function,
we are safe to assume that the cache contents are consistent.
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>
