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>
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>
The commit 44dcbbb1cd introduced the usage of bitreverse helpers but
forgot to add the dependency. This patch adds the selection for
CONFIG_BITREVERSE.
Fixes: 44dcbbb1cd ('ALSA: snd-usb: add support for bit-reversed byte formats')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull sound fixes from Takashi Iwai:
"The big chunks here are the updates for oxygen driver for Xonar DG
devices, which were slipped from the previous pull request. They are
device-specific and thus not too dangerous.
Other than that, all patches are small bug fixes, mainly for Samsung
build fixes, a few HD-audio enhancements, and other misc ASoC fixes.
(And this time ASoC merge is less than Octopus, lucky seven :)"
* tag 'sound-fix-3.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (42 commits)
ALSA: hda/hdmi - allow PIN_OUT to be dynamically enabled
ALSA: hda - add headset mic detect quirks for another Dell laptop
ALSA: oxygen: Xonar DG(X): cleanup and minor changes
ALSA: oxygen: Xonar DG(X): modify high-pass filter control
ALSA: oxygen: Xonar DG(X): modify input select functions
ALSA: oxygen: Xonar DG(X): modify capture volume functions
ALSA: oxygen: Xonar DG(X): use headphone volume control
ALSA: oxygen: Xonar DG(X): modify playback output select
ALSA: oxygen: Xonar DG(X): capture from I2S channel 1, not 2
ALSA: oxygen: Xonar DG(X): move the mixer code into another file
ALSA: oxygen: modify CS4245 register dumping function
ALSA: oxygen: modify adjust_dg_dac_routing function
ALSA: oxygen: Xonar DG(X): modify DAC/ADC parameters function
ALSA: oxygen: Xonar DG(X): modify initialization functions
ALSA: oxygen: Xonar DG(X): add new CS4245 SPI functions
ALSA: oxygen: additional definitions for the Xonar DG/DGX card
ALSA: oxygen: change description of the xonar_dg.c file
ALSA: oxygen: export oxygen_update_dac_routing symbol
ALSA: oxygen: add mute mask for the OXYGEN_PLAY_ROUTING register
ALSA: oxygen: modify the SPI writing function
...
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>
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.
Pull slave-dma updates from Vinod Koul:
- new driver for BCM2835 used in R-pi
- new driver for MOXA ART
- dma_get_any_slave_channel API for DT based systems
- minor fixes and updates spread acrooss driver
[ The fsl-ssi dual fifo mode support addition clashed badly with the
other changes to fsl-ssi that came in through the sound merge. I did
a very rough cut at fixing up the conflict, but Nicolin Chen (author
of both sides) will need to verify and check things ]
* 'for-linus' of git://git.infradead.org/users/vkoul/slave-dma: (36 commits)
dmaengine: mmp_pdma: fix mismerge
dma: pl08x: Export pl08x_filter_id
acpi-dma: align documentation with kernel-doc format
dma: fix vchan_cookie_complete() debug print
DMA: dmatest: extend the "device" module parameter to 32 characters
drivers/dma: fix error return code
dma: omap: Set debug level to debugging messages
dmaengine: fix kernel-doc style typos for few comments
dma: tegra: add support for Tegra148/124
dma: dw: use %pad instead of casting dma_addr_t
dma: dw: join split up messages
dma: dw: fix style of multiline comment
dmaengine: k3dma: fix sparse warnings
dma: pl330: Use dma_get_slave_channel() in the of xlate callback
dma: pl330: Differentiate between submitted and issued descriptors
dmaengine: sirf: Add device_slave_caps interface
DMA: Freescale: change BWC from 256 bytes to 1024 bytes
dmaengine: Add MOXA ART DMA engine driver
dmaengine: Add DMA_PRIVATE to BCM2835 driver
dma: imx-sdma: Assign a default script number for ROM firmware cases
...
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>
Martin Kepplinger
Takashi Iwai
David Henningsson
Linus Torvalds
Stephen Warren
Hui Wang
Roman Volkov