| Commit message (Collapse) | Author | Age | Files | Lines |
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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>
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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>
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The mixer widget on AD1983 at NID 0x0e was missing in the commit
[f2f8be43c5c9: 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>
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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>
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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>
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The commit 44dcbbb1cd61 introduced the usage of bitreverse helpers but
forgot to add the dependency. This patch adds the selection for
CONFIG_BITREVERSE.
Fixes: 44dcbbb1cd61 ('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>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
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
...
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Commit 384a48d71520 "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
6169b673618b "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>
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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>
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for-next
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.
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.14
A few fixes, all in drivers. Nothing stands out particularly, the
biggest set of fixes is some build coverage issues from Sachin.
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'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus
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Export the symbol so that it is accessible to modules. Fixes the
following error:
ERROR: "wm5100_detect" [sound/soc/samsung/snd-soc-lowland.ko] undefined!
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Currently the Negative Terminal Input Routing Configuration is only set
when there is a special routing configuration. If we don't use one of
the inputs IN1 or IN2 as negative terminal input, the PGA and recording
does not work.
This patch adds a route from CM1L/CM1R to the PGA as negative input by
default. With this configuration the PGA can amplify all input signals
and line-in/mic works again.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Playback of a mono stream should output the same stream on both
channels. At the moment only the left analog signal is valid, the right
one is just noise.
This patch maps the left digital channel onto both DACs when receiving a
mono stream.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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mach/dma.h is not referenced by this file. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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'res' could be NULL from one of the operations above (line 1243). Thus
check 'res' for NULL before releasing the region to avoid null pointer
dereference.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Depend on MFD_ARIZONA to avoid the following build errors:
sound/soc/codecs/arizona.c:218: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:226: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:1719: undefined reference to `arizona_request_irq'
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Select S3C24XX_DMA instead of S3C2410_DMA to avoid following dependency issues
and build errors:
warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))
warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))
arch/arm/mach-s3c24xx/built-in.o: In function `s3c2410_dma_add':
arch/arm/mach-s3c24xx/dma-s3c2410.c:134: undefined reference to `s3c2410_dma_init'
arch/arm/mach-s3c24xx/dma-s3c2410.c:135: undefined reference to `s3c24xx_dma_order_set'
arch/arm/mach-s3c24xx/dma-s3c2410.c:136: undefined reference to `s3c24xx_dma_init_map'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/soc/samsung/s3c24xx-i2s.c:293: undefined reference to `s3c2410_dma_ctrl'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/built-in.o: In function `s3c2412_i2s_trigger':
sound/soc/samsung/s3c-i2s-v2.c:432: undefined reference to `s3c2410_dma_ctrl'
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Fixes the following build error and warning when OF is not defined:
sound/soc/samsung/smdk_wm8994.c:191:23: error: ‘samsung_wm8994_of_match’ undeclared (first use in this function)
sound/soc/samsung/smdk_wm8994.c:47:32: warning: ‘smdk_board_data’ defined but not used [-Wunused-variable]
of_match_ptr() is used so that samsung_wm8994_of_match gets dropped (as unused)
by the compiler when OF is not defined.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Fixes the following error introduced by commit eca3b01d0885
("ASoC: switch over to use snd_soc_register_component() on s3c i2s"):
sound/soc/samsung/s3c-i2s-v2.c:732:32: error: ‘drv’ undeclared (first use in this function)
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Recent changes through commits c67d0f29262b ("ARM: s3c24xx: get rid
of custom <mach/gpio.h>"), b0161caa72b6 ("ARM: S3C[24|64]xx: move includes
back under <mach/> scope"), 364374121b78 ("ARM: s3c24xx: explicit
dependency on <plat/gpio-cfg.h>") and 41c3548e6da6 ("ARM: s3c64xx: get rid
of custom <mach/gpio.h>") caused build regressions due to broken
dependencies. Fix the following errors by including the necessary header
files explicitly:
sound/soc/samsung/h1940_uda1380.c:56:3: error: implicit declaration of function ‘S3C2410_GPG’
sound/soc/samsung/h1940_uda1380.c:149:18: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/h1940_uda1380.c:234:21: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/h1940_uda1380.c:270:12: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/neo1973_wm8753.c:239:2: error: implicit declaration of function ‘S3C2410_GPJ’
sound/soc/samsung/rx1950_uda1380.c:67:3: error: implicit declaration of function ‘S3C2410_GPG’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘s3c_gpio_cfgall_range’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘S3C2410_GPE’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘S3C_GPIO_SFN’
sound/soc/samsung/s3c2412-i2s.c:87:10: error: ‘S3C_GPIO_PULL_NONE’ undeclared
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘s3c_gpio_cfgall_range’
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘S3C2410_GPE’
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘S3C_GPIO_SFN’
sound/soc/samsung/s3c24xx-i2s.c:395:10: error: ‘S3C_GPIO_PULL_NONE’ undeclared
sound/soc/samsung/smartq_wm8987.c:112:3: error: implicit declaration of function ‘S3C64XX_GPL’
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Since the GPIO jacks are only supported if gpiolib is built and fail to
compile otherwise add a build depedency. This is unlikely to have any
practical impact outside of coverage testing.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
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Currently everytime we get the following message on boot:
fsl-ssi-dai 202c000.ssi: could not get baud clock: -2
This is not really useful information to get on every boot, so make it a debug
message instead.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Since commit aafa85e71a (ASoC: fsl_ssi: Add DAI master mode support for SSI on
i.MX series) master mode is supported, so update the comments and code to
reflect that.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Latest evaluation of the the device has given some patch file additions
for improved performance.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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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: 1a97b7f22774 ("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>
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There are two loops that are almost identical but only with different
checks. Refactor them with a simple helper, and give a bit more
comments what's doing there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The test here is intended intended to prevent shift wrapping bugs when
we do "1U << idx2". We should consider the number of bits in a u32
instead of the number of bytes.
[fix another chunk similarly by tiwai]
Fixes: 7bb2491b35a2 ('ALSA: Add kconfig to specify the max card numbers')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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