This patch adds the support for capture path in sn95031 codec.
This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs
are connected through a MUX to ADC. The TX paths can be assigned to any of the
ADCs or DMICs.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This card uses separate I2S outputs for the front speakers and
headphones, and reverses the order of the three speaker outputs.
To work around this, add a model-specific callback to adjust the
controller's playback routing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: HDA: Fix automute on Thinkpad L412/L512
ALSA: HDA: Fix dmesg output of HDMI supported bits
ALSA: fix invalid hardware.h include in ac97c for AVR32 architecture
ASoC: correct link specifications for corgi, poodle and spitz
ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s
ASoC: Fix codec device id format used by some dai_links
ALSA: azt3328 - fix broken AZF_FMT_XLATE macro
ALSA: Xonar, CS43xx: Don't overrun static array
ASoC: Handle low measured DC offsets for wm_hubs devices
ASoC: da8xx/omap-l1xx: match codec_name with i2c ids
ASoC: WM8994: fix wrong value in tristate function
ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
snd_soc_dapm_put_volsw() has variables for both the unshifted and
shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in
the middle of DAPM sequences) got confused between the two of these.
Since there's no need to keep a copy of the unshifted mask fix this and
simplify the code by using only one mask variable.
[Completely rewrote the changelog to describe the issue -- broonie.]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers"
moved codec and platform driver refcount increments from soc_bind_dai_link
to more appropriate places.
Adjust a little them so that refcounts are incremented before executing the
driver probe functions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Cc: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the Kernel panic issue on accessing davinci_vc in
cq93vc_probe function. struct davinci_vc is part of platform device's
private driver data(codec->dev->p->driver_data) and this is populated
by DaVinci Voice Codec MFD driver.
Signed-off-by: Manjunathappa, Prakash <prakash.pm@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: http://bugs.launchpad.net/bugs/708521
This Edge 13 model has an internal mic at 0x1a and should
therefore use the asus quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to have just CONFIG_ARCH_OMAP2, 3 and 4. The rest
are nowadays just subcategories of these.
Search and replace the following:
ARCH_OMAP2420 SOC_OMAP2420
ARCH_OMAP2430 SOC_OMAP2430
ARCH_OMAP3430 SOC_OMAP3430
No functional changes.
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Thomas Weber <weber@corscience.de>
Acked-by: Sourav Poddar <sourav.poddar@ti.com>
Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier.
This patch adds support for it and stereo output can be active when
"Jack Function" == "TV-OUT" || "Headphone".
As the TPA6130 can output very high volume levels the output is limited
with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allows drivers to distinguish which subsequence is being notified when
they get called back.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Could just as well live in sysfs but sysfs doesn't have the simple
value export helpers debugfs does.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow hookup of cards registered directly with the core to the PM
operations by exporting the device power management operations to
modules, also exporting the default PM operations since it is
expected that most cards will end up using exactly the same setup.
Note that the callbacks require that the driver data for the card be
the snd_soc_card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As requested by Takashi and Jaroslav, these arrays should not be in the
header file.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Inform users about the newly added support for RayDAT and AIO.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers"
moved codec driver refcount increments from soc_bind_dai_link into
soc_probe_codec.
However, the commit didn't remove try_module_get from soc_probe_aux_dev so
the auxiliary device reference counts are incremented twice as the
soc_probe_codec is called from soc_probe_aux_dev too.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to call snd_pcm_period_elapsed() each time a period
elapses - we can call it after we're done once loading/unloading the
FIFO with data. ALSA works out how many periods have elapsed by
reading the current pointers.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Make the AACI announcement printk say which primecell part number
has been found. Display the revision as an unsigned decimal, and
display only the first 8 hex digits of the base address unless it's
larger.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
When double-rate mode was selected, we weren't setting the additional
two channel mask bits to allow double-rate to work. Rearrange the
hw_params code to allow the correct channel mask to be selected.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: http://bugs.launchpad.net/bugs/707902
More Thinkpad machines with invalid SKU found, that disables
automute between speakers and headphones on these machines.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current format selection of FSI-codecs depended on platform information for FSI,
and chip default settings for codecs. It is not understandable/formal method.
This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt.
But FSI can use I2S/PCM and SPDIF format today.
It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF.
So, this patch change FSI platform information to have DAI/SPDIF mode.
If platform selects DAI mode (default),
FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt,
and if it is SPDIF mode, FSI become SPDIF format.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver assumed master->info is not NULL.
This patch allow NULL in master->info
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC'97 codecs only support two channels for recording, so we shouldn't
advertize that there are up to six channels available. Limit the
selection of 4 and 6 channel audio to playback only.
As this adds additional SNDRV_PCM_STREAM_PLAYBACK conditionals, we can
combine some resulting in the elimination of __aaci_pcm_open() entirely,
and making the code easier to read.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Claiming the IRQ each time a playback or capture interface is opened
is wasteful; the second copy of the registered handler is identical to
the first and just wastes resources. Track the number of opens and
only register the handler when necessary.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Relying on the access time of peripherals is unreliable - it depends
on the speed of the CPU and the bus. On Versatile Express, these
timeouts were expiring, causing the driver to fail.
Add udelay(1) to ensure that they don't expire early, and adjust
timeouts to give a reasonable margin over the response times.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Ensure that a timeout coincident with the condition being waited for
results in success rather than failure. This helps avoid timeout
conditions being inappropriately flagged.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
This typo caused the dmesg output of the supported bits of HDMI
to be cut off early.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the non-compiling AC97C driver for AVR32 architecture by
include mach/hardware.h only for AT91 architecture. The AVR32 architecture does
not supply the hardware.h include file.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
CC: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
CS4271 CODEC driver adapted to recently introduced error handling in
snd_soc_update_bits().
Added snd_soc_cache_sync() error handling.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add platform_device_id which can control
PortA/PortB for FSI2-HDMI from platform data.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
During the multi-component patch the s3c24xx i2s driver was renamed from
"s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not
updated to reflect this change as well.
As a result there is no match between the dai_link and the i2s driver and no
sound card is instantiated.
This patch fixes the problem by updating the sound board drivers to use
"s3c24xx-iis" for the cpu_dai_name.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The id part of an I2C device name is created with the "%d-%04x" format string.
So for example for an I2C device which is connected to the adapter with the id 0
and has its address set to 0x1a the id part of the devices name would be
"0-001a".
Currently some sound board drivers have the id part the codec_name field of
their dai_link structures set as if it had been created by a "%d-0x%x" format
string. For example "0-0x1a" instead of "0-001a".
As a result there is no match between the codec device and the dai_link and no
sound card is instantiated.
This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
We generally refer to registers as unsigned ints (including in the
underlying CODEC driver operation).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cleanly revert to non-macro implementation of
snd_azf3328_codec_setfmt(), to fix last-minute functionality breakage
induced by following checkpatch.pl recommendations without giving them
their due full share of thought ("revolting computer, ensuing PEBKAC").
I would like to thank Jiri Slaby for his very timely (in -rc1 even)
and unexpected (uncommon hardware) "recognition of the dangerous situation"
due to his very commendable static parser use. :)
Reported-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed the Asus A52J quirk to use the asus model instead of the
hp_laptop model, which fixes the external mic input. Added an Asus
U50F quirk to use the asus model. For the cxt5066 codecs, added
checking of the digital output pins to determine which digital output
nodes to use instead of always using node 0x21, since some systems
have node 0x12 connected to a SPDIF out jack.
[A slight modification for better readability by tiwai]
Signed-off-by: Andy Robinson <ajr55555@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added support for EDB93xx sound with CS4271 CODEC.
Features:
- Playback, Capture
- Sample rates from 8kHz to 96kHz tested
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: http://bugs.launchpad.net/bugs/701271
This new model, named "asus", is identical to the "hp_laptop" model,
except for the location of the internal mic, which is at pin 0x1a.
It is used for Asus K52JU and Lenovo G560.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Four very similar procedures - one for each model - now
refactored into one. This isn't all duplicated code, but a step
in the right direction.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of
8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers()
for (i = 2; i <= 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
will overrun the array when 'i == 8'.
I guess that what's needed to fix it is the trivial patch below, but I
must admit that I have no idea about this code, so I may very well be
wrong. Additionally, I have no way to actually test this, so all I know is
that the below compiles. Someone who actually knows this code should take
a look before anything is comitted - consider the below (not much more
than) a bug report.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add snd_soc_dai_ops :: set_fmt to FSI driver and
select master/slave clock mode by snd_soc_dai_set_fmt on
fsi-xxx.c instead of platform infomation code.
This patch remove fsi_is_master function which is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch support LEFT_J / I2S only for now
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch makes it easy to see when the syncing process begins and
ends. You can also enable the snd_soc_reg_write tracepoint to see
which registers are being synced.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for CS4271 codec to ASoC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DC servo codes are actually signed numbers so need to be treated as
such.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c
is not matching with the i2c ids in the board file. Without this fix the
soundcard does not get detected on da850/omap-l138/am18x evm.
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Tested-by: Dan Sharon <dansharon@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37)
Also, update platform_data GPIO handling to have an explicit "don't
touch this pin" option.
Add #defines for the GPIO pin functions.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's an internal function so shouldn't be exported (as sparse points
out).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Incorporate the use of the cache_bypass functionality in the
syncing functions. The snd_soc_flat_cache_sync() need not be
hooked as there is no performance benefit from using the
cache_bypass option.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is primarily needed to avoid writing back to the cache
whenever we are syncing the cache with the hardware. This gives a
performance benefit especially for large register maps.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: http://bugs.launchpad.net/bugs/705323
Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_realtek.c: In function ‘alc_apply_fixup’:
sound/pci/hda/patch_realtek.c:1724:14: warning: unused variable ‘modelname’
snd_printdd() is evaluated only when CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix wrong value in wm8994_set_tristate func. when updating reg bits,
it should use "value", not "reg".
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
In the wm8995_set_tristate() function when updating the register
bits use the value and not the register index as the value argument.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Fix a bunch of
warning: ‘inline’ is not at beginning of declaration
messages when building a 'make allyesconfig' kernel with -Wextra.
These warnings are trivial to kill, yet rather annoying when building with
-Wextra.
The more we can cut down on pointless crap like this the better (IMHO).
A previous patch to do this for a 'allnoconfig' build has already been
merged. This just takes the cleanup a little further.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
With larger devices there may be many widgets of the same type in series
in an audio path. Allow drivers to specify an additional level of ordering
within each widget type by adding a subsequence number to widgets and then
splitting operations on widgets so that widgets of the same type but
different sequence numbers are processed separately. A typical example
would be a supply widget which requires that another widget be enabled
to provide power or clocking.
SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided
allowing this to be used with PGAs and supplies as these are the most
commonly affected widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Rather than passing the sequence to use for DAPM widgets around by reference
explicitly say if we're powering up or down until the point where we need
the sequence itself. This should make no practical difference in itself but
supports future refactoring.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The interface between sst platform driver and intel sst dsp driver
have been changed in Greg's staging tree - next branch
This patch adds the interface changes compatible with the new interface
in Greg's staging tree
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 03b7a1ab55.
This commit was mistakenly re-introduced. While the change is harmless
(as ALC887 uses patch_alc888() now), we should get rid of any wrong code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_dapm_new_widgets will call dapm_power_widgets at
the end, so there is no need to call snd_soc_dapm_sync
after snd_soc_dapm_new_widgets.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
WM8750 address is 0x1b, not 0x1a. Without this fix ALSA detects no sound
cards on Zipit
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix jack detection on Zipit Z2, otherwise it
disables headphones output when jack is connected
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While I2S/TDM/AC97 DAI is built-in, others are compiled as modules,
SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols.
Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We don't want to use internal frame syncs otherwise we sometimes
get out of sync, so don't enable them when setting up the SPORT.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to tweak how we query the active capture/playback state after
the recent overhauls of common code.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
One spot was missed in this driver when converting from
snd_soc_dai.private_data to snd_soc_dai_get_drvdata.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix initialization for HP 2011 notebooks
ALSA: hda - Add support for VMware controller
ALSA: hda - consitify string arrays
ALSA: hda - Add add multi-streaming playback for AD1988
ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
ASoC: WM8990: msleep() takes milliseconds not jiffies
ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu
ALSA: constify functions in ac97
ASoC: WL1273 FM radio: Fix breakage with MFD API changes
ALSA: hda - More coverage for odd-number channels elimination for HDMI
ALSA: hda - Store PCM parameters properly in HDMI open callback
ALSA: hda - Rearrange fixup struct in patch_realtek.c
ALSA: oxygen: Xonar DG: fix CS4245 register writes
ALSA: hda - Suppress the odd number of channels for HDMI
ALSA: hda - Add fixup-call in init callback
ALSA: hda - Reorganize fixup structure for Realtek
ALSA: hda - Apply Sony VAIO hweq fixup only once
ALSA: hda - Apply mario fixup only once
ALSA: hda - Remove unused fixup entry for ALC262
The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)
It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */
It must be raised before the first SPI write happens, or the write will
fail, leading to:
[ 23.248721] invalid CS8427 signature 0x0: let me try again...
I theorize that 4eb4550ab3
is no longer needed, it was a different way to workaround
the problem.
[fixed variable decleration by tiwai]
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes for HP 2011 notebooks: enable dock ports and disable BTL
initialization in the driver.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio
Controller.
[changed to use AZX_DRIVER_GENERIC by tiwai]
Signed-off-by: Bankim Bhavsar <bbhavsar@vmware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Attached a patch which add a new model to support multi-streaming
playback for ad1988.
playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.
Tested with ad1988a rev2 codec on asus P5B-V motherboard.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.
I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3
Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:
Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.
Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.
So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.
In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.
2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.
DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...
If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).
In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Directly accessing the register cache means that we can't use anything
except a flat register cache so use snd_soc_update_bits().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for
handsets rich in multimedia features such as GPS, mobile TV, digital audio
playback and gaming.
This driver was originally written by Graeme Gregory and has been maintained
out of tree by Mark Brown and Dimitris Papastamos.
Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com>
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The machine driver can't register the card directly and need to do this thru
soc-audio device creation
This patch allows the register and unregister card to be directly called by
machine drivers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix playback/capture channels patch to change supported playback
channels of au8830 to 1,2,4 and capture channels to 1,2.
This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to
set 3 Channels
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'linux-next' of git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci-2.6:
PCI/PM: Report wakeup events before resuming devices
PCI/PM: Use pm_wakeup_event() directly for reporting wakeup events
PCI: sysfs: Update ROM to include default owner write access
x86/PCI: make Broadcom CNB20LE driver EMBEDDED and EXPERIMENTAL
x86/PCI: don't use native Broadcom CNB20LE driver when ACPI is available
PCI/ACPI: Request _OSC control once for each root bridge (v3)
PCI: enable pci=bfsort by default on future Dell systems
PCI/PCIe: Clear Root PME Status bits early during system resume
PCI: pci-stub: ignore zero-length id parameters
x86/PCI: irq and pci_ids patch for Intel Patsburg
PCI: Skip id checking if no id is passed
PCI: fix __pci_device_probe kernel-doc warning
PCI: make pci_restore_state return void
PCI: Disable ASPM if BIOS asks us to
PCI: Add mask bit definition for MSI-X table
PCI: MSI: Move MSI-X entry definition to pci_regs.h
Fix up trivial conflicts in drivers/net/{skge.c,sky2.c} that had in the
meantime been converted to not use legacy PCI power management, and thus
no longer use pci_restore_state() at all (and that caused trivial
conflicts with the "make pci_restore_state return void" patch)
With this change, I can capture from a microphone plugged into the
mic jack on Harmony (after unmuting Left Input PGA, and maybe turning
up the gain there too).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
That config variable doesn't exist in the mainline kernel, and hence
the dependency shouldn't either.
In linux-tegra-2.6.36, the dependency did exist to avoid a conflict with
the old non-ALSA Tegra I2S driver. However, this isn't and won't be
upstreamed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.
Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit ad09fc9d21 didn't cover the
case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called.
Put the hw_constraint there, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in
hinfo, but these aren't properly set back to the current runtime
record since these have been set beforehand in azx_pcm_open().
This patch fixes the behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It looks like that HDMI codecs don't support the odd number of channels
although HD-audio spec doesn't have the restriction. Add the
hw_constraint to limit to only the even number of channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the soc_probe initializes the card hence it does the card list
initialzation. But if machines directly register the card they would need to
do these steps, so putting them as inline would save lot of code
This patch adds an inline to do list initialzation
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <harsha.priya@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The card debugfs initialization is done in soc_probe but would be better if it
is done when the card in registered
This patch moves the debugfs initialization to register_card()
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <harsha.priya@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In soc_remove() the card resources are cleaned up.
This can also be done in card_unregister()
This patch move this cleanup into a new function and calls it from
card_unregister. This paves way for further work to allow card registartion
from machine.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume. The new action type
ALC_FIXUP_ACT_INIT is used for this case.
So far, only ALC275_FIXUP_SONY_HWEQ uses this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries. This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A recent discussion on linux-arm-kernel noted that the value returned by
clk_get_sys is an opaque token, and not strictly a pointer; it is
meaningful only to the clock API, clients should not dereference the value,
and the clock API must accept any non-IS_ERR value it returned.
Hence, only IS_ERR is appropriate to interpret the result, not
IS_ERR_OR_NULL.
I checked that clk_get_sys in both ASoC's for-next and Tegra's for-next
do behave as described; NULL is not returned in the case of error.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that all calls to readable_register()/volatile_register() go via
the snd_soc_codec function pointers.
If the default register access table has been given but no functions
for handling readable()/volatile() registers, use the default ones provided
by soc-cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For common scenarios, device drivers can provide a table of all the
registers that are at least either readable/writable/volatile. The idea
is that if a register lookup fails, all of its read/write/vol members
will be zero and will be treated as default. This also reduces the
size of the register access array.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When only one mic is available and it's an analog mic, the current
IDT/STAC parser may give an Oops.
Reference: bko#25692
https://bugzilla.kernel.org/show_bug.cgi?id=25692
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Simplify the use of reg_size, by calculating it once and storing it in
the codec structure for later reference. The value of reg_size is
reg_cache_size * reg_word_size.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the addition of the cache fallback functionality, it is necessary
to ensure that if the register defaults cache was marked as __devinitconst
and the LZO compression is not compiled in the kernel, the fallback to
flat compression will still use a copy of the defaults cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With GPIO2-fixup, another fixup for NID 0x19 was missing because the
fixup is applied only once. Add the corresponding verb to the entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
SONY VAIO ALC275 default BIOS verb set the hardware EQ to disable.
Enable it when driver is loading.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo NB 0x9e54 use the external AMP in an inverted manner.
Set EAPD to low will enable the AMP.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added hardware constraint in patch_hdmi.c to disable
channels 4/6 which are not supported by some older
NVIDIA GPUs.
Signed-off-by: Nitin Daga <ndaga@nvidia.com>
Acked-By: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dynamic PCM restriction based on ELD information may lead to the
problem in some cases, e.g. when the receiver is turned off. Then it
may send a TV HDMI default such as channels = 2. Since it's still
plugged, the driver doesn't know whether it's the right configuration
for future use. Now, when an app opens the device at this moment,
then turn on the receiver, the app still sends channels=2.
The right solution is to implement some kind of notification and
automatic re-open mechanism. But, this is a goal far ahead.
This patch provides a workaround for such a case by providing a new
module option static_hdmi_pcm for snd-hda-codec-hdmi module. When
this is set to true, the driver doesn't change PCM parameters per
ELD information. For users who need the static configuration like
the scenario above, set this to true.
The parameter can be changed dynamically via sysfs, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls':
sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm'
Introduced by commit 39646871a4 ("ASoC:
tpa6130a2: Replace DAPM code with direct interface").
The DAPM code has been removed from the driver, but the
dapm struct remained.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The L/R LOM line can be invertined side of the
corresponding DAC, or inverted from the corresponding
LOP.
Add control for user space to select the source of the
LOM inversion.
When only the analog bypass is enabled, and the LOM
is inverted from DAC output, we need to power the
corresponding DAC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This codec is to be used by the DMIC driver to
control the DMIC codec. This driver will be used on future
implementations of the DMIC driver to support codec specific
features.
At this time, the codec driver just registers the codec DAI.
Signed-off-by: David Lambert <dlambert@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl().
Wake up poll waiters before returning from usb_stream_hwdep_ioctl().
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The US-122L always reads 9 bytes per urb unless they are set to 0xFD.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use Takashi's clean up code to make the cache manipulation code more
readable.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The size of the lzo syncing bitmap was incorrectly set to the size
of the cache times the word size, however, the correct size is the
size of the cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a mixer control to switch between the optical and coaxial S/PDIF
inputs on the HT-Omega Claro and Claro halo cards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable the X-Meridian's CD input and the X-Meridian 2G's potential
MIDI ports.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch remove including unnecessary/duplicated headers which relative
with Samsung SoCs.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update snd_soc_update_bits() so that it returns a negative error code if the
the read or write operation fails.
Note that currently, a lot of the lower-level read functions have an unsigned
integer return type (and some of them even try to return a negative number),
but this code still appears to work in those cases.
An examination of the code shows that all current callers are compatible with
this change.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Few funtions can be modularized in this driver to make them look cleaner
like managing the stream status with locks and filling pcm parameters.
This patch adds helper functions to do the same.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 6d803ba736 "ARM: 6483/1: arm & sh:
factorised duplicated clkdev.c" broke compilation of migor audio. Use the
correct header to fix the problem.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the CS4270 driver to use ASoC's internal codec register cache feature.
This change allows ASoC to perform the low-level I2C operations necessary to
read the register cache. Support is also added for initializing the register
cache with an array of known power-on default values.
The CS4270 driver was handling the register cache itself, but somwhere along
the conversion to multi-compaonent, this feature broke.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Everything else is using snd_soc_ so we should use it here too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many portions of Tegra ASoC machine drivers will be similar or identical.
To avoid cut/paste, this file will act as a repository for all that common
code. For now, it solely includes code to reprogram the audio PLL for
44.1KHz- vs. 48KHz-based sample rates.
Signed-Off-By: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This provides an ASoC DAI interface for Tegra's I2S controller.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This provides an ASoC platform driver that manages Tegra's APB DMA
controller.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DAS (Digital Audio Switch) is a mux/crossbar which sits between
the DACs (Digital Audio Controllers) and the DAPs (Digital Audio
Ports). Audio data may be routed between DACs and DAPs in various
combinations as required by board design/application.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.
Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The infrastructure for handling NULL default register maps is already
included in the soc-cache code, just ensure that we don't try to dereference
a NULL pointer while accessing a NULL register cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of the generic Oxygen, use the actual card name, if known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apparently, the revision is 2 on all sold sound cards, so this
information is not actually useful.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to select between the on-board and extension board
S/PDIF inputs for the X-Meridian (2G).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to prevent capturing S/PDIF samples that are not
marked as valid (non-audio or corrupted samples).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the 24-bit audio I/Os of the Edirol SD-90 interface.
Reported-any-tested-by: Jim Grusendorf <alsa-user@grusendorf.ca>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar HDAV1.3 Slim sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar DG sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the AuzenTech X-Meridian 7.1 2G sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate
modes (64-96 kHz) can be reduced to 128x without reducing sound quality.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the get_i2s_mclk callback with tables of MCLK values. This
simplifies the MCLK-handling code in both the framework and the model-
specific drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not apply the headphone gain offset to any but the front DAC. These
DACs would not be used in headphone mode, so this saves a few register
writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the DAC Oversampling mixer control because this setting does not
make much sense.
For cards with the H6 daughterboard, 128x oversampling was disabled
anyway because these high MCLK frequency would not be compatible with
the connector cable.
For cards without the H6 daughterboard, 128x gives a slightly higher
output quality; there is no reason to reduce it to 64x except for saving
power, but then these cards have not been designed to be power efficient
anyway (the D2's blinkenlights cannot be disabled).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because of the unshielded connector cable, it is important to use as low
a master clock frequency as possible with the H6.
For double rate modes (64-96 kHz), the MCLK rate is unconditionally
lowered from 512x to 256x because the higher rate would not improve
anything.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clock output of the CS2000, which is used as master clock for the
DACs, was using half the actual master clock frequency for some reason.
Using the theoretically correct frequency seems also to work in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST Deluxe, remove all mixer controls that would
require I2C communication with the third DAC, which does not work
because of an addressing conflict with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the PCM format used for the PCM1796 from left-justified to I2S to
ensure that the correct format is used even for the Essence ST Deluxe's
center/LFE DAC, where I2C does not work because of an address conflict
with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM1796 needs the master clock for I2C communication to work, so
add delays after clock changes to ensure that the clock is stable when
we try to write the DACs' registers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the I2C communication reliable when using the H6 daughterboard,
reduce the I2C clock frequency.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix wrong register bits for SPI clock cycle times longer than 160 ns,
and adjust the polling loop timeout for these speeds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of DACs can now be deduced from the dac_channels_mixer field,
so the private_data field is no longer needed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For cards like the Xonar HDAV1.3, differentiate between the number of
PCM channels that can be played and the number of channels whose volume
can be adjusted.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs. But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs. But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins. This results in some
mis-handling of these pins for Realtek codec parser. It takes as if
these are pure line-out jacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Enable Mic Jack during glue driver init, otherwise capture will not work.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since direct register writes may confuse the drivers and are supposed
to be used only in diagnostic situations discourage their use in
production by tainting the kernel when we do a write.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch moves PCM specific definitions, they doesn't need to be
shared, into 'pcm.c' from 'pcm.h'.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves DMA specific definitions, they doesn't need to be
shared, into 'dma.c' from 'dma.h'. And remove unnecessery definitions
on 'dma.h'.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves AC97 specific definitions, they doesn't need to be shared,
into 'ac97.c' from 'ac97.h'. This patch also remove header file, 'ac97.h',
and remove includes on machine drivers.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
previous commit e62255f2ad introduced DDEBUG flag in Makefile
This causes all debug statemenets to be ON.
This patch removes this flag
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this patch fixes the style inconsistency by removing empty space in MODULE_ALIAS
also removes a redundant return statement
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this patch fixes inconsistencies commented by Mark.
This also fixes few other style things in audio_map & header file
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_pcm_hw_param_near() will leak the memory allocated to 'save' if the
call to snd_pcm_hw_param_max() returns less than zero.
This patch makes sure we never leak.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5184
A user reported on the alsa-devel mailing list that he needs to use
the vostro model quirk to have audible playback, so apply it for his
PCI SSID.
Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.
Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.
Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'next-spi' of git://git.secretlab.ca/git/linux-2.6: (77 commits)
spi/omap: Fix DMA API usage in OMAP MCSPI driver
spi/imx: correct the test on platform_get_irq() return value
spi/topcliff: Typo fix threhold to threshold
spi/dw_spi Typo change diable to disable.
spi/fsl_espi: change the read behaviour of the SPIRF
spi/mpc52xx-psc-spi: move probe/remove to proper sections
spi/dw_spi: add DMA support
spi/dw_spi: change to EXPORT_SYMBOL_GPL for exported APIs
spi/dw_spi: Fix too short timeout in spi polling loop
spi/pl022: convert running variable
spi/pl022: convert busy flag to a bool
spi/pl022: pass the returned sglen to the DMA engine
spi/pl022: map the buffers on the DMA engine
spi/topcliff_pch: Fix data transfer issue
spi/imx: remove autodetection
spi/pxa2xx: pass of_node to spi device and set a parent device
spi/pxa2xx: Modify RX-Tresh instead of busy-loop for the remaining RX bytes.
spi/pxa2xx: Add chipselect support for Sodaville
spi/pxa2xx: Consider CE4100's FIFO depth
spi/pxa2xx: Add CE4100 support
...
This patch adds the makefile and kconfig changes for mid asoc drivers:
platform and machine driver which are introduced in 2 preceeding patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the medfield machine driver
Machine driver glues the sn95031 codec driver and sst_platform driver to form the asoc sound driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the platform driver for mid asoc drivers. This platfrom
driver sends commands to sst dsp engine driver for the dai operations.
For this purpose it depends on intel_sst driver which is currently in
staging tree
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the sn95031 asoc codec driver. This driver currently
supports only playback. Capture and jack detection to be added later
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The soc-core takes the platform and codec driver reference during probe. Few of
these references are not released during remove. This cause the platform and
codec driver module unload to fail.
This patch fixes by the taking only one reference to platform and codec module
during probe and releases them correctly during remove. This allows load/unload
properly
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The rest of ASoC is using SND_SOC_ as the prefix for all the Kconfig
symbols so do so for the new Samsung drivers too, rather than using
ASOC_ as they currently are.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of replacing 'milisec' by 'millisec', I decided to use
the more common SI unit. Other drivers use 'milliseconds'
or 'ms', too ('millisec' is never used).
Cc: Geert Uytterhoeven <Geert.Uytterhoeven@sonycom.com>
Cc: Jiri Kosina <trivial@kernel.org>
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Acked-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
- much improved implementation due to clean codec hierarchy
- preparation for potential per-codec spinlock change
NOTE: additionally removes a chip->pcm[codec_type] NULL ptr check
(due to it requiring access to external chip struct),
however I believe this to be ok since this condition should not occur
and most drivers don't check against that either.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use a separate variable for the frequency part, don't always "or" it
- use a "clever"(?) macro to shorten the code
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- correct samples to be POSIX shell compatible
- add logging of jiffies value in _pointer()
- several comments
- cleanup
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sjoerd Simons reports that, without using position_fix=1, recording
experiences overruns. Work around that by applying the LPIB quirk
for his hardware.
Reported-and-tested-by: Sjoerd Simons <sjoerd@debian.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The load_mixer_volumes() function, which can be triggered by
unprivileged users via the SOUND_MIXER_SETLEVELS ioctl, is vulnerable to
a buffer overflow. Because the provided "name" argument isn't
guaranteed to be NULL terminated at the expected 32 bytes, it's possible
to overflow past the end of the last element in the mixer_vols array.
Further exploitation can result in an arbitrary kernel write (via
subsequent calls to load_mixer_volumes()) leading to privilege
escalation, or arbitrary kernel reads via get_mixer_levels(). In
addition, the strcmp() may leak bytes beyond the mixer_vols array.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'spi' of git://git.linutronix.de/users/bigeasy/soda into spi/next
spi/pxa2xx: register driver properly
spi/pxa2xx: add support for shared IRQ handler
spi/pxa2xx: Use define for SSSR_TFL_MASK instead of plain numbers
arm/pxa2xx: reorgazine SSP and SPI header files
spi/pxa2xx: Add CE4100 support
spi/pxa2xx: Consider CE4100's FIFO depth
spi/pxa2xx: Add chipselect support for Sodaville
spi/pxa2xx: Modify RX-Tresh instead of busy-loop for the remaining RX bytes.
spi/pxa2xx: pass of_node to spi device and set a parent device
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but there are quite a few drivers left which now have an unused reg_cache field in
their private device struct.
This patch removes these unused fields.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8753 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Furthermore the generic cache uses zero-based numbering while the wm8753 cache
uses one-based numbering.
Thus we end up with two from each other incoherent caches, which leads to undefined
behaviour and crashes.
This patch fixes the issue by changing the wm8753 driver to use the generic
register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm9090 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm9090 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8962 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8962 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8955 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8955 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8904 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8904 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: Ian Lartey <ian@opensource.wolfsonmicro.com>
Cc: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8741 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8741 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: Ian Lartey <ian@opensource.wolfsonmicro.com>
Cc: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8523 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8523 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: Ian Lartey <ian@opensource.wolfsonmicro.com>
Cc: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the max98088 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the max98088 driver to use the
generic register cache in its private functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: Peter Hsiang <Peter.Hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
Some codec drivers do not initialize the control_type field in their private
device struct, but still use it when calling snd_soc_codec_set_cache_io.
This patch fixes the issue by properly initializing it in the drivers probe
functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37 only)
The assignment to the local variable 'channel' in
snd_ca0106_pcm_pointer_capture() is a little crazy. Order of assignment is
undefined. This fixes it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
They're very verbose and extremely repetitive so bulk up the kernel more
than is ideal. If required we can readd with WRITE_SEQUENCER_n type
definitions that cover the entire register bank in a few defines.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Due to a typographical error in the erratum workaround it was never
functional so just remove it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix GPIO2-fixup for Sony laptops
ALSA: hda - Try to find an empty control index when it's occupied
ALSA: hda - Fix conflict of d-mic capture volume controls
ALSA: hda - Don't apply ALC269-specific initialization to ALC275
ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
ALSA: pcm: remember to always call va_end() on stuff that we va_start()
ALSA: HDA: Add auto-mute for Thinkpad SL410/SL510
pci_restore_state only ever returns 0, thus there is no benefit in
having it return any value. Also, a large majority of the callers do
not check the return code of pci_restore_state. Make the
pci_restore_state a void return and avoid the overhead.
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Signed-off-by: Jon Mason <jon.mason@exar.com>
Signed-off-by: Jesse Barnes <jbarnes@virtuousgeek.org>
The fix-up entries by the commit 2785591a97
ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
weren't applied in the right position. They had to be before the quirk
entry matching to all Sony devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for 24 bit audio (with S32_LE msbits 24).
The reason to limit the msbits to 24, is that the FIFO
can be configured for 16 or 24 bit layout.
It is unknown how the codec would downsample from 32 to
24 bit, if the interface is configured to receive 32
bit data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the structure of FIFO handling in order to
pave the way for adding 32/24 bit audio support.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The manual FIFO configuration was the first version to enable
the use of the FIFO in the codec.
It had served it's purpose as debugging aid, but the automatic
FIFO configuration is much safer to use.
The removal of the manual controls, and configuration makes
it easier to add new features for the codec later, since
the manual mode neded different ways to calculate, and
protect against misconfiguration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make the phoenix HS and HF drivers use the new DAPM driver
widget in order to guarantee power ON/OFF order sequence.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds McBSP support for the OMAP4 CPU
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When a mixer control element was already created with the given name,
try to find another index for avoiding conflicts, instead of breaking
with an error. This makes the driver more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the d-mics are assigned to the same purpose of another analog mic
pins, the driver doesn't compute the index properly, resulting in an
error with "existing control". This patch fixes it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conflicts:
MAINTAINERS
arch/arm/mach-omap2/pm24xx.c
drivers/scsi/bfa/bfa_fcpim.c
Needed to update to apply fixes for which the old branch was too
outdated.
The WM8995 is a digital audio hub CODEC designed for smartphones.
The current driver supports most of the basic functionality of the
WM8995.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_jack_new()'s first parameter was changed from snd_soc_card to
snd_soc_codec after Multi-Component support patches. So, this patch
fixes parameter that we missed.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add jack detection interrupt trace to Wolfson CODEC drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This avoids blocking the IRQ thread and allows further bounces to extend
the debounce time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
FSI-AK4642 and FSI-DA7210 are depend on I2C, not I2C_SH_MOBILE
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make LZO cache compression optional as it pulls in the kernel wide LZO
implementation and rbtree compression is generally more efficient for
typical register maps, especially in terms of CPU performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This makes it easier to make cache types build time configurable as we
don't have a hard dependency on a given cache being built in.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
These would have been used if we'd done manual clock divider setup,
but we didn't.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ALC275 doesn't require the ALC269 (and its variants) specific init
sequences. Add the check of codec id.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set GPIO2 for some Sony VAIO with ALC275 to fix speaker output.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Coverity checker spotted that we do not always remember to call
va_end() on 'args' in failure paths in snd_pcm_hw_rule_add().
Here's a patch to fix that up (compile tested only) - it also removes
some annoying trailing whitespace that caught my eye while I was in the
area..
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change non-standard mic control names to standard control names
to clean up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Usually external microphones are just labelled "Mic", so rename
"Ext Mic" and "External Mic" to "Mic" to clear up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Int Mic" and "Internal Mic" both mean the same thing, so rename
the former to the latter in order to clean up the namespace a little.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ASoC machine driver for SMDKV310/C210 boards that have
a WM8994 attached to I2S-0.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since most newer SMDKs have I2S0 routed to the WM8580's Primary DAI,
future changes can be minimized if the default CPU DAIs are set to
0, rather than 2.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Flush the FIFO while stopping the channel rather than starting.
This saves time during stream start and keeps the FIFOs clean
when the channel is idling.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the rclk_srcrate is cleared upon startup, it should be
initialized upon second and later 'open' calls to the device
with same root-clock source. The bug is otherwise visible in
Codec-Slave mode.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: http://launchpad.net/bugs/580006
SKU turns off auto-mute for these machines, so ignore the SKU.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new type is a virtual version of snd_soc_dapm_mux. It is used
when a backing register value is not necessary for deciding which
input path to connect. A simple virtual enumeration control e.g.
SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used
to choose which path to connect.
The snd_soc_dapm_virt_mux type ensures that during the initial
path setup, the first (which is also the default) input path will
be connected.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix conflict of Mic Boot controls
ALSA: HDA: Enable subwoofer on Asus G73Jw
ALSA: HDA: Fix auto-mute on Lenovo Edge 14
ASoC: Fix bias power down of non-DAPM codec
ASoC: WM8580: Fix R8 initial value
ASoC: fix deemphasis control in wm8904/55/60 codecs
Due to the recent change for multiple mics assignment, we need to handle
the index of each Mic Boost control respectively. Otherwise the driver
gets the control element conflicts, and gives the unsable state.
Reference: kernel bug 25002
https://bugzilla.kernel.org/show_bug.cgi?id=25002
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver had printed under/over run error
if status register have its error bit.
But runtime print cause the next error
because print out is slow.
This patch add error counter and print error when sound stop
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Attempt to minimise audible effects from mixer and mux updates by
implementing the actual register changes between powering down widgets
that have become unused and powering up widgets that are newly used.
This means that we're making the change with the minimum set of widgets
powered, that the input path is connected when we're powering up widgets
(so things like DC offset correction can run with their signal active)
and that we bring things down to cold before switching away. Since
hardware tends to be designed for the power on/off case more than for
dynamic reconfiguration this should minimise pops and clicks during
reconfiguration while active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Multiple quirk functions were using the exact same code to verify if the Mic
jack was plugged and mute the Mic accordingly
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ramp functions for the headset and handsfree outputs
in order to reduce the pops during power on/off sequences.
In order to give more control to volume ramp, step size and delay
between steps can be specified.
The patches are based on wm8350 implementation from Liam
Girdwood.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.
This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.
DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set default association/sequence right on pin 0x17 in order for
the automatic parser to recognize the subwoofer correctly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/690530
The SKU value of this machine dictates that auto-mute should be
disabled. Since the SKU value is similar to the PCI SSID, the most
likely conclusion is that the SKU value should be ignored.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
ALSA: hda - Reset sample sizes and max bitrates when reading ELD
ALSA: hda - Always allow basic audio irrespective of ELD info
ALSA: hda - Do not wrongly restrict min_channels based on ELD
ASoC: Correct WM8962 interrupt mask register read
ASoC: WM8580: Debug BCLK and sample size
ASoC: Fix resource leak if soc_register_ac97_dai_link failed
ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain
ASoC: Fix off by one error in WM8994 EQ register bank size
ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic
ALSA: hda - Enable jack sense for Thinkpad Edge 13
ALSA: hda - Fix ThinkPad T410[s] docking station line-out
ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture
Gain for LineInAmp Right uses LINEGAIN[5:3], which means that
offset for right channel should be 4.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some gains were incorrectly configured for dB values.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
After coming back from suspend, the timeout waiting for Phoenix
chip to complete its power up sequence is not enough, which leaves
the codec cache value for some registers in an outdated state.
Increase the timeout value to wait for the power up sequence
to correclty complete.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Enable plug detection interrupt mask in order to get headset
PLUGINT/UNPLUGINT interrupts.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
On Phoenix 1.1, the INTID register default value is an invalid
one, causing the interrupt handler to think the phoenix power on
sequence is ready before it actually finishes.
This causes some i2c errors when trying to configure twl.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Phoenix 1.1 supports automatic power on sequence, a
verification is added to use it with new revision of
the chip.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The mux control has 4 elements not 3
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The twl6040 can support more sample rates other than 88.2 and 96k.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch moves all the PCM error handling for clock config
out of trigger() and startup() and into prepare().
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch restores the CODEC bias level at resume().
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for the twl6040 headset and handset
MUX controls.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Multiples interrupts can be received. The irq handler is modified
to attend all of them.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Update the codec to use the new twl core register macros
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use jack framework to enable detection for the headset microphone
and stereo output in the sdp4430.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: David Anders <x0132446@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for reporting twl6040 headset and
handset jack events.
The machine driver retrieves and report the status through
twl6040_hs_jack_detect.
A workq is used to debounce of the irq.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
twl4030 series of codecs supports S32_LE with msbits=24.
Replace the S24_LE with S32_LE format, and add constraint
for 24msbit in case of 32 S32_LE format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Remove redundant parentheses/spaces in the use of the sizeof
operator.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps ensure that the ramp logic is reset when powering back up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.
Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
soc_unregister_ac97_dai_link() takes a CODEC as an argument, not a
rtd like the registration function, so give it what it's looking for.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Fix "ASoC: Fix bias power down of non-DAPM codec" for 3.6.37 will cause a
build error when merging into ASoC for-2.6.38. Fix the issue by doing a
change that commit ce6120c "ASoC: Decouple DAPM from CODECs" would do.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.
Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mute speakers when a line-out jack is plugged as well as headphone jacks
with the new Conexant codec parser in the auto mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The compiler really ought to have been warning about unreferenced
variables...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
flush_scheduled_work() is deprecated and scheduled to be removed.
* cancel[_delayed]_work() + flush_scheduled_work() ->
cancel[_delayed]_work_sync().
* wm8350, wm8753 and soc-core use custom code to cancel a delayed
work, execute it immediately if it was pending and wait for its
completion. This is equivalent to flush_delayed_work_sync(). Use
it instead.
Signed-off-by: Tejun Heo <tj@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide the user with a boolean control then automatically select
the deemphasis filter most closely matching the sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We're already flagged as using symmetric rates so we don't need to
have a custom implementation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If the following scenario has been followed:
1. Enable analog bypass
amixer sset 'Analog Left Bypass' on
amixer sset 'Analog Right Bypass' on
2. Start playback
aplay -fdat -d3 /dev/zero
After the playback stopped (3 sec), and the soc timeout (5 sec),
the digital parts of the codec will remain powered up.
This means that the DAI clocks are continue to run, the
oscillator remain operational, etc.
Use the SND_SOC_DAPM_POST_PMD widget to get notification
about the stopped stream, and power down the digital
part of the codec.
If the analog bypass is enabled, than the codec will remain in
BIAS_ON level, and things will work correctly.
In case, if the bypass is disabled, than the codec will
fall to BIAS_STANDBY than to BIAS_OFF level, as it used
to.
The digital part of DAC33 is initialized at every stream start
(DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec)
will have working DAI.
When the codec is coming out from BIAS_OFF, the full power-up
sequence followed by the same DAPM_PRE widget event will power up
the digital part.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on
H1940.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The oversampling rate of the DAC and ADC can be controlled to optimise
for either low power consumption or maximum performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Tune the FLL gain for optimal performance according to evaluation
results.
Signed-off-by: Mario Becroft <mb@gem.win.co.nz>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It does not make sense to set clientdata to onyx in onyx_i2c_remove()
as we are going to kfree onyx.
What we really want here is i2c_set_clientdata(client, NULL);
Since the i2c core will take care of it now, so this patch just removes it.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changes to both I2S and PCM code:
- Rates list extended up to 96kHz, it's tested on EDB9302 and works for both capture and
playback.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Add FM stereo pins to the machine driver and add them as a
dapm widget.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Fix the compilation error introduced by patch:
ASoC: tlv320dac33: Avoid multiple soft power up
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The power for the DACs need to be enabled, even when only
the analog bypass is in use with the codec, otherwise
the audio is going to be distorted.
Make sure that the DACs are powered all the time, when
there is audio activity.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use better name for the widget, and remove the 'Power'
from it's name.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2ccad), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.
Cc: stable@kernel.org (2.6.35+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create fixup function for the mario model and override amp capabilities
for NID 0x2
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Facilitate fixup for realtek codecs via modelname lookup of fixup
data. Fallback to quirk based lookup in absence of model definition.
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 supports mono signals - enable this in the driver. With DSP
mode an automatic data channel selector is available, activate this
when in mono mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In case the codec driver did not provide a read/write function,
codec->driver->read|write will be NULL. Ensure that we use the one
specified in codec->read|write to avoid oopsing when we access
the debugfs entries. This is achieved by using snd_soc_read() and
snd_soc_write().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.
Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.
Reported-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.
Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).
Fix that by not restricting min_channels based on ELD information.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Jean-Yves Avenard <jyavenard@gmail.com>
Tested-by: Jean-Yves Avenard <jyavenard@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free the resources in the case of snd_card_register failure
and soc_register_ac97_dai_link failure.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free the resources in the case of soc_register_ac97_dai_link failure.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to mark this function as inline. Inline functions
usually are small and concise functions that benefit from not needing
to set up a stack frame and undergo a call/ret sequence upon each
invocation.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using strncpy() if the source string does not have a null byte in the
first n bytes, then the destination string is not null-terminated.
This can be fixed in a two-step process by manually null-terminating the
array after the use of strncpy() or by using strlcpy().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs"
added much of code that can be shared with DAI link codec probing/removal.
Merge now this common code into new soc_probe_codec, soc_remove_codec and
soc_post_component_init functions.
Error prints in these functions are converted to use dev_err and to print
the error code.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a patch to the sound/ac97_bus.c file that fixes up a 80 character
line limit issue found by the checkpatch.pl tool.
Signed-off-by: Jeffrin Jose <ahiliation@yahoo.co.in>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.
Remove redundant newline in source code.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixed intial return value to be a '0' as asuccess on
set_audio_clock_heirachy(). This avoids unintended error on initialize.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As jack detection can trigger DAPM and the latency in debouncing can create
confusing windows in operation provide some trace events which will hopefully
help in diagnostics. The soc-jack core traces all reports that it gets and
the resulting notifications to upper layers. An event for jack IRQs is also
provided for instrumentation of debounce, and used in the GPIO jack code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reformat and update the comments that describe the hardware connections
on the various models.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the hardcoded "CMI8788", show the actual chip name.
Note: This is neither what the chip is (it's always the same),
nor what the chip is actually called.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To help with debugging, add the registers of the model-specific
codecs to the controller and AC97 register dump in the proc file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "Front Panel" switch on the Xonar D1/DX actually switches only the
output direction, so mark it appropriately.
The front panel microphone is controlled by the FMIC2MIC bit of the
CM9780. It was unconditionally enabled on the D1/DX and never set on
the ST(X); add a control for it. Selecting the front panel microphone
as source does not actually disable the microphone jack, but this is
bug-compatible with the Windows driver, and users rely on it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO bit that enables analog output on the Xonar HDAV1.3 also
disables the HDMI audio output, so we better add a switch for it.
Hopefully, this is sufficient to make the HDMI output work.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the configuration of some unknown GPIO output bits (that
might not be used at all) to be the same as in the Windows driver, just
to be sure.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the comments of snd_soc_instantiate_cards() said,
snd_soc_instantiate_cards() must be called with client_mutex.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit.
Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: https://launchpad.net/bugs/685161
The reporter of the bug states that he must use position_fix=1 to enable
capture for the internal microphone, so set it for his machine's PCI
SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build.
Reported-and-tested-by: Ralph Wabel <rwabel@gmx.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx
HDMI), as the codec appears to work fine with it.
Note that the codec is still limited to stereo output only, despite it
reportedly being multichannel capable. Some as of yet unknown quirks
will be needed to get that working.
Testing was done on 2.6.36 by John Ettedgui.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: John Ettedgui <john.ettedgui@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13.
Reference: http://launchpad.net/bugs/685015
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
ALSA: hda: Use "alienware" model quirk for another SSID
ASoC: WM8731: Fix incorrect mask for bypass path disable
s6105-ipcam: fix compilation
s6000-pcm: fix compilation
s6000-i2s: fix compilation
ASoC: Fix missing spin_unlock_irqrestore
ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation
ASoC: Add missing dev_set_drvdata in p1022_ds_probe
ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
ASoC: Fix compile error for nuc900-pcm.c
ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
ASoC: Fix compile error for nuc900-ac97.c
ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
When we disable the WM8994 FLL code path sharing means that we end up
writing out a configuration. Currently this is the currently active
input and output frequency (which causes snd_soc_update_bits() to
suppress actual writes both immediately and in the common case where
we reenable the same configuration later) but we allow machine drivers
to pass through a source of zero. Since the register values written
are one less than the source constants this causes corruption of other
bitfields in the register.
Fix this by using the most recently configured FLL source when none is
provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
We want the index of DAI's driver here.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch allows machine drivers to override the compression type
provided by the codec driver.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary. This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The snd_soc_codec_conf struct now holds codec specific configuration
information.
A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.
In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead. In all other cases the one
set in the snd_soc_codec_conf struct takes effect.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with. The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.
Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Blind copy of codec finding algorithm from soc_bind_dai_link does not work
in soc_probe_aux_dev if matching codec name is not found. In that case the
code falls through and tries to start the probing procedure with invalid
codec pointer.
Fix this and add an error print showing the codec name that cannot be found.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In alc5623_i2c_probe(),
the default case for checking alc5623->id behaves the same as case 0x23.
However, In alc5623_probe() the default case for checking alc5623->id
becomes to be the same as case 0x21.
This makes the meaning of default case inconsistent.
Since we have checked codec id in alc5623_i2c_probe() by comparing
vid2 with id->driver_data, it is not possible to run into the default case now.
In case we may add more supported devices to alc5623_i2c_table in the future,
this patch changes the default case return -EINVAL to let people know that
they should not run into this case. They should also add a new case accordingly
for the new id.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was checking register range on fsi_master_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was checking register range on fsi_reg_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There was a strange part where fsi->xxx had been used
for fsi_master_xxx function instead of master->xxx in current FSI.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Silly little grammar nit but it bugs the hell out of me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of hard-coded magic numbers, properly define and use macros
for improve the readability. Also, dell_automute is handled samely
as thinkpad, since it also sets port_d_mode, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.
Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected. To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense. Another user corroborates this
workaround on separate hardware. Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.
Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update some of the default configuration for the device to improve
the performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The deemphasis filter should be selected based on sample rate for
optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is much more maintainable than the array.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The use of DAPM widgets, and extra routing can cause ordering
problems in the system.
Machine drivers should use the exported direct interface with
SND_SOC_DAPM_HP's event callback to manage the state of the
amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.
Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.
Thanks to Tony Lindgren <tony@atomide.com> for noticing.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work. Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.
Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute/unmute is controled by SMUTE (Soft Mute Control bit):
0: Normal Operation (Default)
1: DAC outputs soft-muted
I think this change improves readability.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
