* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: HDA: Do not announce false surround in Conexant auto
ALSA: HDA: Conexant auto: Handle multiple connections to ADC node
ALSA: HDA: Add position_fix quirk for an Asus device
ALSA: caiaq - Fix possible string-buffer overflow
ALSA: au88x0 - Modify pointer callback to give accurate playback position
Add " Playback Volume" to 10 bands Equalizer Controls of au88x0 so that
alsa-lib won't regard them as "Capture Volume".
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch, one line-out and one speaker and
Conexant's auto parser would announce (non-working) surround
capabilities.
BugLink: http://bugs.launchpad.net/bugs/721126
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant 20641 has several inputs to its ADC node, with one selector
and individual amps for all inputs. This patch adds support in the
Conexant auto parser to handle that case.
It also means that the pin node's volume is being renamed to "Boost"
to avoid name clash with the new volume controls on the ADC node.
BugLink: http://bugs.launchpad.net/bugs/719524
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make newly created AC97 emulation of azt3328 known to the AC97 layer
side.
- relocate common functions to the top (due to definition after use)
- rename control names
- adjust 3D settings to the card's custom layout of this register
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the very flexible ALSA ac97 layer (hooks for custom I/O!)
on this weird AC97 copycat hardware,
via semi-extended I/O translation/emulation.
Some 5kB binary/loaded size saved (well... additional huge AC97 module
penalty not factored in, of course ;-P).
Given that the driver previously had 20kB that's not bad,
but the much more important thing is to have AC97 layer stress-tested
with a thoroughly weird AC97 copycat (or, simply put, if it were not for
this AC97 test aspect, this effort would merely have been a nut job ;).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The microphone detection functionality requires a clock to work. In any
non-detection case where the MICBIAS is enabled CLK_SYS will be needed
anyway so there is no negative impact on power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The only thing that should ever be calling this is soc-core and that is
built as part of the same module so doesn't need the export.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pulseaudio doesnt work with current driver and it was root caused to absense of
hw_params function and malloc_pages in it.
This patch adds this and allows pa to work fine with these 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>
This patch removes the old method of soc-audio device creation in mfld machine
and makes use of new soc_register_card API to register the card
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 tlv scale is defined as (min, step, mute). The mute is not supported here so
put the value to 0
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 makes the DMIC dynamically connect to TX Mux, earlier code had
erroneously made this as static path
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 makes the codec playback rails (headset and speaker) depend on
actual pins they control. This enables better power management of the codec
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>
It is more usefull to report headset instead of video out cable in response
to jack insertion as this is more usual use-case and because now the headset
feature is supported. Automatic accessory detection is not possible at the
moment so most sensible static accessory type have to be used.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for headset microphone in Nokia RX-51/N900. The mic
signal from audio jack is routed to codec A LINE1L via two switches and the
mic bias is coming from codec B part.
First switch is the tv-out switch that is already supported and the second
switch selects between voltage detection circuit and codecs. As there is
no use for voltage detection at the moment the second switch is connected
statically to codecs in rx51_soc_init.
Headset can be active when control "Jack Function" is set to "Headset".
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Just slight cleanup to be sync with upcoming change.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The bug reporter claims that position_fix=1 is needed for his
microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40).
Reported-by: Kjell L.
BugLink: http://bugs.launchpad.net/bugs/718402
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use strlcpy() to assure not to overflow the string array sizes by
too long USB device name string.
Reported-by: Rafa <rafa@mwrinfosecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of cases has increased so use switch-case rather than
if-statements.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MK2 generation of Native Instruments' sound cards are in fact
compliant to the USB audio standard of version 2 and other approved USB
standards. However, they come up as vendor-specific device when first
connected but can be told to come up with a new set of descriptors
upon their next enumeration. The interfaces announced by the new
descriptors will be handled by the kernel's class drivers. This is done
by issuing a vendor specific device request and sending the device to
reset.
There are also some vendor-specific USB requests for some mixer elements
that can't be exported in a standard compliant way. The driver now
supports them with quirks handling mechanisms.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the core code where sparse complains. In most cases, this means
just adding annotations to confirm that we indeed want to do the dirty
things we're doing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8903 interrupts are clear on read so if the WM8903 detection is
enabled from platform data when the IRQ is in use (rather than using a
direct signal from a GPIO) status may be lost during startup. Help users
spot this misconfiguration by adding a WARN_ON().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
With the appropriate MODULE_ALIAS in place, the audio modules will be
automatically loaded; there is no longer a need for manual modprobes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Harmony has both an external mic (a regular mic jack) and an internal mic
(a 0.1" two-pin header on the board).
The external mic is connected to the WM8903's IN1L pin, and is supported
by the current driver.
The internal mic is connected to the WM8903's IN1R pin, and is not supported
by the current driver.
It appears that no Harmony systems were shipped with any internal mic
connected; users were expected to provide their own. This makes the
internal mic connection less interesting.
The WM8903's Mic Bias signal is used for both of these mics. For each mic,
a GPIO drives a transistor which gates whether the mic bias signal is
actively connected to that mic, or isolated from it.
The dual use of the mic bias for both mics makes a general-purpose complete
implementation of mic detection using the mic bias complex. So, for
simplicity, the internal mic is currently ignored by the driver.
This patch configures the relevant GPIOs to enable the mic bias connection
to the external mic, and disable the mic bias connection to the internal
mic. Note that in practice, this is the default state if these GPIOs aren't
configured.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Add jack definition for mic jack
* Request wm8903 to enable mic detection
* Force mic bias on, since it's required for mic detection
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a widget has been force enabled then not only do we need to keep the
widget itself enabled, we also need to keep any supplies the widget
requires enabled. The user could force all the individual widgets on but
this requires too much knowledge of device internals.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
On WM8994 revision D and earlier ensure proper playback robustness
as some rare use cases can trigger issues.
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
Ensure that on disabling certain registers such as AIF1DAC1L,
AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly
when enabling those registers, AIF1CLK and AIF2CLK will remain
disabled.
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
The mic detection HW should be enabled when either mic or short detection
is required, not when only both are required.
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>
Cc: stable@kernel.org
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662
ALSA: hrtimer: remove superfluous tasklet invocation
ALSA: hrtimer: handle delayed timer interrupts
ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G
ALSA: hda - Don't handle empty patch files
ALSA: hda - Fix missing CA initialization for HDMI/DP
ALSA: usbaudio - Enable the E-MU 0204 USB
ALSA: hda - switch lfe with side in mixer for 4930g
ASoC: Improve WM8994 digital power sequencing
ASoC: Create an AIF1ADCDAT signal widget to match AIF2
asoc: davinci: da830/omap-l137: correct cpu_dai_name
ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()
It is safe to use sleeping gpio in snd_soc_jack_gpio_detect as it is not
called from interrupt context. This avoids WARN_ON from __gpio_get_value
if sleeping gpio is registered for jack.
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 patch modifies the mfld_machine to use the new jack apis for adding the
voltage zones for jack type detection. It also modifed TI sn95031 codec driver
to use these new apis
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 makes no real difference compared to the write sequencer sequence
that was previously used but can run without a clock being provided.
Also remove the write sequencer support code as this was the last use
of it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The write sequencer sequencer sequence takes longer than is desirable
as it brings up a full playback path which is not required at this
point. Open coding the sequence cuts the startup time by two thirds.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Content of the CX20442's snd_soc_codec_driver.reg_cache_default pointed
area, introduced with my recent NULL pointer dereferece fix (commit
f019ee5feb), occured wrong after further
testing, more thorough than just booting successfully. There are two
problems with it:
1) It should read
(1 << CX20442_TELOUT) | (1 << CX20442_MIC),
not
CX20442_TELOUT | CX20442_MIC.
2) While correctly matching actual codec hardware state on boot when
fixed per 1), a few more code modifications would still be required
to reflect that state not only into register cache, but also force
them into DAPM pins state, otherwise an inconsitency occures which
may prevent further codec state changes from being applied correctly.
As a result, the phone stops ringing after reboot, until someone
picks up the handset for the first time.
Revert that reg_cache_default content to a working, previous de facto
default value of 0, in hope this change can still be accepted as an rc
cycle fix.
Created and tested against linux-2.6.38-rc4
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This netbook has a only one jack output and an internal mic.
By default, mic and jack sense aren't working. Using lenovo-101e
parameters makes both work.
The device seems based on a Sharetronic Q70, so this should fix audio for
this model too.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bb758e9637 removed snd_hrtimer_callback() from the hardware
interrupt handler, thus moving it into a tasklet, but did not tell the
ALSA timer framework about this, so the timer handling would now be done
in the ALSA timer tasklet scheduled from another tasklet.
To fix this, add the flag to tell the ALSA timer framework that the
timer handler is already being invoked in a tasklet.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a timer interrupt was delayed too much, hrtimer_forward_now() will
forward the timer expiry more than once. When this happens, the
additional number of elapsed ALSA timer ticks must be passed to
snd_timer_interrupt() to prevent the ALSA timer from falling behind.
This mostly fixes MIDI slowdown problems on highly-loaded systems with
badly behaved interrupt handlers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Difference in major.minor between driver and firmware is an error now.
Release version mismatch give a warning.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Interrupt flag used for message handshake will be required for
stream interrupts, so conditionally compiled code without
HPI6205_NO_HSR_POLL defined can never be used; removing it.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create and use HPI_ERROR_DSP_COMMUNICATION _DSP_BOOTLOAD, rather than
backend-specific error codes (now returned as data with the error).
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
asihpi.c don't link playback and capture streams, there is too much
offset between them.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some error codes had duplicate meanings. Just use one.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reported samples_played from card may be inaccurate, so don't use it.
Update control names to be closer to alsa standard practice.
Also fixed some accidentally lowercased strings.
[Removed adriver.h inclusion for external module builds by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove many unused functions.
Update some message and cache structs.
Use pci info directly from pci_dev.
Allow control cache elements with variable size, and handle
large message/response from dsp.
hpi6000 and hpi6205: fix error path when adapter bootload fails.
hpimsgx.c get rid of code duplicated in hpicmn.c
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Log HPI messages and responses in consistent numeric format,
which can be post-processed to get strings.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the reporter, node 0x15 needs to be muted for subwoofer
to stop sounding. This pin is marked as unused by BIOS, so fix that.
BugLink: http://bugs.launchpad.net/bugs/715877
Cc: stable@kernel.org (2.6.37+)
Reported-by: Hans Peter
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an empty string is passed to patch option, the driver should
ignore it. Otherwise it gets an error by trying to load it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explicitly cache the DC servo offsets for digital paths in the driver,
allowing them to be preserved over suspend and resume, and ensure that
we recalibrate analogue outputs paths when they are in use so that we
cover any changes in the input offset.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds soc-jack support for adding voltage zones and for
detecting jack type
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>
ASoC generally uses the register defaults for everything, but in some
cases the hardware will default to enabling some of the DAPM widgets
(clocks for example). Ensure that DAPM knows about the actual widget
state at initialisation by reading the enable bits after instantiating
the widgets so they don't get left enabled needlessly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8903 register map does not mute the DAC by default at startup
so we need to explicitly do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This simplfies the code and slightly reduces the startup time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
It causes noisy -codecs to appear in things like .codec_name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch modifies the Davinci i2s and mcasp drivers to make use of
ioremap() instead of IO_ADDRESS()
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case of any error in probe() function, clk_disable() and clk_put()
should be called if clk_enable() and clk_get() went through.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modifies the Davinci i2s and mcasp drivers
to make use of the resource_size() helper function for readability.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support to read the mic bias voltage
when a jack is inserted. It uses ADC to measure.
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 support for registering jack interupt
and registering jack with core
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 support for jack detection and reporting in the codec
It however is not fully functional as it doesn't measure adc to figure
out what got inserted which will 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>
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>
The commit 53d7d69d8f
ALSA: hdmi - support infoframe for DisplayPort
dropped the initialization of CA field accidentally.
This resulted in only two-channel LPCM mode on Nvidia machines.
Reference: kernel bug 28592
https://bugzilla.kernel.org/show_bug.cgi?id=28592
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The patch c358e640a6 "ASoC: soc-cache: Add trace event for
snd_soc_cache_sync()" introduced a dereference of "codec->cache_ops"
before we had checked it for NULL.
I pulled the check forward, and then pulled everything in an indent
level.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move Chip Select control out of the CODEC code for CS4271.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
rt->rate is an unsigned char so it's never equal to -1. It's not a huge
problem because the invalid rate is caught inside the call to
usb6fire_pcm_set_rate() which returns -EINVAL. But if we fix the test
then it prints out the correct error message so that's good.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AACI TRM requires the MAINCR enable bit to be held zero for two
bitclk cycles plus three apb_pclk cycles. Use a delay of 1us to
ensure this.
Ensure that writes to MAINCR to change the addressed codec only happen
when required, and that they take effect in a similar manner to the
above, otherwise we seem to occasionally have stuck slot busy bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Built-in sub-woofer can now be controlled by lfe slider instead of
side slider on Acer Aspire 5930g
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wm8753 codec supports switching between different DAI modes.
The current drivers tries to implement this by changing the DAI driver at
runtime. But to properly work this would require support from the ASoC core.
So this patch takes a different approch on how the DAI mode switching is
implemented.
The only difference, from a driver point of view, between the different DAI modes
is how to program the DAI format to the hardware. So what this patch is, it
stores the current format for each DAI in the drivers private struct and when
the DAI mode is changed the format gets simply reprogrammed according to the
new DAI mode.
Futhermore this patch restricts the changing of the DAI format to when the
codec is inactive.
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>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: use linux/io.h to fix compile warnings
ALSA: hda - Fix memory leaks in conexant jack arrays
ASoC: CX20442: fix NULL pointer dereference
ASoC: Amstrad Delta: fix const related build error
ALSA: oxygen: fix output routing on Xonar DG
sound: silent echo'ed messages in Makefile
ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()
ASoC: DaVinci: fix kernel panic due to uninitialized platform_data
ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
ASoC: Fix module refcount for auxiliary devices
ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF output
ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
On WM8994 revision D and earlier ensure optimal sequencing with
simultaneous usage of AIF1 and AIF2 by tying the signals together
so if paths through both are connected the streams are started
simultaneously.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
Due to the different routing for AIF1 and AIF2 we weren't using a
single widget to represent the ADCDAT signal. For consistency add
one.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
McASP1 is used on the DA830/OMAP-L137 platform for the codec.
This is different from the DA850/OMAP-L138 platform which uses McASP0.
This is fixed by adding a new snd_soc_dai_link struct.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The .card member of the snd_soc_pcm_runtime structure pointed to by the
snd_soc_dai_link.init() argument used to be initialized before the
function being called. This has changed, probably unintentionally,
after recent refactorings. Since the function implementations are free
to make use of this pointer, move its assignment back before the
function is called to avoid NULL pointer dereferences.
Created and tested on Amstrad Delta againts linux-2.6.38-rc2
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For some codecs with large register maps, it was not possible to dump
all registers via the codec_reg file but only up to PAGE_SIZE bytes.
This patch fixes this problem.
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>
Signed-off-by: Sven Neumann <s.neumann@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For helping to reduce Greert's regression list...
src/sound/drivers/mtpav.c: error: implicit declaration of function 'inb'
src/sound/drivers/mtpav.c: error: implicit declaration of function 'outb'
...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Conexant codec driver adds the jack arrays in init callback which
may be called also in each PM resume. This results in the addition of
new jack element at each time.
The fix is to check whether the requested jack is already present in
the array.
Reference: Novell bug 668929
https://bugzilla.novell.com/show_bug.cgi?id=668929
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the format of the codec_reg file more easily parsable. Remove
the header field which gives the codec name. These changes are important
when it comes to extend the debugfs codec_reg file to dump more than
PAGE_SIZE bytes to make it easier to calculate offsets within the
file.
We still need to handle the case when the snd_soc_read() call fails
and <no data: %d> is outputted.
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 CX20442 codec driver never provided the snd_soc_codec_driver's
.reg_cache_default member. With the latest ASoC framework changes, it
seems to be referred unconditionally, resulting in a NULL pointer
dereference if missing. Provide it.
Created and tested on Amstrad Delta against linux-2.6.38-rc2
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Amstrad Delta ASoC driver used to override the digital_mute()
callback, expected to be not provided by the on-board CX20442 CODEC
driver, with its own implementation. While this is still posssible when
substituting the whole empty snd_soc_dai_driver.ops member (the CX20442
case), replacing snd_soc_dai_ops.digital_mute only is no longer correct
after the snd_soc_dai_driver.ops member has been constified, and results
in build error.
Drop this actually not used code path in hope the CX20442 driver never
provides its own snd_soc_dai_ops structure.
Created and tested against linux-2.6.38-rc2
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO
driver has been fixed to work this way, so remove the redundant !!
previously required when it didn't accept values >1.
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>
Earphone in Nokia RX-51/N900 is connected to left HP output of B part of the
TLV320AIC34 dual codec. In RX-51 the codec A is used as a traditional codec
and the codec B as an auxiliary device.
Audio from codec A goes via the codec B to earphone:
MONO_LOUT of A -> LINE2R of B (B interconnects) -> HPLOUT of B -> Earphone.
Take earphone into use by utilizing the recent ASoC auxiliary and
cross-device support.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This driver does not use any of the functionality provided by the scoop
hardware. Remove the unneeded header.
Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.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>
Instead, have the machine driver provide storage for the utility data
somehow.
For Harmony in particular, store this within struct tegra_harmony, itself
referenced by snd_soc_card's drvdata.
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>
Indent with TABs not spaces.
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>
Add DAPM widget definitions for the internal speaker paths. Currently, this
path is always enabled while playback is active.
Add code to control the speaker amplifier GPIO.
The GPIO is requested during _init, since that's the first time it is
guaranteed that the WM8903 module is loaded, probed, and hence has exported
its GPIO chip.
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>
Previously, snd-soc-tegra-harmony internally instantiated a platform device
object whenever the module was loaded. Instead, switch to a more typical model
where arch/arm/mach-tegra defines a platform device, and snd-soc-tegra-harmony
acts as a driver for such a platform device.
Define a new struct tegra_harmony to store driver data in the future.
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>
All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was
only called for cards backed by a "soc-audio" platform device, via
soc_probe(). However, it's also needed for cards backed by other platform
devices, and registered directly via snd_soc_register_card().
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 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>
