Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.
As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a pure code motion patch intended to improve reviewability of a
following patch moving WM8731 to use more standard device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a bit more idiomatic and makes identifying a configuration
based on the board type work better.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have software control of the MCLK for the WM8731 so save a bit of
power by actively managing it within the machine driver, enabling it
only while the codec is active.
Once ASoC supports multiple boards and doesn't require the soc-audio
device the initial clock setup should be pushed down into the arch/arm
code but for now this reduces merge issues.
Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8731 bias level configuration function was written slightly
obscurely - streamline the code a little and refresh the comments.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
WM8753 uses a tricky way to switch DAIs "on the fly", for that it
registers 2 dummy DAIs and substitutes them depending on mixer control.
List element of registered dummy DAIs should be preserved to allow
unregistering of DAIs on module unload.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since cs4232 and cs4236 drivers are merged, there is no reason to keep
snd-cs4236-lib module separately. Let's merge it into the main driver
as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.
Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs4232 and cs4236 driver merge to solve PnP BIOS detection.
Also, the patch adds recognition if the chip is cs4236b+
or earlier part. This unifies drivers for both cs4232
and cs4236+ chips. It allows to use the PnP BIOS
detection for the cs4236+ chips. Previously, only
the snd-cs4232 could be detected by the PnP BIOS.
The cs4232+ cards reports two separate PnP BIOS ids.
The patch adds search for the second id to find out
resources assigned to a control port.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz
in its USB device descriptor. This patch extends an existing workaround
in usbaudio.c to also cover the CM6207.
This resolves issue 0004249 in the ALSA bug tracker.
Signed-off-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The detection of non-continuous rates (given via rate tables) isn't
processed properly (e.g. for type II).
This patch fixes and simplifies the detection code.
Tested-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the snd_usbmidi_create_endpoints_midiman() function, which forgot to
set the out_interval member of the endpoint info structure for Midiman/
M-Audio devices. Since kernel 2.6.24, any non-zero value makes the
driver use interrupt transfers instead of bulk transfers. With EHCI
controllers, these random interval values result in unbearably large
latencies for output MIDI transfers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: David <devurandom@foobox.com>
Tested-by: David <devurandom@foobox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force speaker pin config with model=hp-dv5 model for cases when bios
doesn't set it up properly. All reported hp laptops using model=hp-dv5
model have speaker at pin 0x0d with same config, so it's safe to add
this within hp-dv5 model.
Reference: alsa-devel mailing list thread on
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 32e176c14d.
That commit caused a regression with suspend on Thinkpad SL300.
Reference: kernel bug#12711
http://bugzilla.kernel.org/show_bug.cgi?id=12711
Tested-by: Alexandre Rostovtsev <tetromino@gmail.com>
Acked-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix for the error when the audio module is unloaded. On unregistering
the platform_device, platform_device_release will free the platform
data.If platform data is static the kernel panics when it is freed.
Instead use the platform device helper function to add data.
This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
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:
ASoC: Only register AC97 bus if it's not done already
ALSA: hda - Add snd_hda_multi_out_dig_cleanup()
ALSA: hda - Add missing terminator in slave dig-out array
ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model
ALSA: hda - Register (new) devices at reconfig
ALSA: mtpav - Fix initial value for input hwport
ALSA: hda - add id for Intel IbexPeak integrated HDMI codec
ALSA: hda - compute checksum in HDMI audio infoframe
ALSA: hda - enable HDMI audio pin out at module loading time
ALSA: hda - allow multi-channel HDMI audio playback when ELD is not present
ASoC: Update SDP3430 machine driver for snd_soc_card
ALSA: hda - Add quirk for Asus z37e (1043:8284)
sound: Remove OSSlib stuff from linux/soundcard.h
ASoC: WM8990: Fix kcontrol's private value use in put callback
ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
ASoC supports both explicit codec drivers for AC97 devices and a simple
driver which uses the standard ALSA AC97 framework for codec support.
When used with the generic AC97 codec support that will provide the
ad hoc AC97 device for drivers like touchscreens to attach to so the
core shouldn't do so.
Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the CS4270 codec driver to allow applications to use the mixer to
control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute.
Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first
initializes the hardware, but these features either don't work or interfere
with normal ALSA behavior. However, they can now be re-enabled by an
application if desired.
Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute
bits. The driver previously and erroneously assumed that these bits
control only external muting circuitry, but they also control internal
muting circuitry, so they should always be used.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the support of multiple digital outputs via auto-probing for
Realtek ALC88x codecs. The multiple outputs are handled as slave
streams, so only one PCM stream (and the corresponding IEC958*
elements) is provided to control both digital outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the helper function snd_hda_multi_out_dig_cleanup() to clean up
the digital outputs with multi setup. This call is needed in cases
the codec supports multiple digital outputs as slaves. Otherwise the
slave widgets aren't properly cleaned up.
For a single digital output (e.g. in patch_conexant.c), this call isn't
needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force model=auto for Acer AX1700-U3700A with ALC888 codec.
Since Acer devices are handlded as model=acer as default, the auto
parsing has to be specified explicitly.
(Maybe it's better rather to remove this default model=acer handling,
though.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change HP dv7 quirk: although reported to work with hp-m4 model
(https://bugzilla.novell.com/show_bug.cgi?id=445321), the original
report doesn't contain info about testing of internal microphone.
Recently I received a report about internal mic not working
(https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be
related with the forced line in on pin 0x0e done with hp-m4 model. Thus
change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported
to be fixed on a provided kernel with this change
(https://qa.mandriva.com/show_bug.cgi?id=44855#c196).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the HP connector on X200 dock doesn't detect when a HP is connected
nor allows sound to be played using it. This patch fixes the problem by adding
a quirk for this specific model. It's possible that others have the same NID
(0x19) to report when dock HP is connected, but I don't have access to any.
Please Cc me in the reply since I'm not subscribed to alsa-devel@.
Signed-off-by: Aristeu Rozanski <aris@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins
in pin_nids for stac92hd83xxx. Also reorganized connection selection
code for the respective ports per quirk define.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5F needs the fixed codec-slots to probe to override the BIOS
problem.
Tested-by: Giovanni Moser Frainer <giovanni@redix.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have broken BIOS and they don't set the codec probe-bit
properly after cleared by the driver. This makes the driver skipping
the necessary codec slots.
Since BIOS update isn't always easy, now the semantics of probe_mask
option is changed a bit. When it contains the bit 8 (0x100), the
lower bits are used to probe that slots regardless of codec-probe bits
returned by the hardware.
For example, probe_mask=0x103 will force to probe the codec slot #0
and #1.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After commit "ALSA: hda - Fix restore of pin configs at resume for
STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks
already for pins == NULL case, saving then default pin configs from
machine with stac92xx_save_bios_config_regs. So we can remove the
extra checks when stac927x_brd_tbl[spec->board_config] == NULL.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices that have been newly added during reconfig must be
registered. Otherwise they won't be visible to user-space.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial value for input hwport. The old value (-1) may cause
Oops when an realtime MIDI byte is received before the input port is
explicitly given.
Instead, now it's set to the broadcasting as default.
Tested-by: Holger Dehnhardt <dehnhardt@ahdehnhardt.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect multiple digital-out pins in snd_hda_parse_pin_defconfig().
The dig_out_pin and dig_out_type fields become arrays.
The codec parser still doesn't use this multiple pins detection, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found that enabling/disabling HDMI audio pin out at stream start/stop
time will kill the leading 500ms or so sound samples. Avoid this by enabling
pin out once and for ever at module loading time.
The leading ~500ms audio samples will still be lost when switching from
X-channel playback to Y-channel playback where X != Y. However there's no
much we can do about it: the audio infoframe has to change and it looks like
either G45 or YAMAHA requires some time to switch the configuration.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel
mapping to play >2 channel HDMI audio. In theory that mapping should be
derived from its speaker configurations contained in its ELD. However we
currently cannot get ELD in console before the KMS functionalities are ready.
This is a more or less general issue at least in the near future. As a
workaround, we propose to allow playback of mult-channel audio when ELD
is not available.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch replaces "snd_soc_machine" structure by "snd_soc_card" in
SP3430 driver. This change is needed in SDP3430 driver to reflect
changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch
(875065491f).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the digital beep support for ALC268. It was missing in the
last patches.
However, ALC268 has a strange pin use for widget 0x1d, which could be
used as another purpose. So, the patch adds a check of the beep control
before creating the hook for input layer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix decrement the timeout will reach -1 rather than 0,
so the warning will not be issued.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
TLV320AIC3X volume controls are logarithmic. Export their dB ranges.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a minor fix but helps to define dB ranges for volume controls.
Only DAC digital volume has full register value range from 0 to 127 but
ADC PGA gain and output stage volume controls don't.
For ADC PGA, maximum value is 119 and then it saturates to the same
gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB
and is muted for values 118 and above.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking for input amps on pins 0x0a, 0x0d and 0x0f, and
initializing them for 92hd71xxx codec models, we must skip nid 0x0f
for 4-port models too like with 5-port models, as it is unused
(nid 0x0f is vendor reserved in 4-port models).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a new sound quirk entry (model=ecs202) for an ecs motherboard
with IDT STAC9221 codec (1019:2950).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver.
Pin setup should be done during board init via pxa2xx_mfp_config
instead.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This machine driver enables sound functions on Mitac mio
a701 smartphone. Build upon ASoC v1, it handles :
- rear speaker
- front speaker
- microphone
- GSM
A global "Mio Mode" switch is not yet provided to cope with
audio path setup. As balance on audio chip line is no more
assured, an incorrect setup can produce a lot of heat and
even fry the battery behind the wm9713 and the speaker
amplifier.
It doesn't cope with :
- headset jack
- mio master mode
- master volume control
This driver is backported from ASoc v2, and amputated from
scenario setups and master volume control.
[Minor mods for terminology in comments -- broonie]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use digital beep instead of analog pc-beep for AD codecs.
Create the beep mixer controls dynamically on demand.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the Freescale MPC8610 sound drivers, relocate all code from the _prepare
functions into the corresponding _hw_params functions. These drivers assumed
that the sample size is known in the _prepare function and not in the
_hw_params function, but this is not true.
Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create
fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it.
Turn off snooping for DMA operations to/from I/O registers, since that's not
necessary.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- make sport number handling more dynamic as not all
Blackfins have a linear sport map starting at 0
- indexes can be macroed away too
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc.
This is very similar fix than fix to TLV320AIC3X codec made by
Eero Nurkkala <ext-eero.nurkkala@nokia.com>. This fix is compile tested
only.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc.
This was causing arbitrary register writes when touching the controls
defined with SOC_DAPM_SINGLE_AIC3X.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For audio devices that do not have proper audio descriptors (e.g.,
Edirol UA-20), we use hardcoded parameters from our quirks list.
However, we must still read the maximum packet size from the standard
endpoint descriptor; otherwise, we might use packets that are too big
and therefore rejected by the USB core.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the SPDIF pin as slave digital out to enable concurrent
HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec.
Tested-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the AC97 controller found on Atmel
AVR32 devices.
Tested on ATSTK1006 + ATSTK1000 with a development board with a AC97
codec.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the Audio Bistream DAC found on Atmel
AVR32 devices. The ABDAC is an Atmel IP which might show up on AT91
devices in the future, hence making a generic driver which can be
utilized by AT91 arch if needed.
Datasheet describing the ABDAC peripheral is available in the AT32AP7000
datasheet, http://www.atmel.com/dyn/products/datasheets.asp?family_id=682
Tested on ATSTK1006 + ATSTK1000 with a class D amplifier stage.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: cleanup
snd_pcm_new takes a char *id argument, although it is not modifying
the string. it can therefore be declared as const char *id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Indicates fixes affecting control messages and switching of input mode
on Audio 8 DJ and Audio 4 DJ.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove a duplicate control which causes an error when it is registered,
and causes later controls to not be registered. The device does not have
a fourth ground lift control.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the context of the Audio 4 DJ (when compared to Audio 8 DJ), hardware
input mode 0 is not used. Expose modes 1 (line) and 2 (phono) to the user
as modes 0 and 1 respectively.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not start the device with input mode undefined. Mimic the behaviour of
the Audio 8 DJ and start in phono input mode.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes a bug where an incorrect command was sent which had no effect on the
device.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the ADATOut nid to a slave digital outs struct to allow output
via the DigOut pin.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config
in patch_stac92hd71bxx (no SPDIF in).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect the number of connected ports and number of smuxes dynamically,
looking at pin configs, using new introduced functions
stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also
use proper input mux configuration for 4port and 5port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin
complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f
are vendor reserved. This commit changes code so it'll consider the
additional pin complexes for models that have it, and avoid reserved
nids to be touched on 4 port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fixes regarding snd-hda-intel workqueue:
- Use create_singlethread_workqueue() instead of create_workqueue()
as per-CPU work isn't required.
- Allocate workq name string properly
- Renamed the workq name to "hd-audio*" to be more obvious.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AFMT_S24_LE is set twice in return value
vi sound/core/oss/pcm_oss.c +640
#define AFMT_S24_LE 0x00008000
#define AFMT_S24_BE 0x00010000
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wss_base is disuised parameter for one function.
It is converted to function parameter.
The code_type is only set but never read.
It is removed.
The midi_vol is set only to 0 so it does not work
as detection of change in midi volume. It is fixed.
The xport variable is alias to the port[dev]. Use
the port[dev] directly to increase readability.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A digital beep generator can be used via input layer.
Signed-off-by: Kusanagi Kouichi <slash@ma.neweb.ne.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function
requested the wrong SSP port. Correcting this unveiled another bug
where ssp_init tries to request the already-requested SSP port again.
So this patch replaces the ssp_init/exit calls with their internals
from mach-pxa/ssp.c, leaving out the redundant ssp_request and the
unneeded IRQ request. Effectively, that leaves us with not much more
than enabling/disabling the SSP clock.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch splits set_dai_fmt into three variants (single interface,
dual interface playback only, dual interface capture only) so that
data input and output formats can be configured separately for dual
interface setups.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this fix driver switches to WSPLL in uda1380_pcm_prepare
even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK
register to disable R00_DAC_CLK before flushing reg cache)
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a
rather odd configuration template. As a result, the white AUX connector
can not be used. This rewrites the default config to more accurately
reflect the connector layout, colour and function.
Unfortunately the black CD_IN connector, which is suspected to be widget
0x1c refuses to switch into input (0x20), instead opting to remain on 0x0.
As such, no mixer controls are exposed for it. Autoswitching is implemented
between the front headphone output and back line output.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This just updates my email address on some drivers I'd forgotten in a
previous patch.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace printk calls with dev_xxx calls. Set the 'dev' field of the codec
and codec_dai structures so that these calls work.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a oversight in the CS4270 codec driver that caused a build break.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MIDI_TX IRQ seems always pending when any bytes on FIFO is available.
Thus, it's better to enable MPU_TX only when any bytres are really
stored in the substream, and disables immediately when the queue
becomes empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
omap_pcm_trigger is called also in interrupt context so CPU flags must
be restored when returning.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Volume-knob widgets have no widget selection although they have widget
connections. Thus, the connection list in the proc output shouldn't
contain the selection (*).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The missing module license generates warning
during module loading.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix decrement time will reach -1 rather than 0,
so the warning will not be issued.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update pandora board file for recent TWL4030 codec changes.
Also move output related snd_soc_dapm_nc_pin() calls to
omap3pandora_out_init(), where they belong.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Spruce up the documentation in the CS4270 codec. Use kerneldoc where
appropriate. Fix incorrect comments.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ess1688 driver uses the same port
for PCM audio (SB compatible) and OPL3
synthesis. It is not always right so allow to
choose a different port for OPL3 synthesis.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself.
When a codec driver registers with ASoC, a probe function is called. Most
codec drivers call ASoC first, and then register with the I2C bus in the ASoC
probe function.
However, in order to support multiple codecs on one board, it's easier if the
codec driver is probed via the I2C bus first. This is because the call to
i2c_add_driver() can result in the I2C probe function being called multiple
times - once for each codec. In the current design, the driver registers
once with ASoC, and in the ASoC probe function, it calls i2c_add_driver().
The results in the I2C probe function being called multiple times before the
driver can register with ASoC again.
The new design has the driver call i2c_add_driver() first. In the I2C probe
function, the driver registers with ASoC. This allows the ASoC probe function
to be called once per I2C device.
Also add code to check if the I2C probe function is called more than once,
since that is not supported with the current ASoC design.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
X86_PC is the only remaining 'sub' architecture, so we dont need
it anymore.
This also cleans up a few spurious references to X86_PC in the
driver space - those certainly should be X86.
Signed-off-by: Yinghai Lu <yinghai@kernel.org>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
In the case of having a selector instead of mixer while initing input
sources, the case that happens with ALC889, we must select instead
of muting input. Problem was found while testing with hda-emu.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the analog loopback/bypass support for twl4030 codec.
Details for the implementation:
It seams that the analog loopback needs the DAC powered on on the channel,
where the loopback is selected. The switch for the DACs has been moved from
the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be
powered while the audio playback is used or/and the loopback is enabled for
the channel.
The twl4030 codec powering has been reworked. Now the codec will be powered as
long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are
when the given change in the registers needs the codec power down/up cycle in
order to take effect. Otherwise the codec is on.
When the codec enters to STANDBY state and none of the loopback paths are
enabled, than the amplifiers, which are no in the DAPM path are forced to turn
off and the PLL is disabled. When playback/capture starts the disabled gains
are restored and the PLL is enabled.
When one of the loopback enabled in STANDBY mode, the disabled gains are
restored and the PLL is enabled also.
In short: the codec always goes to the lowest power state based on the
bias_level and the bypass_state.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch explicitly initializes McBSP Transmit Configuration
Control Register (XCCR) and Receive Configuration Control
Register (RCCR) to their reset values. Reset values are 26 ns
of DX delay and Transmit DMA disabled for XCCR register;
receive full cycle mode enabled and Receive DMA disabled for
RCCR register.
This patch requires a counterpart in OMAP McBSP driver before
to apply it. The required changes in McBSP were sent and approved
in linux-omap mailing list and patch is going upstream
(commit 3127f8f859 from linux-omap-2.6
tree).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
[ jarkko.nikula@nokia.com: Commit id for counterpart patch corrected ]
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8753 driver multiplexes the DAI structures it exposes to the
outside world, leaving them uninitialised until the codec probes. Since
the DAI name is used during the registration and setup process provide a
dummy name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While debugging some code paths in AC97 codec patches and its suspend
and resume functions, getting to know the flags has proved useful to
follow those code paths.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently code for capture source support in ALC889 only considers
capture mixers. This change adds additional support for ADC+selector
present in ALC889, taking into account also the presence of an
additional DMIC connection item in the selector.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check in a quirk list if it should do cold reset when AC97 power saving
is enabled. Some devices do not resume properly when cold reset,
although power saving works OK.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another MacBook Pro 4,1 SSID (106b:3800). It seems that latter revisions,
(at least mine), have different IDs to earlier revisions.
Signed-off-by: Luke Yelavich <themuso@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is driver for Turtle Beach Multisound cards:
Classic, Fiji and Pinnacle.
Tested pcm playback and recording and MIDI playback
on Multisound Pinnacle.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the twl4030_power_up and twl4030_power_down function
earlier to facilitate the analog bypass implementation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the power switches for the physical ADC and for the
amplifiers for the analog capture path to map more closely
the actual path inside of the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Headset Left anti-pop and bias ramp does not need to be
enabled, if the headset is not in use.
Move the code to DAPM event handler for HeadsetL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge the codec up and down functions to a simple one.
Codec is powered down by default (reg_cache change).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The offset cancelation bit in ANAMICL register is self cleanig.
Make sure that the reg_cache holds the same value as the HW
register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather use printf format instead of hardcoding prefix like 0x.
A next step would be to predefine certain formats.
Signed-off-by: Andreas Bergmeier <lcid-fire@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Distribute the packets evenly among the URBs, instead of making all URBs
except the last one to have the maximum size. This makes the timing of
pointer updates more regular and removes some special cases from the
code.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Once our URBs contain enough packets, queueing more URBs does not give
us any additional underrun protection (as we use double-buffering) but
just increases latency unnecessarily. Therefore, we try to limit the
queue length to some reasonable value.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When sending a silence URB (before playback has started, or when it is
paused), use the number of frames that would be normally sent instead of
a single frame so that the rate at which completion interrupts arrive is
consistent.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some revisions of the 92hd71xxx codec families don't have input amps
on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and
check for support, if found run snd_hda_sequence_write_cache() on the
stac92hd71xxx_unmute_core_init verb list.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further improvements in the I2C initialization sequence of the CS4270 driver.
All ASoC initialization is now done in the I2C probe function.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensures that the DAI and socdev exported by the codec match up with
their exported prototype.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kbuild ignores dependency from things that are themselves selected so
ASoC machine drivers need to ensure that the control bus is being built.
This also avoids issues where multiple buses are supported by a given
codec.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- add support for gpio ports (2 GPI, 2 GPO) of pcxhr stereo cards
- minor bugfixes : allow setting clock to internal by the mixer
even if there is no stream (but monitoring)
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pxa-regs.h and hardware.h are not intended for use directly in driver
code and references to them have been removed in other code - remove
them from the newly added e740 and e750 machine drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mic-boosts (0x14 and 0x15) on AD1884* codecs are input-amps,
not output-amps. Fix the invalid initialization verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a merge issue caused by context overlap.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4270 supports stand-alone mode, where the codec is not connect to the
I2C or SPI buses. Instead, input voltages configure the codec at power-on.
The CS4270 ASoC device driver has partial support for this mode, but the
code was never tested, and partial support doesn't help anyone. It also made
the rest of the code more complicated than necessary.
[Removed redundant CS4270 dependency on I2C -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of fixed kcontrol_new element, build "Input Source" controls
dynamically. If the number of input-source items is 0 or 1, we don't
need to create such a control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reference NID for the analog outputs of STAC/IDT codecs is set
to a fixed number 0x02. But this isn't always correct and in many
codecs it points to a non-existing NID.
This patch fixes the initialization of the PCM reference NID taken
from the actually probed DAC list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another LanParty reference board SND_PCI_QUIRK to quirk lists of
all codec families.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some revisions of the 92hd8xxx codec's not supporting port power
downs in which the using of it causes capture and also randomly
playback streams to not function at all. Thus by disabling it by
default and adding a option to enable it manually will fix all issue
on current and future revisions.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Port 0xe power mapping was incorrect set to 0x80 changed to the correct
value 0x40.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gateway T1616 laptop needs EAPD always on while the current STAC9205
code turns off per HP plug. Added a new model "eapd" to keep it on.
Reference: Novell bnc#467597
https://bugzilla.novell.com/show_bug.cgi?id=467597
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the present pin control bit and avoid the write if it's already
set in patch_sigmatel.c. This will reduce the number of verb execs at
jack plugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most cards have OPL3 FM synthetiser but
they do not have OPL3 interrupt wired to
a sound chip or CPU.
Do not create OPL3 timers for such cards
as the timers are useless witthout interrupt.
This patch removes OPL3 timers for following
alsa drivers: snd-ad1816a, snd-opti93x,
snd-opti92x, snd-sc6000, snd-cmi8330.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dc06102a0c in the asoc tree
did not include the necessary Kconfig and Makefile changes. This patch
completes the support for Beagleboard
Signed-off-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add OPL3 handling to the driver
and volume control for FM synthesis.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the 92hd8xxx codecs port 0xe needs the connection selected to be the
last DAC in the list.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the acpture switch name so that it better reflects its
purpose.
Signed-off-by: Ian Molton <iann@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the Kconfig and Makefile options for Freescale MPC8610 audio drivers
so that they can be compiled as modules, and simplify the Kconfig choices
so that only the platform is selected.
Also fix the naming of the driver files to conform to ALSA standards.
[Removed extraneous SND_SOC dependency -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Freescale MPC8610 driver was defining two SOC card (snd_soc_card)
structures, partially initializing each one, but registering only one of
them with ASoC.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some laptops like VAIO have multiple codecs and uses ALC262 only for
the SPIDF output without analog I/O. So far, the codec-parser assumes
the presence of analog I/O and returned an error for such a case.
This patch adds some hacks to allow the digital-only configuration for
ALC262.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assign the proper PCM digital I/O type (HDA_PCM_TYPE_*) for the digital
I/O on STAC/IDT codecs. HDA_PCM_TYPE_HDMI is assigned for the HDMI I/O.
A similar framework is implemented to patch_realtek.c, but it's not
set up and still using only HDA_PCM_TYPE_SPDIF yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add dig_out_type and dig_in_type fields to autocfg struct.
A proper HDA_PCM_TYPE_* value is assigned to these fields according
to the pin-jack location type value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't break the loop in snd_hda_codec_build_pcms() even if the item
has no substreams.
It's possible that it's an empty item and the next item containing
the valid substreams (e.g. realtek codecs may create the analog
and alt-analog but no digitl streams).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT codecs have often too large volume scales such as -96dB,
and exposing this as is results in too large scale in percentage
representation.
This patch adds the check of the volume scale and halves the
volume range if it's too large automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the volume offset to base for the standard volume controls
to handle elements with too big volume scales like -96dB..0dB.
For such elements, you can set the base volume to reduce the range.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM operations tables are not exported directly but are instead
included in the platform structure so should be declared static.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch takes fixes a number of bugs in the caching code used by
several ASoC codec drivers. Mostly off-by-one fixes.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740.
Note:
The e740 has a hard headphone switch that turns the speaker off and is not
software detectable or controlable. Also both headphone and speaker amps
share a common output enable.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Zylonite supports switching the MCLK for the WM9713 between the
AC97CLK and CLK_POUT outputs of the PXA processor via switch SW15 on
the board. This patch adds support for configuring the system to use
CLK_POUT.
Unfortunately it is not possible to read the state of SW15 from software
so this feature is controlled by a module option 'clk_pout' which should
be set to a non-zero value to enable the use of CLK_POUT.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mention in the Kconfig help text that the HDAV1.3 code is rather
experimental.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This hardware has a better chance of working correctly if we don't
forget to enable it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore errors (wrong usb interface data) found when using the serato
scratch live box with alsa
Thus the alsa controls can be accessed (beware: they don't work though -
but at least it's one ugly error message less)
Signed-off-by: Andreas Bergmeier <lcid-fire@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed invalid power mappings for ports 0xd and 0xe on 93hd83xxx codecs.
They were shifted right one too many bits.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A WSS codec is autocalibrated each time before
playing sound. Do only one calibration during
codec initialization.
Complete snd_wss_calibrate_mute to mute loopback
volume as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM9713 driver does not support configuring the PLL output frequency
so the output frequency parameter is irrelevant. Allow users to set it
to zero by ignoring it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the wm9712 ac97 codec as used in the Toshiba e800
PDA. It includes support for powering up / down the external headphone and
speaker amplifiers on this machine.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the wm9705 ac97 codec as used in the Toshiba e750
PDA. It includes support for powering up / down the external headphone and
speaker amplifiers on this machine.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver adds support for the wm9705 ac97 codec. The driver supports
audio input and output.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
<mach/hardware.h> doesn't exist on AVR32 and therefore this driver won't
build on that arch. AFAICT this driver doesn't actually use the content
of that header so easiest just to remove it.
Signed-off-by: Ben Nizette <bn@niasdigital.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Added support for two new audio devices from Native Instuments,
'Audio4DJ' and 'GuitarRig mobile'
- Add missing statement about 'Session IO' in Kconfig help text
- Version number bumped to 1.3.11
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the 82801H variants 1071:8227 and 8086:2503 to use ALC883_MITAC
Reference: Ubuntu bug 210865
https://bugs.launchpad.net/bugs/210865
Signed-off-by: Luke Yelavich <themuso@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove dependency on sffsdr_fpga_set_codec_fs() when the
SFFSDR FPGA module is not selected.
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some machines have a master amp GPIO that needs to be toggled to
get sound output, in addition to speaker/headphone/line-out amps.
This makes snd-aoa handle it, if present in the device tree, thus
making snd-aoa be able to output sound on PowerMac3,6, which was
previously handled by snd-powermac which also doesn't use the
master amp GPIO.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes snd-aoa to handle some older machines that are
currently handled by snd-powermac. snd-aoa has a number of advantages
though, notably it can autoload better and is generally a more modern
driver.
By hardcoding the accepted device-ids (last hunk of the patch) I'm
trying to avoid regressions because this driver will otherwise load
automatically and not let snd-powermac load. People who are unhappy
with snd-powermac and have a device-id property in the device tree
are encouraged to read this patch and make a patch to amend this as
appropriate.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec-parsing of STAC925x was utterly broken due to its unique
design unlike other STAC codecs. It has a volume control only in NID
0x0e (similar as STAC9200), but the parser assumes that the amp is
available on each DAC widget.
The patch fixes the whole wrong stories: fix the initial volume,
assign the fixed "Master" volume, and avoid to create wrong volume
controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value set in the commit 2465fb6605
is actually wrong. The value range is from 0 to 0x1f while the patch
sets to 0x7f. Let's fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify the check for the mux type to also handle the
snd_soc_dapm_value_mux type in a same way as the snd_soc_dapm_mux.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Asus Xonar D2 and D2X models, the SPI chip select signal for the
fourth DAC shares its pin with the serial clock for the EEPROM that
contains the PCI subdevice ID values. It appears that when DAC
registers are written and some other unknown conditions occur (probably
noise on the EEPROM's chip select line), the EEPROM gets overwritten
with garbage, which makes it impossible to properly detect the card
later.
Therefore, we better avoid DAC register writes and make sure that the
driver works with the DAC's registers' default values. Consequently,
the sample format is now I2S instead of left-justified (no user-visible
change), and the DAC's volume/mute registers cannot be used anymore
(volume changes are now done by the software volume plugin).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The selector widget 0x0e isn't initialized properly in the whole probe
process, thus it can be a wrong value depending on the BIOS setup.
This patch adds the init verb to set it to the max & unmuted.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Logitech QuickCam Pro 9000 does not appear to any product identification
strings in its USB device descriptor. Therefore it receives a device name of
"USB Device 0x46d:0x990". Th e attached patch below adds a USB quirk to
provide a more friendly name.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the bug introduced in commit c15c5060fc:
sound/pci/hda/patch_sigmatel.c: In function ‘patch_stac92hd83xxx’:
sound/pci/hda/patch_sigmatel.c:4765: warning: assignment from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Resetting HP pinctl at the unplugged state may cause a sort of regression
on some devices because of their wrong pin configuration.
A simple workaround is to disable the pin reset. This is ugly and may be
not good from the power-saving POV (if any), but damn simple.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: stable@kernel.org
Added amp nid for stac92hd8xxx families of codecs so the input amp
mixer is created.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CF-74 does the headphone/speaker switching on hardware, thus the driver
shouldn't do any software-toggling of pins. Otherwise it results in a
silent headphone output.
This patch simply resets the hp_detect flag to fix the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A comment states that one should wait up to 5 secs
while a waiting loop waits only 5 system ticks.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers in soudn/usb/* don't handle the error code properly
from snd_card_create(). This patch fixes these places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers in sound/isa/* don't handle the error code properly
from snd_card_create(). This patch fixes these places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduced snd_card_create() function as a replacement of snd_card_new().
The new function returns a negative error code so that the probe callback
can return the proper error code, while snd_card_new() can give only NULL
check.
The old snd_card_new() is still provided as an inline function but with
__deprecated attribute. It'll be removed soon later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-hda-intel driver used schedule_work() fot the delayed DMA pointer
updates, but this has several potential problems:
- it may block other eventsd works longer
- it may deadlock when probing fails and flush_scheduled_work() is
called during probe callback (as probe callback itself could be
invoked from eventd)
This patch adds an own workq for each driver instance to solve these
problems.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Intel DX58SO board works fine with model ALC883_3ST_6ch_INTEL.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is required to allow the sscape driver
to autodetect installed codec.
Also, do not create a timer if detected codec
has no hardware timer (e.g. AD1848).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Cc: Rene Herman
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call the snd_soc_free_pcm and snd_soc_dapm_free when the
codec driver is unloaded.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.
This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.
[Edited to raise priority of error log message and document parameters.
-- broonie]
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a jack reporting interface to ASoC. This wraps the ALSA
core jack detection functionality and provides integration with DAPM to
automatically update the power state of pins based on the jack state.
Since embedded platforms can have multiple detecton methods used for a
single jack (eg, separate microphone and headphone detection) the report
function allows specification of which bits are being updated on a given
report.
The expected usage is that machine drivers will create jack objects and
then configure jack detection methods to update that jack.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following lockdep warning by not reusing the hda_codec.spdif_mutex.
ALSA sound/pci/hda/hda_codec.c:882: hda_codec_cleanup_stream: NID=0x2
=======================================================
[ INFO: possible circular locking dependency detected ]
2.6.28-next-20090102 #33
-------------------------------------------------------
mplayer/3151 is trying to acquire lock:
(&pcm->open_mutex){--..}, at: [<ffffffffa004ced3>] snd_pcm_release+0x43/0xd0 [snd_pcm]
but task is already holding lock:
(&mm->mmap_sem){----}, at: [<ffffffff810c0252>] sys_munmap+0x42/0x80
which lock already depends on the new lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd-usb-caiaq driver causes Oops occasionally when accessing MIDI
devices. This patch fixes the Oops and invalid URB submission errors
as well.
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The soc_value_enum has been merged to soc_enum.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge the recently introduced soc_value_enum structure to the soc_enum.
The value based enums are still handled separately from the normal enum types,
but with the merge some of the newly introduced functions can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a patch which adds correct auto detection of model for
snd-hda-intel for Acer Aspire 5930G and 6930G. Tested on my 5930G. It
finally adds hp jack sense and 5.1 speaker system sliders.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-2.6:
sparc64: Work around branch tracer warning.
sparc64: Fix unsigned long long warnings in drivers.
sparc64: Use unsigned long long for u64.
sparc: refactor code in fault_32.c
sparc64: refactor code in init_64.c
sparc64: refactor code in viohs.c
sparc: make proces_ver_nack a bit more readable
The Toshiba Multimedia Center SB-0500 is a rebranded version of the
Creative Technology SB Live! 24-bit External: it shares the same chipset
and only has minor cosmetic differences. Remote controller works with
alsa_usb module, basic audio is there and mixer controls are mostly
untested.
Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch allows you to define the mixer paths as having the same name as the
paths they represent.
This is required to support codecs such as the wm9705 neatly without extra
controls in the alsa mixer.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (24 commits)
trivial: chack -> check typo fix in main Makefile
trivial: Add a space (and a comma) to a printk in 8250 driver
trivial: Fix misspelling of "firmware" in docs for ncr53c8xx/sym53c8xx
trivial: Fix misspelling of "firmware" in powerpc Makefile
trivial: Fix misspelling of "firmware" in usb.c
trivial: Fix misspelling of "firmware" in qla1280.c
trivial: Fix misspelling of "firmware" in a100u2w.c
trivial: Fix misspelling of "firmware" in megaraid.c
trivial: Fix misspelling of "firmware" in ql4_mbx.c
trivial: Fix misspelling of "firmware" in acpi_memhotplug.c
trivial: Fix misspelling of "firmware" in ipw2100.c
trivial: Fix misspelling of "firmware" in atmel.c
trivial: Fix misspelled firmware in Kconfig
trivial: fix an -> a typos in documentation and comments
trivial: fix then -> than typos in comments and documentation
trivial: update Jesper Juhl CREDITS entry with new email
trivial: fix singal -> signal typo
trivial: Fix incorrect use of "loose" in event.c
trivial: printk: fix indentation of new_text_line declaration
trivial: rtc-stk17ta8: fix sparse warning
...
Add support for reporting new jack types SND_JACK_VIDEOOUT and
SND_JACK_AVOUT (a combination of LINEOUT and VIDEOOUT) to the jack
reporting API.
Also add the corresponding SW_VIDEOOUT_INSERT switch to the input system
header.
Signed-off-by: Jani Nikula <ext-jani.1.nikula@nokia.com>
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:
ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses
ASoC: Use snd_soc_dapm_nc_pin() in at91sam9g20ek
ASoC: TWL4030: Convert the bitfield enums to VALUE_ENUM type
ASoC: New enum type: value_enum
pxa2xx-ac97: switch AC unit to correct state before probing
ASoC: Clocking fixes for davinci-evm.c
ASoC: Mark non-connected TWL4030 pins for pandora
ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
Several subsystem open handlers dereference the fops_get() return value
without checking it for nullness. This opens a race condition between the
open handler and module unloading.
A module can be marked as being unloaded (MODULE_STATE_GOING) before its
exit function is called and gets the chance to unregister the driver.
During that window open handlers can still be called, and fops_get() will
fail in try_module_get() and return a NULL pointer.
This change checks the fops_get() return value and returns -ENODEV if NULL.
Reported-by: Alan Jenkins <alan-jenkins@tuffmail.co.uk>
Signed-off-by: Laurent Pinchart <laurent.pinchart@skynet.be>
Acked-by: Takashi Iwai <tiwai@suse.de>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Dave Airlie <airlied@linux.ie>
Acked-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix warnings caused by the unsigned long long usage in sparc
specific drivers.
The drivers were considered sparc specific more or less from the
filename alone.
Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
For codecs that have both SPI and I2C support we need to ensure that we
don't try to make the codec driver built in when I2C is modular since
that won't link. Do this by creating a helper variable which uses
conditional defaults to pick up the correct value for all combinations.
We don't need to do anything special for I2C-only codecs since a
conditional select passes on the full value for a tristate.
Reported-by: Ingo Molnar <mingo@elte.hu>
Tested-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is always "an" if there is a vowel _spoken_ (not written).
So it is:
"an hour" (spoken vowel)
but
"a uniform" (spoken 'j')
Signed-off-by: Frederik Schwarzer <schwarzerf@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Convert the bitfield coded enums to the new VALUE_ENUM type.
Remove the enum check, since the VALUE_ENUM type can handle
the bitfield coding and also handles the 'holes' in the bitfield.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces a new enum type.
In this enum type each enumerated items referred with a value.
This new enum type can handle enums encoded in bitfield, or any other
weird ways. twl4030 codec has several mux selection register, where the
input/output mux is coded in a bitfield. With the normal enum type this type
of mux can not be handled in a clean way.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If AC97 unit is in partially enabled state, early request_irq can trigger
IRQ storm or even full hang up. Workaround this by forcibly switching ACLINK off
at the start of the probe.
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Let's have audio playback not sound like chipmunks, 'k? :)
ASP1 on the DM355 EVM uses a 27 MHz external audio clock, not
the slower clock used with ASP0 on the DM6446 EVM.
Also, that slower ASP0 clock on the DM6446 is 12.288 MHz,
not 22.5792 MHz ... 48 KHz sample rate (x256), not a double
speed 44.1 KHz sample rate (which could be done, but isn't
what the board init code now sets up).
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
