Added functions to report jack sense.
As CXT5051_PORTB_EVENT has the same value as CONEXANT_MIC_EVENT two input
devices for the microphone will be created if using CXT5051.
Signed-off-by: Ulrich Dangel <uli@spamt.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since internal to v4l2 the ioctl prototype is the same regardless of it
being called through .ioctl or .unlocked_ioctl, we need to convert it all
to the long return type of unlocked_ioctl.
Thanks to Jean-Francois Moine for posting an initial patch for this and
thus bringing it to our attention.
Cc: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Introduce a struct v4l2_file_operations for v4l2 drivers.
Remove the unnecessary inode argument.
Move compat32 handling (and llseek) into the v4l2-dev core: this is now
handled in the v4l2 core and no longer in the drivers themselves.
Note that this changeset reverts an earlier patch that changed the return
type of__video_ioctl2 from int to long. This change will be reinstated
later in a much improved version.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Use the USB functions usb_get_intfdata and usb_set_intfdata instead of
dev_get_drvdata and dev_set_drvdata, respectively.
The semantic patch that makes this change for the usb_get_intfdata case is
as follows: (http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@header@
@@
#include <linux/usb.h>
@same depends on header@
position p;
@@
usb_get_intfdata@p(...) { ... }
@depends on header@
position _p!=same.p;
identifier _f;
struct usb_interface*intf;
@@
_f@_p(...) { <+...
- dev_get_drvdata(&intf->dev)
+ usb_get_intfdata(intf)
...+> }
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the invalid dma channel to -1 (and check properly for it) in
pxa2xx_pcm_hw_free(). Was assuming 0 is an invalid channel number but 0
is a valid pxa dma channel num.
Signed-off-by: stephen <stephen.ware@eqware.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the more specific preset for ALC1200 above the general one for
ALC888, so that it will have the chance to get matched and selected.
Reported-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This set of patches introduces calls to the following set of functions:
usb_endpoint_dir_in(epd)
usb_endpoint_dir_out(epd)
usb_endpoint_is_bulk_in(epd)
usb_endpoint_is_bulk_out(epd)
usb_endpoint_is_int_in(epd)
usb_endpoint_is_int_out(epd)
usb_endpoint_num(epd)
usb_endpoint_type(epd)
usb_endpoint_xfer_bulk(epd)
usb_endpoint_xfer_control(epd)
usb_endpoint_xfer_int(epd)
usb_endpoint_xfer_isoc(epd)
In some cases, introducing one of these functions is not possible, and it
just replaces an explicit integer value by one of the following constants:
USB_ENDPOINT_XFER_BULK
USB_ENDPOINT_XFER_CONTROL
USB_ENDPOINT_XFER_INT
USB_ENDPOINT_XFER_ISOC
An extract of the semantic patch that makes these changes is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r1@ struct usb_endpoint_descriptor *epd; @@
- ((epd->bmAttributes & \(USB_ENDPOINT_XFERTYPE_MASK\|3\)) ==
- \(USB_ENDPOINT_XFER_CONTROL\|0\))
+ usb_endpoint_xfer_control(epd)
@r5@ struct usb_endpoint_descriptor *epd; @@
- ((epd->bEndpointAddress & \(USB_ENDPOINT_DIR_MASK\|0x80\)) ==
- \(USB_DIR_IN\|0x80\))
+ usb_endpoint_dir_in(epd)
@inc@
@@
#include <linux/usb.h>
@depends on !inc && (r1||r5)@
@@
+ #include <linux/usb.h>
#include <linux/usb/...>
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds DAPM implementaion for the capture path
on twlx030.
TWL has two physical ADC and two digital microphone (stereo) connections.
The CPU interface has four microphone channels.
For simplicity the microphone channel paths are named as:
TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid
TX2 (Left/Right)
Input routing (simplified version):
There is two levels of mux settings for TWL in input path:
Analog input mux:
ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic}
ADCR <- {Off, Sub mic, AUXR}
Analog/Digital mux:
TX1 Analog mode:
TX1L <- ADCL
TX1R <- ADCR
TX1 Digital mode:
TX1L <- Digimic0 (Left)
TX1R <- Digimic0 (Right)
TX2 Analog mode:
TX2L <- ADCL
TX2R <- ADCR
TX2 Digital mode:
TX2L <- Digimic1 (Left)
TX2R <- Digimic1 (Right)
The patch provides the following user controls for the capture path:
Mux settings:
"TX1 Capture Route": {Analog, Digimic0}
"TX2 Capture Route": {Analog, Digimic1}
"Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic}
"Analog Right Capture Route": {Off, Sub Mic, AUXR}
Volume/Gain controls:
"TX1 Digital Capture Volume": Stereo gain control for TX1 path
"TX2 Digital Capture Volume": Stereo gain control for TX2 path
"Analog Capture Volume": Stereo gain control for the analog path only
Important things for the board files:
Microphone bias:
"Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path)
"Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path)
"Headset Mic Bias": Bias for Headset mic
When the routing configured correctly only the needed components will be
powered/enabled.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the enum filter to more generic that it will filter
out the enums with text "Invalid".
The enum filter also required for the capture path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (583 commits)
V4L/DVB (10130): use USB API functions rather than constants
V4L/DVB (10129): dvb: remove deprecated use of RW_LOCK_UNLOCKED in frontends
V4L/DVB (10128): modify V4L documentation to be a valid XHTML
V4L/DVB (10127): stv06xx: Avoid having y unitialized
V4L/DVB (10125): em28xx: Don't do AC97 vendor detection for i2s audio devices
V4L/DVB (10124): em28xx: expand output formats available
V4L/DVB (10123): em28xx: fix reversed definitions of I2S audio modes
V4L/DVB (10122): em28xx: don't load em28xx-alsa for em2870 based devices
V4L/DVB (10121): em28xx: remove worthless Pinnacle PCTV HD Mini 80e device profile
V4L/DVB (10120): em28xx: remove redundant Pinnacle Dazzle DVC 100 profile
V4L/DVB (10119): em28xx: fix corrupted XCLK value
V4L/DVB (10118): zoran: fix warning for a variable not used
V4L/DVB (10116): af9013: Fix gcc false warnings
V4L/DVB (10111a): usbvideo.h: remove an useless blank line
V4L/DVB (10111): quickcam_messenger.c: fix a warning
V4L/DVB (10110): v4l2-ioctl: Fix warnings when using .unlocked_ioctl = __video_ioctl2
V4L/DVB (10109): anysee: Fix usage of an unitialized function
V4L/DVB (10104): uvcvideo: Add support for video output devices
V4L/DVB (10102): uvcvideo: Ignore interrupt endpoint for built-in iSight webcams.
V4L/DVB (10101): uvcvideo: Fix bulk URB processing when the header is erroneous
...
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (407 commits)
[ARM] pxafb: add support for overlay1 and overlay2 as framebuffer devices
[ARM] pxafb: cleanup of the timing checking code
[ARM] pxafb: cleanup of the color format manipulation code
[ARM] pxafb: add palette format support for LCCR4_PAL_FOR_3
[ARM] pxafb: add support for FBIOPAN_DISPLAY by dma braching
[ARM] pxafb: allow pxafb_set_par() to start from arbitrary yoffset
[ARM] pxafb: allow video memory size to be configurable
[ARM] pxa: add document on the MFP design and how to use it
[ARM] sa1100_wdt: don't assume CLOCK_TICK_RATE to be a constant
[ARM] rtc-sa1100: don't assume CLOCK_TICK_RATE to be a constant
[ARM] pxa/tavorevb: update board support (smartpanel LCD + keypad)
[ARM] pxa: Update eseries defconfig
[ARM] 5352/1: add w90p910-plat config file
[ARM] s3c: S3C options should depend on PLAT_S3C
[ARM] mv78xx0: implement GPIO and GPIO interrupt support
[ARM] Kirkwood: implement GPIO and GPIO interrupt support
[ARM] Orion: share GPIO IRQ handling code
[ARM] Orion: share GPIO handling code
[ARM] s3c: define __io using the typesafe version
[ARM] S3C64XX: Ensure CPU_V6 is selected
...
* 'timers-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
hrtimers: fix warning in kernel/hrtimer.c
x86: make sure we really have an hpet mapping before using it
x86: enable HPET on Fujitsu u9200
linux/timex.h: cleanup for userspace
posix-timers: simplify de_thread()->exit_itimers() path
posix-timers: check ->it_signal instead of ->it_pid to validate the timer
posix-timers: use "struct pid*" instead of "struct task_struct*"
nohz: suppress needless timer reprogramming
clocksource, acpi_pm.c: put acpi_pm_read_slow() under CONFIG_PCI
nohz: no softirq pending warnings for offline cpus
hrtimer: removing all ur callback modes, fix
hrtimer: removing all ur callback modes, fix hotplug
hrtimer: removing all ur callback modes
x86: correct link to HPET timer specification
rtc-cmos: export second NVRAM bank
Fixed up conflicts in sound/drivers/pcsp/pcsp.c and sound/core/hrtimer.c
manually.
The card based on stv0299 or stv0288 demodulators.
Signed-off-by: Igor M. Liplianin <liplianin@me.by>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.
- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
error is populated also there
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add check to determine if dinput_mux is set by any of patch_stac*() functions,
otherwise a invalid pointer my be referenced causing gibberish to mixer values.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing __devexit annotation to wm8350_codec_remove():
sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.
Add dsp_a, dsp_b mode options
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Minor, just move a block of code to make next patch clearer.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just at little cleanup of davinci_i2s_set_dai_fmt
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Document the current polarity choices.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add constants with a value of 0 to show more explicitly
what is being requested.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The capture with 44.1kHz on ca0106 seems to cause loud noises on
later playbacks, which doesn't support 44.1kHz. A simple fix is to
disable 44.1kHz, as the "default" PCM with dsnoop is anyway only with
48kHz.
Reference: Novell bnc#447624
https://bugzilla.novell.com/show_bug.cgi?id=447624
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no jack detection is available, the pins should be always
turned on since it can't be turned on/off dynamically via unsol
events.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.
Signed-off-by: Stanley Miao <stanley.miao@windriver.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added probe_only module option to hd-audio driver.
This option specifies whether the driver creates and initializes the
codec-parser after probing. When this option is set, the driver skips
the codec parsing and initialization but gives you proc and other
accesses. It's useful to see the initial codec state for debugging.
The default of this value is off, so the default behavior is as same
as before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line_out has only one DAC and it's unique (i.e. not shared
by other outputs), assign a more reasonable and distinct mixer name
such as "Headphone" or "Speaker".
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current auto-configuration code has several problems especially
for the new IDT codecs, e.g. wrong assignment of pins and DACs or
coupled volume for speaker and headphone.
This patch is a fairly large rewrite of the auto-configuration code.
Some remaks
- mic_switch and line_switch contain NIDs instead of bool
- dac_list isn't fixed for IDT 92HD* codecs now, they are all probed
- extra HP and speakers are stored in extra_dacs[].
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit re-enabled hp_nid setup for IDT92HD73*, but
it's unneeded indeed for Dell laptops that have multiple headphones.
Setting the extra hp_nid results in a non-working "Headpohne" mixer
control. Thus hp_nid should be 0 for these dell models.
Also, the automatic addition of hp_nid should check whether it's
a dual-HP model or not. For dual-HPs, the pins are already checked
by the early workaround.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added "IEC958 PCM Stream" controls for the per-stream IEC958 status
bits. Using this instead of "IEC958 Default" is safer since the status
bits will be recovered to the default states after closing the PCM
stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the call of snd_ctl_add() by replacing with snd_hda_ctl_add()
so that this mixer element can be tracked for re-configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The re-initializations of codec amp and verb caches are missing
at reconfig, which may cause Oops occasionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model without the jack-detection for some desktops that
have really no jack-detection. The recent driver caused regressions
regarding the sound output on such machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 07f455f779.
ALSA: hda: removed unneeded hp_nid references
Removed unneeded hp_nid references for 92hd73xx codec family.
This caused the silent output on some Intel desktops due to missing
routing of widget 0x0a and 0x0d.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.
The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.
This driver was originally written by Liam Girdwood with further updates
from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This should never happen and it's helpful to identify the specific control
that failed when it does happen.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than listing lots of architectures per line in Kconfig and
Makefile, causing merge conflicts all the time, have one per line
in alphabetical order.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace all tasklet_hi_schedule() callers with the normal
tasklet_schedule(). The former often causes troubles with
RT-kernels, and has actually no merit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove codec vendor names from the codec name strings.
The vendor name is already given from the vendor name table, so
displayed doubly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some desktops seems to have no HP/mic jack detection on the front panel,
which results in the silent output in the recent driver, because the
driver mutes the output (to save power) when no plug is detected.
This patch adds a new model that disables the jack-detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes an inconsistency that became apparent when I
documented the fields of snd_ca0106_details. spi_dac is always
used in a 'boolean' sense, so this cleanup should make no difference.
[Actually, there is one place checking explicitly spi_dac == 1, so
this will change the behavior. But, supposing it's rather a typo,
I apply this clean-up patch -- tiwai]
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi wrote an email [1] explaining the fields of snd_ca0106_details,
so I captured the information into the ca0106.h header file.
[1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_type
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include sound/core.h in sound_core.c so that sound_class is declared
before it is defined, avoiding it looking like it should be static.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Netwinder was using gpio_xxx names which could clash with the GPIO
layer. Add a 'nw_' prefix to ensure that these remain separate.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
A special start-up sequence is required to reduce the pop-noise of Class D
amplifier when enable hands-free on TWL4030.
Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed a compile warning below:
sound/isa/sb/sb8.c: In function ‘snd_sb8_probe’:
sound/isa/sb/sb8.c:104: warning: ‘err’ may be used uninitialized in this function
by setting the return value correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the compile warning regarding the unused function when built
with CONFIG_PM=n:
sound/pci/hda/hda_intel.c:1905: warning: ‘snd_hda_codecs_inuse’ defined but not used
snd_hda_codecs_inuse() is used only in the resume callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the registration of dais in s3c2443-ac97.c.
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_init':
sound/soc/s3c24xx/s3c2443-ac97.c:401: warning: passing argument 1 of 'snd_soc_register_dai' from incompatible pointer type
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_exit':
sound/soc/s3c24xx/s3c2443-ac97.c:407: warning: passing argument 1 of 'snd_soc_unregister_dai' from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver now registers the codec and DAI when probed as an I2C device.
Also convert the driver to use a single dynamic allocation to simplify
error handling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Redo the instantiation of the WM8900 to do most of the initialisation
work when the I2C driver probes rather than when the ASoC device is
instantiated, registering the codec with the ASoC core when done.
Also move all dynamic allocations into a single kmalloc() to simplify
error handling and rename the I2C driver to make output more sensible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The GPIO stuff for OLPC in cs5535audio_olpc.c is implemented only for
Geode-LX, and enabled only when CONFIG_MGEODE_LX=y. Without this
config option, the driver gets build errors.
This patch adds a workaround to make it dependent on CONFIG_MGEODE_LX.
Ideally, the OLPC-GPIO stuff should be implemented in a way
independent from CPU type selection...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add copyright info to _olpc.c
- minor layout fixes
- make Makefile more concise
- silence a warning
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Always turn off mic bias; the MIC LED should never come on when the
driver is first loaded.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This drops the AD1888 V_REFOUT control, and replaces it with a MIC Bias
Enable control. It also moves the MIC bias enabling into a separate
function.
Signed-off-by: Andres Salomon <dilinger@debian.org>
The OLPC has a privacy light hooked up in series with the microphone's
V_Ref bias. We want to activate the bias while we are capturing audio.
Signed-off-by: Chris Ball <cjb@laptop.org>
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checking the HPF register is irrelevant; HPF is secondary to the AI mode.
Instead, check for Analog Input mode via GPIO.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously, we had two separate controls; there's no need to have AD1888's
HPF control, so drop it if we're on an OLPC machine. Also, as per Arjun's
request, rename OLPC's Analog Input Switch control to "DC Mode Enable".
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We shouldn't be touching V_REFOUT when we toggle HPF/analog input, so just
drop that code.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix an audible pop described in <http://dev.laptop.org/ticket/977>. Originally
based upon fixes by Mitch Bradley and Chris Ball.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Use basic infrastructure code; geode_gpio* (rather than indexed i/o
EC access), and do an OLPC machine check in olpc_quirk.
[dilinger@debian.org: don't return failure in olpc_quirks if !OLPC]
[dilinger@debian.org: drop the <B2 workarounds; those machines are EOL'd]
Signed-off-by: Jordan Crouse <jordan.crouse@amd.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
This is a 2nd cut at adding support for OLPC analog input.
Signed-off-by: Jaya Kumar <jayakumar.lkml@gmail.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
snd_cs5535audio_suspend and snd_cs5535audio_resume are only defined when
CONFIG_PM is set; make that clear in the header file.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per <http://dev.laptop.org/ticket/1420>, we need to properly turn off
the PCM if we're closing the device in order to save power. This also
causes the MIC led to turn off properly.
Signed-off-by: Jaya Kumar <jayakumar.lkml@gmail.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.
Thanks to Guillem Jover for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion the names for the DACs changed:
DACL1 -> DAC Left1
...
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mux switch related texts fits to on line, no need to wrap
them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BUG() should be marked as not returning but for at least some
configurations (including some widely deployed compilers) that's either
not happening or being forgotten by the compiler. Add some extra return
statements to the affected paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The return value of pci_enable_device() must be checked even in resume
callback:
sound/pci/ca0106/ca0106_main.c:1779: warning: ignoring return value of ‘pci_enable_device’, declared with attribute warn_unused_result
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add snd_ prefix to avoid the conflict of symbols in omac-mcbsp.c:
sound/soc/omap/omap-mcbsp.c:503: error: static declaration of 'omap_mcbsp_init' follows non-static declaration
arch/arm/plat-omap/include/mach/mcbsp.h:373: error: previous declaration of 'omap_mcbsp_init' was here
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the function name of module init entry for twl4030.c, which
conflicted with the existing hardware init function:
sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init'
sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here
Also fixed the section type of init function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes use of the support for delayed DAI registration to allow the
WM8900 I2C device to be registered by general platform/architecture code
rather than as part of the ASoC device probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will allow codec drivers to be refactored to allow them to be
registered out of line with the ASoC device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the lists of platforms, platform DAIs and cards to check to see that
everything has registered. Since relationships are still specified by
direct references to the structures in the drivers and the drivers all
register everything at modprobe there should be no practical effect yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently this is done at module probe time since ASoC ties in codec
device probe to the instantiation of the entire ASoC device. Subsequent
patches will refactor the codec drivers to handle probing separately.
Note that the core does not yet use this information.
AC97 is special since the codec is controlled over the AC97 link but
we want to give the machine driver a chance to set up the system before
trying to instantiate since it may need to do configuration before the
AC97 link will operate
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a separate gain control for the Headset output already.
Do not reset the gain to 0 dB at power up.
In power-down, there is no need to set the Headset output gain
to power-down mode, since if the CODECPDZ is in powered off this
setting has no effect.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Handsfree outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Carkit outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Headset outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the PreDrive outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Earpiece output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four APGA switch to DAPM routing and widgets.
Add user control for DA enable for all APGA as normal
control.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four DACs to dapm_widgets with power switch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the license and misc comments at the beginning of the code.
Also, use ns_to_ktime() for simplification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the restoration of the standard PCI configuration registers
in the snd_hda_intel driver to a ->resume_early() callback executed
with interrupts disabled, since doing that with interrupts enabled
may lead to problems in some cases.
This patch addresses the regression from 2.6.26 tracked as
http://bugzilla.kernel.org/show_bug.cgi?id=12121 .
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds basic support for OMAP3 Pandora.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code overrides the event type on input pins always to
PWR_EVENT. Although this still works (PWR_EVENT and INSERT_EVENT
are handled samely), it'd be better to avoid such overrides.
Also, currently the unsol events are registered even for fixed pins
which will never raise the pin-detection event.
This patch fixes both issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin-detection function used in patch_sigmatel.c shouldn't be specific
to HP pin because it's used for input pins in general, too.
This patch fixes the detection function, removes the HP check from it
and moves to stac92xx_hp_detect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce the command timeout to 0.5sec. Should be enough to allow a
working command interface but removes a RCU stall and slow resume on
some revisions where the AC97 revision detection stalls in resume.
Signed-off-by: Peter Gruber <nokos@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add aic3x_set_headset_detection() function to define the headset
detection mode for tlv32aic3x chips
- added aic3x_button_pressed()
- Read from the real-time registers in aic3x_headset_detected() to query
headset presence without an occured interrupt
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TWL4030 codec device has two ADCs. Both of them can have
several inputs routed to them, but TRM says that only one source
can be selected for every ADC, even though every source has a
dedicated bit in the registers.
This patch adds input source controls. It modifies default register
values to have no inputs selected and ADCs disabled. When some
input is selected, control handlers enable apropriate input
amplifier and ADC. If a microphone is selected, bias power is
automatically enabled. When some input is deselected, unused
chip parts are disabled.
Microphone and line input recording tested on OMAP3 pandora board.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a sparse warning caused by the lack of a connection with the
prototype for ac97_bus_type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the conversion (drop) from 24bit in the DSP to 16bit in AC97,
the maximum capture level on Audigy seems lower than it could be.
This patch adds a workaround to enable the artificial capture boost
switch. When this switch is on, the whole analog capature level is
boost up. However, this results in the lower capture resolution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
All outputs have dedicated gain controls except the
HandsFree output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Playback volume controls for all four DACs.
All four paths has three levels of volume controls:
Digital Fine gain, Digital Coarse gain, Analog gain.
The controls are named to reflect their connection to the DACs.
Per DAC volume can be performed, if needed:
amixer sset 'DAC1 Analog' 5,10
DACL1 analog gain to 5
DACR1 analog gain to 10
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital Capture gain control has a range:
0 to 31 dB in 1 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the old-style trigger callback in s3c2443-ac97.c:
sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrong shutdown callback type. Also removed the unused variables
there:
sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
sound/soc/pxa/corgi.c: At top level:
sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9171e5e6a2.
I can't reproduce the compile warnings any more. The warnings
might be some weird cross-compiling set up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig,
thus no more need in Kconfig of each sub directory.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit e669dae614, since it
is incomplete, and clashes with fuller patches and the sparc 32/64
unification effort.
Requested-by: David Miller <davem@davemloft.net>
Acked-by: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The WM9715 is software compatible with the WM9711 and WM9712.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hide annoying uninitialized warnings:
sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function
sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- compatibility issue : change firmware filenames
the pcxhr driver version <= 1.0.18a does not work
with new firmware > 1.0.17. Keep the old firmware files
and add new firmware files with different names
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power-state changes in patch_sigmatel.c are accessed via *_cached()
but they shouldn't be really cached. Fixed to the normal write.
Also, stac92hd71xx_suspend and resume are no longer necessary as the
power-state changes are handled properly in the common routine.
Removed these hacks now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp switch of SPDIF outputs have to be cached in the amp cache
instead of codec cache. Otherwise it conflicts with the IEC958
playback switch control in hda_codec.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'merge' of git://git.kernel.org/pub/scm/linux/kernel/git/paulus/powerpc:
powerpc: Fix system calls on Cell entered with XER.SO=1
powerpc/cell: Fix GDB watchpoints, again
powerpc/mpic: Don't reset affinity for secondary MPIC on boot
powerpc/cell/axon-msi: Retry on missing interrupt
powerpc: Fix boot freeze on machine with empty memory node
powerpc: Fix IRQ assignment for some PCIe devices
powerpc/spufs: Fix spinning in spufs_ps_fault on signal
powerpc/mpc832x_rdb: fix swapped ethernet ids
powerpc: Use generic PHY driver for Marvell 88E1111 PHY on GE Fanuc SBC610
powerpc/85xx: L2 cache size wrong in 8572DS dts
powerpc/virtex: Update defconfigs
powerpc/52xx: update defconfigs
xsysace: Fix driver to use resource_size_t instead of unsigned long
powerpc/virtex: fix various format/casting printk mismatches
powerpc/mpc5200: fix bestcomm Kconfig dependencies
powerpc/44x: Fix 460EX/460GT machine check handling
powerpc/40x: Limit allocable DRAM during early mapping
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Check model for Dell 92HD73xx laptops
ALSA: hda - mark Dell studio 1535 quirk
ALSA: hda - No 'Headphone as Line-out' swich without line-outs
ALSA: hda - Fix AFG power management on IDT 92HD* codecs
ALSA: hda - Fix caching of SPDIF status bits
ALSA: hda - Add a quirk for Dell Studio 15
ALSA: hda: Add STAC_DELL_M4_3 quirk
sound/sound_core: Fix sparse warnings
ALSA: hda: STAC_DELL_M6 EAPD
switch to __init for those; unlike powerpc sparc has no hotplug support
for that stuff and their ->probe() tends to call __init functions while
being declared __devinit.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Gateway notebooks have their ID inside codec vendor ID, not at PCI ID. Due to
that, model auto-detection were not possible with the standard seek method.
This is what is found at lspci -vnn:
00:14.2 Audio device [0403]: ATI Technologies Inc SB450 HDA Audio [1002:437b] (rev 01)
Subsystem: ATI Technologies Inc SB450 HDA Audio [1002:437b]
Yet, autodetection is possible, since the codec properly reflects the vendor at
the Subsystem ID:
$ cat /proc/asound/card0/codec#0 |head -4
Codec: SigmaTel STAC9250
Address: 0
Vendor Id: 0x83847634
Subsystem Id: 0x107b0367
This patch adds a new autodetection function that seeks for codec subsystem ID.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The global functions in hda_codec.c and other core parts are only
for HD-audio codec and controller drivers. When the HD-audio driver
is built in kernel, all stuff have to be statically linked, thus
we don't need any exports.
This patch introduces a conditional macro to do export only
when needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
