Dont create input devices for phantom jacks.
Here, we extend snd_jack_new() to support phantom jack creating:
pass in a bool param for [non-]phantom flag, and a bool param
initial_jack to indicate whether we need to create a kctl at
this stage.
We can also add a kctl to the jack after its created meaning we
can now integrate the HDA and ASoC jacks.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Configure the XIO2001 bridge on PCI Express cards so that it does less
needless prefetching.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make it possible for cards to have three stereo analog input pairs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cards without S/PDIF output do not need those controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a similar cleanup like the commit [3db084fd0a: ALSA: fm801:
PCI core handles power state for us].
Since pci_set_power_state(), pci_save_state() and pci_restore_state()
are already done in the PCI core side, so we don't need to it doubly.
Also, pci_enable_device(), pci_disable_device() and pci_set_master()
calls in PM callbacks are superfluous nowadays, too, so get rid of
them as well.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_pcm_suspend() function tests whether its argument is NULL and then
returns immediately. Thus the test around the call is not needed.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALSA core takes care that all preallocated memory is freed when the PCM
itself is freed. There is no need to do this manually in the driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds partial support for the Xonar Xense.
[trivial coding style fixes by tiwai]
Signed-off-by: Harley Griggs <hgriggs@posteo.co.uk>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect and handle the H6 daughterboard; it works the same as with the
ST, except that there is no conflict with the CS2000 chip.
Tested-by: Andreas Allacher <andreas.allacher@gmx.at>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add one more option to the "Headphones Impedance" control to synchronize
with recent versions of the Windows driver.
Tested-by: fugazzi® <fugazzi99@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should prefer `struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to
meet kernel coding style guidelines. This issue was reported by checkpatch.
A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@
- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;
// </smpl>
[bhelgaas: add semantic patch]
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Bjorn Helgaas <bhelgaas@google.com>
Just add the PCI ID for the STX II. It appears to work the same as the
STX, except for the addition of the not-yet-supported daughterboard.
Tested-by: Mario <fugazzi99@gmail.com>
Tested-by: corubba <corubba@gmx.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code introduced in commit 1f91ecc14d ("ALSA: oxygen: modify
adjust_dg_dac_routing function") accidentally disregarded the old value
of the playback routing register, so it broke the "Stereo Upmixing"
mixer control.
The unmuted parts of the channel routing are the same for all settings
of the output destination, so it suffices to revert that part of the
patch.
Fixes: 1f91ecc14d ('ALSA: oxygen: modify adjust_dg_dac_routing function')
Tested-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove old SPI control functions, change anti-pop init
sequence, remove some garbage from structures. The 'Apply' functions
must be called at the mixer initialization, otherwise
mixer settings sometimes will not be applied at startup.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
First of all, we should not touch the GPIOs. They are not
for selecting the capture source, but they seems just enable
the whole audio input curcuit. The 'put' function calls the
'apply' functions to change register values. Change the order
of capture sources.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the input_vol_* functions to use the new SPI routines,
There is a new applying function that will be called when
the capture source changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
I tried both variants: volume control and impedance selector.
In the first case one minus is that we can't change the
volume of multichannel output without additional software
volume control. However, I am using this variant for the
last three months and this seems good. All multichannel
speaker systems have internal amplifier with the
volume control included, but not all headphones have
this regulator. In the second case, my software volume
control does not save the value after reboot.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the order of elements in the output select control. This will
reduce the number of relay switches. Change 'put' function to call the
oxygen_update_dac_routing() function. Otherwise multichannel playback
does not work. Also there is a new function to apply settings, this
prevents from duplicating the code.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Actually CS4245 connected to the I2S channel 1 for
capture, not channel 2. Otherwise capturing and
playback does not work for CS4245.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Moving the mixer code away makes things easier. The mixer
will control the driver, so the functions of the
driver need to be non-static.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the function to read the data from the new shadow buffer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio output destinations (headphones,
FP headphones, multichannel output), the channel routing
should be changed depending on what destination selected.
Also unnecessary I2S channels are digitally muted. This
function called when the user selects the destination
in the ALSA mixer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change CS4245 initialization: different sequence and GPIO values,
according to datasheets and reverse-engineering information.
Change cleanup/resume/suspend functions, since they use
initialization.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add the new SPI write and read functions. The SPI read function
is used for creating initial registers dump and may be used for
debugging purposes. SPI operations are cached, so there is a new
function to manage the cache (shadow). I have to remove
the shift from the CS4245_SPI_* constants, since when
we are performing the reading, we need to shift by 8 instead
of 16.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add additional constants to the xonar_dg.h file:
capture and playback sources. Move GPIO_* constants and the
dg struct to the header file from the xonar_dg.c file.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add some additional information in comments and my copyright.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When the user switches the output from stereo to multichannel
or vice versa, the driver needs to update the channel routing.
Instead of creating additional subroutines, I better export existing
oxygen_update_dac_routing symbol from the oxygen mixer
and call this function. It calls model.adjust_dac_routing()
and my function does the work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the oxygen_write_spi() function to use the newly
introduced oxygen_wait_spi() function. Change return value
from void to int, so it can return error codes. Older
drivers just ignore that return value, new drivers can
check this value. We need to wait AFTER
initiating the SPI transaction, otherwise read
operation will not work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The oxygen_wait_spi() function now performs waiting when the
SPI bus completes a transaction. Introduce the timeout error
checking and increase timeout to 200 from 40.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
As drvdata is cleared to NULL at probe failure or at removal by the
driver core, we don't have to call pci_set_drvdata(pci, NULL) any
longer in each driver.
The only remaining pci_set_drvdata(NULL) is in azx_firmware_cb() in
hda_intel.c. Since this function itself releases the card instance,
we need to clear drvdata here as well, so that it won't be released
doubly in the remove callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for ASUS - Xonar DSX sound cards. Tested on
openSUSE 12.2 with kernel:
Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux
Works:
- play sounds
- adjust volume on master channel.
- mute .
Since Xonar DS uses the same chip, everything that works for DS should
work for DSX as well.
Signed-off-by: Sergiu Giurgiu <sgiurgiu11@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)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=uNqD
-----END PGP SIGNATURE-----
Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
flush[_delayed]_work_sync() are now spurious. Mark them deprecated
and convert all users to flush[_delayed]_work().
If you're cc'd and wondering what's going on: Now all workqueues are
non-reentrant and the regular flushes guarantee that the work item is
not pending or running on any CPU on return, so there's no reason to
use the sync flushes at all and they're going away.
This patch doesn't make any functional difference.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Russell King <linux@arm.linux.org.uk>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ian Campbell <ian.campbell@citrix.com>
Cc: Jens Axboe <axboe@kernel.dk>
Cc: Mattia Dongili <malattia@linux.it>
Cc: Kent Yoder <key@linux.vnet.ibm.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Jiri Kosina <jkosina@suse.cz>
Cc: Karsten Keil <isdn@linux-pingi.de>
Cc: Bryan Wu <bryan.wu@canonical.com>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Cc: Alasdair Kergon <agk@redhat.com>
Cc: Mauro Carvalho Chehab <mchehab@infradead.org>
Cc: Florian Tobias Schandinat <FlorianSchandinat@gmx.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: linux-wireless@vger.kernel.org
Cc: Anton Vorontsov <cbou@mail.ru>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: "James E.J. Bottomley" <James.Bottomley@HansenPartnership.com>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Eric Van Hensbergen <ericvh@gmail.com>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Steven Whitehouse <swhiteho@redhat.com>
Cc: Petr Vandrovec <petr@vandrovec.name>
Cc: Mark Fasheh <mfasheh@suse.com>
Cc: Christoph Hellwig <hch@infradead.org>
Cc: Avi Kivity <avi@redhat.com>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver accidentally exchanged the left/right fields for stereo AC'97
mixer registers. This affected only the aux and CD inputs because the
line input bypasses the AC'97 codec and the mic input is mono; cards
without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected.
Reported-and-tested-by: Abby Cedar <abbycedar@yahoo.com.au>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.31+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
