This patch adds support for the Toshiba M105-S3041 laptop (ALC861).
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
soc-dapm
·Removed list_for_each since the loop is list_for_each_entry() and
not list_for_each(). Thanks to Liam Girdwood and Seth Forshee.
at91-i2s
·Fixed typo in dai modes definition.
·Fixed struct member name in at91_ssc_info->ssc_state.
·Fixed compilation problem, ssc_state is bundled in at91_ssc_info.
Signed-off-by: Raúl Sánchez Siles <rss@barracuda.es>
Signed-off-by: Seth Forshee <seth.forshee@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch changes the default configuration for the Asus P5GD1
motherboard from 5stack to asus, as reported by stelek on
linuxquestions.org
http://www.linuxquestions.org/questions/showthread.php?p=2556497#post2556497
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the support of the ESI Waveterminal 192M soundcard
to the ice1724 familly ALSA driver.
It's a semi-professionnal soundcard for home studio : many I/O and
a quality of sound is good, better than consumer cards, but less
musical than professional cards.
It use a Via Envy24ht chipset as ice1724 soundcard, Sigmatel
stac9640 ADC/DAC for the analog I/O as Prodigy192, and Atmel ak4114
for S/PDIF as ESI Julia.
Is working : the 8 analog outputs, the analog inputs 1&2, the mic
input 1, the coaxial & optical digital outputs.
Signed-off-by: Clement Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the Asus P5W DH to the ALC882 config table
as a 6stack-dig system.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Went rummaging through usbaudio.c and found some castings that
aren't needed as far as I can see. Part of the KernelJanitors
TODO list.
Signed-off-by: John Daiker <daikerjohn@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Previously, ac97_codec.c was coded to support AD1986 and AD1986A
CODECs using code written for the AD1985 CODEC. This allowed the
LINE_OUT and HEADPHONE jacks to function properly, however register
differences between the CODECs prevented line and microphone inputs
from functioning.
Specifically, this patch fixes issues with the following mixer
controls: 'V_REFOUT', 'Spread Front to Surround and Center/LFE',
'Exchange Front/Surround', 'Surround Jack Mode', and 'Channel Mode'.
This patch removes the undocumented AD1888 control
'High Pass Filter Enable' and adds the new control
'Exchange Mic/Line In'.
Signed-off-by: Randy Cushman <rcushman_linux@earthlink.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch replaces the 'V_REFOUT Enable' mixer switch control
with a listbox control for the AD1985 CODEC.
Previous patch 'AD1888 mixer controls for DC mode' added
controls that were propogated to multiple codecs. For the
AD1985 codec, the bits VREFH and VREFD function differently,
preventing the 'V_REFOUT Enable' control from setting V_REFOUT
to Hi-Z.
This patch also corrects an issue in which register bits relating
to mixer controls 'Surround Jack Mode' and 'Channel Mode'.
The register bits controlled by these controls were being set
at boot time to states inconsistent with the stored values of
these controls.
Signed-off-by: Randy Cushman <rcushman_linux@earthlink.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix the changes realted to delayed_work in soc/codecs/wm8750.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes the Microphone and LINE_IN select logic for
Analog Devices surround codecs with shared jacks. The existing
code can never utilize the shared jacks for Microphone and LINE_IN
due to the reversed jack selection logic. The patched code
correctly selects the shared jack for input if the 'Channel Mode'
selector does not specify that the jack is to be used for output.
Specifically, in '2ch' mode the Center/LFE jack is used for
microphone input and the Surround jack is used for LINE_IN,
in '4ch' mode the Center/LFE jack is used for microphone input
and the Surround jack is used for output, and in '6ch' mode
both jacks are used for output.
Signed-off-by: Randy Cushman <rcushman_linux@earthlink.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use global workqueue for simplicity instead of own workqueue
in SoC core and wm8750 codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use global workqueue for simplicity instead of own workqueue.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use global workqueue for simplicity.
The unsolicited event frequency isn't so high to have own queue.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Change the two remaining instances in the tree of kcalloc(1,...) to
the corresponding kzalloc() call.
Signed-off-by: Robert P. J. Day <rpjday@mindspring.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The STAC9708/11 AC97 codecs implement the PCM Out Path & Mute bit in
the General Purpose register (0x20:F), even though they don't implement
the actual function in the mixer.
Since the alsa tests for the function by toggling the bit and reading
it back to see if it changed, it mistakenly creates a useless control.
This patch explicitly removes the control when the codec is an
STAC9708/11.
I put the check in patch_sigmatel_stac9708_specific(), because I have
an SBLive with this chip on it. I don't know if the STAC9758 or other
codecs also behave this way. If they do, then this check could maybe go
in patch_sigmatel_stac97xx_specific(), or some other more general
function.
Signed-off-by: James C Georgas <jgeorgas@rogers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This is a comment fix to avoid misleading about locking in the
dbri_cmdsend.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds copyright and credit for my good friend Richard Purdie
from OpenedHand for his help and code contribution throughout the
development of the core code. Many thanks Richard (I guess we overlooked
this in trying to get everything working well).
It also adds some extra comments wrt to DAI clock matching.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch removes some trailing white space from the WM9712 ASoC codec
driver.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the Turbo-X Coeus G610P to the alc880 config table,
based on user provided information.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
When the machine resumes the onyx codec might be in a weird state. Hence,
simply fully reset it once (and keep the code to take it out of suspend in
case the suspend of the codec chip survives a reset).
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Adds support for handling EAPD on 9205 codecs
Signed-off-by: Matt Porter <mporter@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch supports Audiotrack 7.1 XT.
7.1XT is almost same hardware as 7.1LT. so using 7.1 LT's code.
Signed-off-by: Toshimune Konno <heitouk@nifty.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds limited support for Intel-based MacPro workstations.
Currently, the front headphone jack is not functioning, but line out
and line in are working. S/PDIF not tested.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
From: Andrew Morton <akpm@osdl.org>
I converted the workqueues to per-device while I was there. It seems
strange to create a new kernel thread (on each CPU!) and to then only
have a single global work to ever be queued upon it.
Plus without this, I'd have to use the _NAR stuff, gawd help me.
Does that workqueue really need to be per-cpu?
Does that workqueue really need to exist? Why not use keventd?
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
create sysfs driver symlink for snd-aoa in /sys/bus/aoa-soundbus/devices/*/
Acked-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Olaf Hering <olaf@aepfle.de>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
create sysfs device symlinks for snd-aoa in /sys/class/sound/controlC0 This
allows hald to recognize the device as sound device. Furthermore it allows
the desktop user to actually access the sound device nodes. hald and
related packages will modify the acl attributes.
Fixes https://bugzilla.novell.com/show_bug.cgi?id=106294
Acked-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Olaf Hering <olaf@aepfle.de>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Notebook.
Digital playback and capture now works, but it is not bit accurate because it
passes through a resampler.
Bit accurate playback and capture will be implemented later via the p17v.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Implement functionallity in order to fixe ALSA bug#2058.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Following patch will make the driver to use the 44.1kHz SRC automatically
if the pcm source is 44.1kHz signed 16bit stereo.
The SRC is available in YMF754 only.
Signed-off-by: Teru KAMOGASHIRA <teru@sodan.ecc.u-tokyo.ac.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch is VIA first release for HD audio codec, VT1708(A) and
it provides geneneral HD audio driver features.
Signed-off-by: Joseph Chan <josephchan@via.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Make the following needlessly global functions static:
- dapm_power_widgets()
- dapm_mux_update_power()
- dapm_mixer_update_power()
- dapm_free_widgets()
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The following patch creates a new 'Mono speaker' control in alsamixer
when the Realtek 'acer' model is used with hda_intel. This is needed so
the internal mono speaker (when present) can be controlled.
This new control won't do anything in Acer laptops which are not fitted with
a mono speaker. Acer models which are known to have a mono speaker are the
C20x tablet series but there may be others. I guess we could define a new
model specifically for Acers with mono speakers but this seems a bit silly
given that such a model will be identical to the normal 'acer' model except
for this added control.
This patch also adds the C20x tablets to the list of PCI ids associated with
the 'acer' model. This means that owners of C20x machines will no longer
have to supply 'model=acer' when loading hda_intel.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch by Philipp Zabel fixes a bug whereby the BCLK matching fails
when the Codec BCLK is constant and the CPU BCLK is based upon a
divider.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use snd_pci_quirk_lookup() for looking up a board config table.
The config table is sorted in numerical order of PCI SSIDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remove uesless typedefs and clean up the code a bit to follow
the standard coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Changes were required to support latest AT91 header files.
Also updated to remove AT91RM9200-specific code in the ASoC
platform drivers to support the AT91SAM9260 and AT91SAM9261
chips, but no testing was performed on these chips.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added spdif_aclink module option to specify whether the board
has SPDIF over AC-link or a direct connection from the controller
chip.
NForce and ICH4 (or newer) boards may be equipped with SPDIF
through AC97 codec. In such a case, SPDIF should be handled
as if the old ICH style (the same slot for analog and digital).
A quirk list is added to detect this automatically for known
hardwares.
Corresponds to ALSA bug#2637.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clean up quirks in snd-ens1371 driver using snd_pci_quirk_lookup().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clean up dxs_support quirk list using snd_pci_quirk_lookup().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clean up maestro3 amp and GPIO quirks using snd_pci_quirk_lookup().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a helper function snd_pci_quirk_lookup()
to look up PCI SSID quirk list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds TLV support to the echoaudio driver.
All gains are in the range -127dB to +6dB with steps of 1dB, and -128 is
mute. VU-meters levels go from -128 to 0dB. The input gain of the Layla20
ranges from -25dB to +25dB in steps of 0.5dB.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a new model 'asus-laptop' for ASUS F2*/F3* laptops
with ALC861 (equivalent with ALC660) codec chip.
Also fixed the model for PCI SSID 1043:1338.
Corresponding to ALSA bug#2480.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
sound/pci/korg1212/korg1212.c:2359: warning: format '%d' expects type 'int', but
argument 4 has type 'size_t'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Call pci_intx() to disable/enable INTX when MSI is used/unused.
Nvidia and AMD boards seem to have problems with MSI when INTX
isn't disabled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Now that everyone uses snd_ctl_new1() and noone is using snd_ctl_new()
anymore, we can make it static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a missing array to the conexant driver.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add an option to specify the AC'97 codec instead of
probing. This is a fix for bugzilla #7467.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch makes the needlessly global stac92xx_dmic_labels[] static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the Evesham Voyager C530RD series laptops.
So far, only playback has been tested, but microphone should also work.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a proper model entry (model=laptop-eapd) for ASUS W3j laptop
with AD1986A codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix the wrongly set SET_CONNECTION verb for NID 0x0f of ALC861.
The widget has only a single connection although the init verb
sets to 0x01.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch moves the entry for the Gigabyte K8N51 from the 6stack
grouping to the 6stack-digout grouping, allowing for S/PDIF output
functionality.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the model entry (model=hippo) for Sony UX-90s with ALC262 codec.
Although the device has no SPDIF output, the hippo model adds a
PCM output, but it must be harmless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This driver adds limited support for the Conexant 5045 and 5047 HD Audio
codecs. Some issues still need to be resolved. The code is based
primarily on code from the Analog Devices AD1981 support and the Realtek
ALC260 support. Some code came from the original code developed by Alex
Pototskiy (see alsa bugtracker 2485).
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the (experimental) support of M-Audio Audiophile 192 board.
Currently, the analog and the digital playbacks seem working fine.
The inputs seem not working as far as I've tested yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch suggested by Richard Purdie changes the names of some WM8731
and WM8750 mixers so that they will be recognised by some older OSS
mixer apps.
Changes:-
o WM8731 Playback changed to Master Playback
o WM8750 Out1 changed to Headphone
o WM8750 Out2 changed to Speaker
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
We need to enable External Amplifier on this laptops. This patch basicly
adds laptop-eapd model to ALC883 codec.
Signed-off-by: Andrew L. Neporada <nepal@asplinux.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
sound/pci/rme9652/hdspm.c: In function 'snd_hdspm_hw_params':
sound/pci/rme9652/hdspm.c:3681: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
sound/pci/rme9652/hdspm.c:3692: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add a proper model (3stack) for ASUS M2N-MX with AD1986A codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the ASSP codes using request_firmware(), if possible, instead of
using the built-in blobs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the DSP code using request_firmware(), if possible, instead of
using the built-in blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the YSS225 register initialization data using request_firmware(),
if possible, instead of using the built-in data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Instead of using a somewhat algorithmic approach of initializing the
YSS225's registers, just use a simple series of port/value pairs.
This makes it easier to later replace or entirely remove the register
data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the CSP programs using request_firmware(), if possible, instead of
using the built-in firmware blobs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch by Frank Mandarino and Hubert Kahlert fixes a bug in the AT91
SSC (i2s) shutdown code that would erroneously disable other AT91
peripheral clocks.
Signed-off-by: Hubert Kahlert <hkahlert@hk-datentechnik.de>
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Adds support for digital microphone pin widgets on SigmaTel codecs.
Enables support only on the 9205 codecs for now.
Signed-off-by: Matt Porter <mporter@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add a proper model entry (3stack) for Lenovo A60 desktop with
AD1986a codec to fix noise problems.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Don't enable power-saving mode on drivers that don't support
it. The supporting drivers set AC97_SCAP_POWER_SAVE to scaps
at creation of ac97 instance.
Currently enable on the following drivers: intel8x0, intel8x0m,
atiixp, atiixp-modem, via82xx and via82xx-modem.
Also, a bit clean up of power-saving stuff:
- Don't create an own workq
- Remove superfluous ifdefs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for Asus laptops (for example: Asus
A6Rp-AP002).
Signed-off-by: Mariusz Domanski <mariook@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch by Frank Mandarino updates the AT91RM9200 I2S DAI audio modes
as follows:-
o fixes a typo in the 16k mode
o removes experimental 24k mode
o adds a 32k mode.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add DDS register support for RME9632 rev >= 152.
This register sets the sample rate for these cards and is required
in addition to the standard control register. It corresponds to a
quartz divisor.
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Changes from Realtek driver:
- New models hippo and hippo_1 for ALC262
- New models tagra-dig and tagra-2ch-dig for ALC883
- New id for ALC660 codec chip
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for Toshiba laptops. Code is from
RealTek's alsa-driver-1.0.12-4.05b tree.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes some build warnings in soc-core.c
Changes:-
o Check the return value of soc_ac97_dev_register()
o Check return value of calls to device_create_file()
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add support for AES32. Difference between MADI and AES32 is done
through revision. Master support is not finished for now (RME so-called DDS
feature is not supported yet)
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remove trailing whitespaces from soc/* files added by the
conversion to C99-style initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes a build failure when ASoC debug is enabled.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch suggested by Takashi changes the DAI capabilities definitions
in pxa-i2s.c, at91rm9200-i2s.c, wm8731.c, wm8750.c and wm9712.c to use a
label = value style.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds audio support for Medion's line of laptops,
based on code shipped with the laptops. Microphone support is
still being explored.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds pxa2xx AC97 ASoC audio support. It's based on
sound/arm/pxa-ac97 by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks.
o Removed AC97 configuration function (all handled in ASoC core)
o Added and exported ASoC DAI configuration table.
o Added DMA support for AUX DAC and Mic ADC
o Separated out AC97 reset into cold and warm reset functions.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Nicolas Pitre <nico@cam.org>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds pxa2xx ASoC DMA audio support. It's based on
sound/arm/pxa-pcm.c by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks and data
structures.
o Registration with ASoC core.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Nicolas Pitre <nico@cam.org>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the DSP and controller microcode using request_firmware(), if
possible, instead of using the built-in firmware.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed the irq handler in soc/at91-at91rm9200-i2s.c to follow the
new style without pt_regs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the support for mixer matrix of RME9632 rev 152.
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Let the emu10k1 driver select FW_LOADER because the new Emu1010 support
requires it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER
instead of depending on it so that they can be configured without having
to enable FW_LOADER manually.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Properly quote a string that had an embedded newline.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a Makefile and Kconfig to build the ASoC AT91RM9200
support.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the Endrelia ETI_B1 machine using the WM8731
codec and the AT91RM9200 platform.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds I2S support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
o 8k - 48k sample rates.
o ssc0, ssc1 and ssc2 supported as I2S ports.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC audio DMA support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds an ASoC Makefile and Kconfig for the WM8731, WM8750 and
WM9712 codecs.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch allows the std Alsa AC97 codec driver to use any AsoC AC97
controller driver. Currently, only HiFi playback and Capture are
supported atm.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM9712 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o Aux DAC.
o 8k - 48k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM8750 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM8731 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for building the ASoC core and the dynamic audio
power management support.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds Dynamic Audio Power Management (DAPM) to ASoC.
Dynamic Audio Power Management (DAPM) is designed to allow portable and
handheld Linux devices to use the minimum amount of power within the
audio subsystem at all times. It is independent of other kernel PM and
as such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM:-
1. Codec domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
2. Platform/Machine domain - physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP are
inserted
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
Enabled and disabled when stream playback/capture is started and stopped
respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting
an audio routing map of the whole machine. This map is specific to each
machine and consists of the interconnections between every audio
component (including internal codec components).
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use pci_iomap and ioread*/iowrite*() functions for accessing
hardwares. pci_iomap is suitable for hardwares like ICH and
compatible that have both PIO and MMIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch converts most uses of list_for_each to list_for_each_entry all
across alsa. In some place apparently an item can be on a list with
different pointers so of course that isn't compatible with list_for_each, I
therefore didn't touch those places.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch changes i2sbus_attach_codec to implement a proper error handling
strategy using labels to jump to the right part. Since it has an elaborate
set-up sequence it also needs that tear-down, which I had hard-coded
inbetween all the checks. This increases readability and should reduce .text
size as well.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch makes a few whitespace cleanups and makes i2sbus assign the new
struct device pointer in struct snd_pcm so that the proper device symlink
shows up in sysfs.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility. This allows
to control the analog loopback of the card.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility. Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add pause capabilities for both USB playback and capture streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The hardware information structures for playback and capture streams,
respectively, are the same, so we can use just one structure for both
streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The recent change for a new sysfs tree with card* object breaks the
/sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled.
The device in each entry doesn't point the correct device object:
/sys/class/sound
...
|-- pcmC0D0c
| |-- dev
| |-- device -> ../../../class/sound/card0
| |-- pcm_class
| |-- power
| | `-- wakeup
| |-- subsystem -> ../../../class/sound
| `-- uevent
Also, this change breaks some drivers (like sound/arm/*) referring
card->dev directly to obtain the device object for memory handling.
This patch reverts the semantics of card->dev to the former version,
which points to a real device object. The card* object is stored in a
new card->card_dev field, instead. The device parent is chosen either
card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to
keep the tree compatibility.
Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled. The
reason of card* object is a root of all beloing devices, and it makes
little sense if each sound device points to the real device object
directly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Monty Montgomery <xiphmont@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
The previous patch 'Repair snd-usb-usx2y for usb 2.6.18' assumed
urb->start_frame roll over beyond MAX_INT for both UHCI & OHCI.
This isn't true until now (kernel 2.6.20).
Fix this by only looking at the common between OHCI & UHCI Frame number
range.
This is for mainline and stable kernels >= 2.6.18.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed the error from kobject_add() at reconnection the usb audio device.
This happens when an app keeps opening a device while the device is
replugged, due to the confliction of the internal bookkept index and
the really empty slot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Without the patch below namelist[0] will not be freed in case
of kmalloc error.
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Playing with spdif output on cmipci i've noticed the SPDO5V option does
not change appropriate bits the register.
The _snd_cmipci_uswitch_put checks the change in flags in wrong way.
If 'active' state of an option corresponds to a _zero_ bits in a hw
register then function fails. The SPDO5V is the sample.
In the most cases 'active' state of option is set through an non-zerio
bits in a register. This case works fine.
The fix attached.
Unfortunately i was unable to change spdif output voltage anyway.
Although the register changes right at least.
From: Timofei V. Bondarenko <tim@ipi.ac.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the Intel ICH9 HD Audio controller DID's for ALSA.
Signed-off-by: Jason Gaston <jason.d.gaston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The C-Media CM6501 chip's descriptors say that altsetting 5 supports
48 kHz, but it actually plays at 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix races between the timer handler and the close function.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add the support for HD audio controllers of MCP51,MCP55,MCP61,MCP65 & MCP67.
Signed-off-by: Peer Chen <pchen@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
SBUS: Change IRQ-handler return value from 0 to IRQ_HANDLED and
fix some initialisation problems.
Change period_bytes_min from 4096 to 256 to allow driver to work with
low latency (VOIP) applications. Hope this does not break EBUS.
Signed-off-by: Georg Chini <georg.chini@triaton-webhosting.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch fixes a couple of bit update functions in
alsa-kernel/pci/ac97/ac97_codec.c, which could possibly corrupt bits not
in the given mask.
Specifically, it'll clobber unset bits in the target that are not in the
mask, when the corresponding bit in the given new value is set.
Signed-off-by: James C Georgas <jgeorgas@rogers.com>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
