trivial patch to support the alsa module parameters `index', `id'
and `enable'
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/oxygen:
sound: virtuoso: add Xonar Essence ST support
sound: virtuoso: enable HDAV S/PDIF input
sound: virtuoso: add another DX PCI ID
sound: oxygen: reset DMA when stream is closed
* topic/misc:
ALSA: sgio2audio.c: clean up checking
ALSA: burgundy: timeout message is off by one.
ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'99
ALSA: parisc/harmony: fix printk format warning
ALSA: keywest: Get rid of useless i2c_device_name() macro
* topic/maya44:
ALSA: ice1724 - Add ESI Maya44 support
ALSA: ice1724 - Allow spec driver to create own routing controls
ALSA: ice1724 - Add PCI postint to reset sequence
ALSA: ice1724 - Clean up definitions of DMA records
ALSA: ice1724 - Check error in set_rate function
Currently the PCM resources are allocated only once and ever in prepare
callback, assuming that the PCM parameters are never changed. But it's
not true.
This patch adds the call of atc->pcm_release_resources() at hw_params
and hw_free callbacks to assure that the PCM setup is done correctly
for each h/w parameter changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SRC instances may not exist when PCM pointer callback is called at
the state before initialization is finished. Add the NULL check just
to be sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added use_system_timer module option to force to use the system timer
instead of emu20k1 timer irq for debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CTUAA should be checked instead of CTHENDRIX. The latter is for 20k2 chip.
Also, fixed the detection of UAA/HENDRIX models by fixing the mask bits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up probe routines and model detection routines so that the driver
won't call and check the PCI subsystem id at each time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added 7.1 support for MSI GX620 and jack quirk.
Reference: kernel bug#13430
http://bugzilla.kernel.org/show_bug.cgi?id=13430
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EFX playback stream should have periods_min = 2 to avoid the buffer
position overflow (due to restrictions of the pcm-indirect helper).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Clean up Hungarian coding style
- Don't use static variables for I2C information; this unables to use
multiple instances. Now they are stored in struct hw20k2 fields.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Optimize the timer update routine to look up wall clock once instead of
checking the position of each stream at each timer update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Use static tables instead of assigining each funciton pointer
- Add __devinit* to appropriate places; pcm, mixer and timer cannot be
marked because they are kept in the function table that lives long
- Move create_alsa_devs function out of struct ct_atc to mark it
__devinit
Signed-off-by: Takashi Iwai <tiwai@suse.de>
emu20k1 has a native timer interrupt based on the audio clock, which
is more accurate than the system timer (from the synchronization POV).
This patch adds the code to handle this with multiple streams.
The system timer is still used on emu20k2, and can be used also for
emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
with BIOS probing only we offer a non functional headphone swith and
volume slider.
Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAA-mode check in hwct20k1.c is implemented with the endian-dependent
codes. Fix to be more portable (and readable).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this build error when CONFIG_PM is not set:
ound/pci/hda/hda_intel.c: In function 'azx_bus_reset':
sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all'
sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend'
sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume'
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable all three capture channels, including the missing nid 7 which is
the only one capable of capturing DMIC input
Enable Headphone amp for the HP jack. This causes a volume boost for
headphones, but does not cause any noticeable effect for light loads
like other amps, so there is no need to make it configurable.
Add Input Mix capture mux setting to capture the output of the playback
input mux (that is, what goes out the speakers except for PCM)
Hack another coef register because the stereo DMIC for some reason
produces a nonstandard sum/difference signal. I found a bit to make it
just use the sum signal for both channels, which makes it behave like a
standard mono microphone. The stereo is useless anyway (they're 1cm apart).
Tested working: Three capture channels, mic in, line in, DMIC.
Tested not working: CD. Not sure why, might be unconnected in the actual
hardware or a CD drive issue.
Also looked at SPDIF. It appears to work (emitter lights up inside the
HP out jack) but I lack a proper miniTOSLINK cable to test it.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the limitation of PAGE_SIZE to be 4k by defining the own
page size and macros for 4k. 8kb page size could be natively supported,
but it's disabled right now for simplicity.
Also, clean up using upper_32_bits() macro.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device seems supporting only U8, S16, S24_3LE, S32. Other linear
formats result in bad outputs.
Also, added the support for 32bit float format, which wasn't listed
in the original code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM names for surround streams should be also fixed as well as the mixer
element names. Also, a bit clean up for PCM name setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We usually pick up "Surround" mixer for the rear output, and "Side"
for the extra surround. Fix the channel mapping to follow it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The prepare callback can be called multiple times, thus it needs to
release and acquire the resource again by itself at the second or later
call.
Simply add pcm_release_resources() at the beginning of each prepare
callback in ctatc.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playback
devices, but ca0106 and emu10k1x don't support it (unlike emu10k1).
We shouldn't set that flag to avoid confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the vmaster controls are created, they aren't registered thus
they don't appear in the real world. Added the missing snd_ctl_add()
calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Short story: this laptop has 5.1 built-in speakers which you *really*
want to use (the not-so-"sub" woofer is what makes the audio above
average for a laptop), so 6-channel support is important (plus a decent
asound.conf to upmix stereo). It also has the 3 typical jacks that ought
to have a selectable mode. And it's based on ALC889, which sucks.
Rationale/explanations:
The const_channel_count stuff was added because, for a laptop like this,
you always have 6 channels available (internal speakers) but still need
to set the mode for the 3 external jacks. Therefore, the device always
needs to be in 6-channel mode but there still needs to be a mixer
control for the jack mode. You could use line/mic-in at the same time as
the 6 internal speakers, for example. You might be tempted to make it
even smarter by dynamically switching the max channel count when
headphones are plugged in (therefore muting the internal speakers and
reducing the physical channel count to the jack channel mode), but as a
user I consider this to be harmful because I want the audio to blow up
to 6 channels / upmixed as soon as I unplug the headphones, and having
opened the device while in 2-channel mode would prevent this from
working (and always making 6-channel mode available doesn't do any harm).
The hardware needs EAPD turned on and the DACs routed to the internal
speaker pins, so the patch adds those verbs.
The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work
by default, at least here. I wasted much time trying to talk to
Realtek/pshou about this, but they just kept sending me useless updates
to patch_realtek.c that did nothing relevant. In the end I gave up and
brute forced the issue by trying to flip every bit in the proprietary
coefficient registers, and eventually found the two magic registers that
need to be cleared to enable all DACs. I have only heard Acer users
complain, but that might be because ALC889 is pretty new and using 5.1
(and noticing the missing center/lfe channels) might not be that common.
If this is a generalized issue with all ALC889 systems then those verbs
should probably be moved to a common verb array.
The internal mic is untested and probably doesn't work.
These settings will probably work for other Acer Gemstone laptops with
the same 5.1 speaker config. When identified, those should be added to
the PCI subsystem ID list.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reset of a BUS controller during operations is somehow risky and
shouldn't be done inevitably for devices that have apparently no such
codec-communication problems.
This patch adds the check of the hardware and limits the bus-reset
capability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines machine cause a severe CORB/RIRB stall in certain
weird conditions, such as PA access at the start up together with
fglrx driver. This seems unable to be recovered without the controller
reset.
This patch allows the bus controller reset at critical errors so
that the communication gets recovered again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, LG R510 is only able to produce sound on headphones, the
internal speakers are not working.
The user tested and confirmed that with model=Dell headphones,
internal speakers and the microphone are working flawlessly.
Tested-by: Serdar Soytetir <tulliana@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this is a patch against current snapshot that adds:
6 channels support for the MB5 model
Signed-off-by: Kacper Szczesniak <kacper@qwe.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix issues for 3 generations of HP workstations.
The modest modifications do the following:
1. Change the second MIC from device 3 to device 1
2. Init the "boost" values to "0" by default
From: John Brown <john.brown3@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes Line In as Out Switch and Mic In as Out Switch to
enums for consistency, and causes all mic and line in ports to be probed
and controls to be added appropriately.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ICH6_GCTL_RESET was wrongly set to another bit by the commit
b21fadb9c1. This caused a problem when
the codec needs really a reset (e.g. recovering from the communication
error at probe).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of mangling the CONFIG_* variables in the makefiles over and
over, set a few helper variables in Kconfig.
Signed-off-by: Michal Marek <mmarek@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The period_elapsed() call should be called when position moves.
The idea was taken from ALSA bug#4455.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Appearently, the used mask in the .pointer callback is invalid. It should
be in period_bytes range. The period_bytes is pow(2), so simple bitwise
operation is used.
Idea was taken from ALSA bug#4455.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added some missing register bits definitions to reduce magic numbers.
Also renamed some to follow the names on the datasheet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the hardware cannot store the multiple replies
like on RIRB, thus each verb has to sync and wait for the response no
matter whether the return value is needed or not. Otherwise it may
result in a wrong return value from the previous verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A docking mic control is shown by default. The Compaq Presario
CQ60 laptop has no docking connector, so designate it as a
CXT5051_HP model.
This makes the phantom mixer slider disappear.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add missing check of pin vref 50 and others in Realtek codecs
ALSA: hda - Add 5stack-no-fp model for STAC927x
ALSA: hda - Add forced codec-slots for ASUS W5Fm
This patch adds a debug mode to make the codec communication
synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c,
and the call of snd_hda_codec_write*() will become synchronous,
i.e. wait for the reply from the codec at each time issuing a verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the current driver code doesn't do any update
for RIRB just for any safety reason. But, actually the RIRB and
single_cmd mode don't conflict. Unsolicited events can be delivered
even while using the single_cmd mode.
This patch allows the handling of unsolicited events with single_cmd
mode, just always checking RIRB independent from single_cmd flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code doesn't handle the multiple gameports properly,
and uses unnecessary global static variables to store the data.
This patch changes the probe / remove routines to use the driver
data assigned to the dedicated pci device, and adds the support of
multiple devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A code clean up, coding style fixes.
The firmware loading routine is split to an own function to improve
the readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec communication error occurs, the CORB/RIRB counters should
be reset first before re-issuing the verb. Simply call azx_free_cmd_io()
and azx_init_cmd_io() to achieve that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Asus Xonar Essence ST and its daughterboard.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Xonar HDAV1.3 has a digital input jack, so enable the corresponding
device.
This is not related to the HDMI stuff, which stays unsupported.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another PCI ID for a second revision of the Xonar DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a PCM stream is closed, flush the corresponding DMA channel.
Otherwise, the DMA controller would continue to output the last sample
which would result in a DC offset on the output.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs like ALC861 seem to support only VREF50 while the
current driver assumes it's only VREF80. Check other VREF bits to set
the correct value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the headphone volume control on IDT/STAC codecs
resulted in the removal of invalid "Side" volume eventually. But,
if the front panel doesn't exist, this setup could be regarded as a
sort of regression, as reported in kernel bug #13250.
Now as a workaround, a new model 5stack-no-fp is added so that the user
without the front panel can choose this one explicitly.
Reference: bko#13250
http://bugzilla.kernel.org/show_bug.cgi?id=13250
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS
problem like W5F.
Tested-by: Alp Kılıç <kilic.alp@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase
Inversion Playback Switch' truncated to 'Sigmatel Surround Phase
Inversion Playback ' bootup message by omitting weird Sigmatel prefix
in this case; also fix up the related ca0106 mixer control removal
part by using identical naming there.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ctxfi driver requires explicitly the 4k page size, and gives a
build error on architectures with non-4k pages.
As a workaround, just add the kconfig dependency on X86, which is
the only architecture ever tested.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The digital beep tone is calculated in two different ways depending
on the codec chip. The standard one is using a divider, and another
one is a linear tone for IDT/STAC codecs. Currently, only the
latter type is used for all codecs, which resulted in a wrong tone
pitch.
This patch adds the calculation of the standard HD-audio type.
Also clean-up the fields in hda_beep struct.
Reference: bko#13162
http://bugzilla.kernel.org/show_bug.cgi?id=13162
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper prefix to each kernel message in hda_intel.c.
Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used
together with snd_print*().
Reference: bko#13207
http://bugzilla.kernel.org/show_bug.cgi?id=13207
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.
Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI entries of Creative with HD-audio class can be the devices
with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi
driver. With that driver, the device will mutate from HD-audio to
its native class.
This patch adds a simple ifdef to avoid the conflict of device probe
between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still
OK to be added as it has no emu20kx chip, and is a pure HD-audio
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split the name string in hda_codec struct to vendor_name and chip_name
strings to be stored directly from the preset name.
Since mostly only the chip name is referred in many patch_*.c, this
results in the reduction of many codes in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a mixer control for the STAC92XX boards to control the
DC bias of mic ports, allowing recording from both powered and
non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack
Mode" to switch between Mic, Line In, and Line Out.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the codec SSID for MacBook Pro 5,1 as compatible as MP51.
However, the headphone auto-muting function doesn't work. So,
this is just a tentative solution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix increment these variables are incremented beyond
CMDIF_TIMEOUT / MAX_WRITE_RETRY.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=acer for ALC883/889 doesn't work well for the recent Acer
Aspire laptops. Since model=auto works better nowadays, it's safer
to use the default fallback instead of the Acer specific one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hdsp driver is built in kernel, the corresponding firmware
files have to be built into the kernel as well (otherwise the boot
will hang up for fairly long time), but there is no way to control
it in Kconfig yet, unfortunately. Instead, show a comment for user
just to make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and
in-kernel driver.
Also corrected a typo in the comment.
Signed-off-by: Raphael Doursenaud <rdoursenaud@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add specific configuration for Samsung NC10 mini notebook. Internal
mic/speakers will be correctly muted when plugging in external ones.
Mixer controls are added for speakers, headphones and PC beep.
"Boost" is added for the microphones.
Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing descriptions and the model names for Realtek codecs
to the documentation and the config table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of unsol handlers defined in patch_realtek.c can be classified to
two types, mute via amp of pins and mute via ctl bits of pins.
Thus there are a big room to generalize each implementation.
This patch creates two generic functions, alc_automute_amp() and
alc_automute_pin(). The latter is actually changed from the previous
alc_sku_automute(). Each caller needs to initialize hp_pins and
speaker_pins properly at own init_hook.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The speaker auto-muting per HP plugging for ALC262 HIPPO and compatible
devices is slightly buggy as the "Master" or "Front" mixer control can
still toggle the speaker output even if the headphone is plugged.
This patch fixes the issue, and clean up the hippo-related codes
together with fixes of some inconsistent mixer names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of them
should have been handled as the extra digital out, but it's not
properly accessed.
This patch fixes the handling of the secondary SPDIF on these codecs
with the slave dig-out as found in patch_sigmatel.c. This makes the
use of such a device easier (for normal users).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on Mac Mini Core2 Duo sets both INPUT and OUTPUT pinctl bits to
the line-in jack, and it confuses the driver as if it's a valid input.
This patch adds the check of OUTPUT bit so that the driver fixes the
invalid pin setup.
Tested-by: Tino Keitel <tino.keitel@gmx.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new flag, own_routing, to allow spec drivers to create own
routing controls. Also, the basic get/put calls are changed to be
external for later use by maya44 driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The set_rate might return error but the current code doesn't check it.
This patch adds a proper error check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With 5 bits and 1.5 dB per step, the DXS volume range is only 48 dB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers that
really don't give the precise pointer value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commmit dfed0ef9b3 was reverted
accidentally by the merge of auto-detection fix patch.
Fixed again now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable auto-muting in model=auto only for devices with HP and speakers.
For devices with HP and line-outs, don't enable the auto-muting.
Also, add a debug print to show the auto-mute feature.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the realtek auto-probing mode, the initialization of amp with
some magic COEF or EAPD verbs is applied only when the codec SSID
has valid values to satisfy the realtek's definition.
However, many devices don't provide in that way, thus the device
doesn't work as is.
This patch allows the same initialization code even if the SSID
doesn't pass the bit test. Also, alc_subsystem_id() is changed
just to check and define the type, so that it's called in the
parser, instead of the initializer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the compile warning below by initializatin iomidi variable properly.
sound/pci/cmipci.c: In function ‘snd_cmipci_probe’:
sound/pci/cmipci.c:3017: warning: ‘iomidi’ may be used uninitialized in this function
Signed-off-by: Subrata Modak <subrata@linux.vnet.ibm.com>
Cc: linux-pci <linux-pci@atrey.karlin.mff.cuni.cz>
Cc: Balbir Singh <balbir@linux.vnet.ibm.com>
Cc: Sachin P Sant <sachinp@linux.vnet.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current ad1884a-mobile model has a problem that the speaker output
doesn't work sometimes after boot or power-saving on some HP laptops.
It seems that the verbs accessing to the non-functional widgets cause
this problem.
This patch simplifies the init verbs for mobile model not to touch
unnecessary setups so that it avoids the speaker-mute problem.
Reference: Novell bnc#495668
https://bugzilla.novell.com/show_bug.cgi?id=495668
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Appearently, a big delay ~300ms is required before hw is settled and ready
to transfer samples on some hardware variants. Also, return back
"clocking to 48000Hz" message when something fails.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The current error-recovery scheme for the codec communication errors
doesn't work always well. Especially falling back to the
single-command mode causes the fatal problem on many systems.
In this patch, the problematic verb is re-issued again after the error
(even with polling mode) instead of the single-cmd mode. The
single-cmd mode will be used only when specified via the command
option explicitly, mainly just for testing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cache quries for PCM and STREAM parameters as well as ampcap and
pincap sharing the hash table. This will reduce the superfluous
access of the same codec verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Support ASUS F81Se F5Q P80 U20A U80 U50 UX50 for ALC269
- Support ASUS F70SL UX20 X58LE F50Z N80Vc N81Te N505Tp Vx3V N5051A
for ALC663
- Support DELL ZM1 for ALC272
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't unmute unneeded amps for input mixers of ALC662 & co.
It caused possible recording noises.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing definition of max channels for CA0110, which resulted
in an error at opening PCM devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
(Re)set function_id only from the value on FG nodes.
The current code overrides the value with the last widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone can have no unique DAC, the current code doesn't
check the HP-detection although it should. Put the hp-detection check
before the DAC check to fix this bug.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the length to copy via strlen() beforehand to avoid the stack
corruption, or use strlcpy() to be safe in HD-audio codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode.
In the HD-audio mode, no multiple streams are supported by just it
behaves like a normal HD-audio device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subject says it all. Briefly, use hp_only for another Dell Inspiron 8600.
Reference: Ubuntu #41015 (https://launchpad.net/bugs/41015)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While cleaning up quirks, I noticed that there is a duplicated quirk for
the SSID 0x103c0934. Looking back through the bug reports, I've concluded
that there is only one necessary quirk (hp_mute_led), so this patch
removes the conflicting one.
Reference: Ubuntu #44066 (https://launchpad.net/bugs/44066)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit fa00e046b4
added a new bitfield not adjacent to other
bitfields in the same struct. Moved the new one.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the key value generation for get/set amp verbs. The upper bits of
the parameter have to be combined with the verb value to be unique for
each direction/index of amp access.
This fixes the resume problem on some hardwares like Macbook after
the channel mode is changed.
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/memdup_user:
ALSA: sound/pci: use memdup_user()
ALSA: sound/usb: use memdup_user()
ALSA: sound/isa: use memdup_user()
ALSA: sound/core: use memdup_user()
* 'master' of git://git.alsa-project.org/alsa-kernel:
[ALSA] intel8x0: add one retry to the ac97_clock measurement routine
[ALSA] intel8x0: fix wrong conditions in ac97_clock measure routine
[ALSA] intel8x0: do not use zero value from PICB register
[ALSA] intel8x0: an attempt to make ac97_clock measurement more reliable
[ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies
[ALSA] hda_intel: fix unexpected ring buffer positions
Added the models for quirk bitmask 1734:110x and 1734:113x of
Fujitsu laptops.
This will fix the model detection for Amilo Xa3540.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that on some hardware platforms, the first measurement is wrong.
This patch adds second measurement to this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Don't call snd_jack_report at release of sigmatel and conexnat codecs
which results in Oops at unloading the module.
The Oops is triggered by the power-up sequence during the free due to
the pincfg restoration. Since the power-up sequence is involved with
the unsol handling, the jack reporting may be issued during that.
The Oops occurs with this jack reporting because the jack instances
have been already released but the codec doesn't do the proper
book-keeping.
This patch adds the book-keeping of jack instances to avoid the access
to bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
prototype of a driver for the digigram lx6464es 64 channel ethersound
interface.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second go through of the old DMA_nBIT_MASK macro,and there're not
so many of them left,so I put them into one patch.I hope this is the last round.
After this the definition of the old DMA_nBIT_MASK macro could be removed.
Signed-off-by: Yang Hongyang <yanghy@cn.fujitsu.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: Tony Lindgren <tony@atomide.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Greg KH <greg@kroah.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
It seems that the zero value from the PICB (position in current buffer)
register is not reliable. Use jiffies to correct returned value
from the ring buffer pointer callback.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- use monotonic posix clock to measure time
- try to avoid reading zero from PICB (position in current buffer) register
- show also measured samples
- when clock is near 41000 or 44100, use exactly these values
(they appears to be reference clocks for hardware manufacturers)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I found two issues with ICH7-M (it should be related to other HDA chipsets
as well):
- the ring buffer position is not reset when stream restarts (after xrun) -
solved by moving azx_stream_reset() call from open() to prepare() callback
and reset posbuf to zero (it might be filled with hw later than position()
callback is called)
- irq_ignore flag should be set also when ring buffer memory area is not
changed in prepare() callback - this patch replaces irq_ignore with
more universal check based on jiffies clock
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both.
DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add
0x10 as slave output to the existing 0x06.
Reported-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Tested-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added device id in struct for codec 92HD81B1C (0x111d76d5).
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).
Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.
hda_proc.c is still using the direct verbs to get raw values as
much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change... Simply a thinko.
Also, add the similar checks to all places that put output-amp mutes
in the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
Output Intel HDA Function Id in /proc/asound/cardX/codec#X
Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros
Before:
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x100100
After:
Function Id: 0x1
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x0100100
As report on the Kernel Bugzilla #12888
Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode. The automatic mic switch via plugging
isn't implemented yet, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.
This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.
Also, clean a bit the code.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA. As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.
This patch removes the line-out volume control from vmixer-equipped cards.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.
Also, keep power-up during hwdep reconfiguration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.
Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin. This ensures that the pin
works somehow at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.
Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output. Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.
Reference: Novell bnc#482052
https://bugzilla.novell.com/show_bug.cgi?id=482052
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly. Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec. For this device, the model=auto must be chosen
to work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.
Reference: Novell bnc#480753
https://bugzilla.novell.com/show_bug.cgi?id=480753
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's false positive, but annoying.
sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
ALSA: hda - Add quirk for new HP xw series
ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.
For example, to disable hp_detect on the fly,
# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.
Internally, the hint is stored in a pair of two strings, key and val.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts
with the primary codec. The codec#3 is for the digital I/O, and should
be fixed by the driver, but it'd need a bunch of changes.
So, let's fix the probe problem temporarily by setting the default
probe_mask value.
Reference: kernel bugzilla #12735http://bugzilla.kernel.org/show_bug.cgi?id=12735
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to remove an unused variable.
sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’:
sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models
of IDT 92HD71bxx codec, which was wrongly set to zero.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Model drivers assume that model_data is zeroed, so we better use
kzalloc() (like we did before when it was allocated together with the
card structure).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the Device IDs for MCP89 HD audio controller.
Removed the IDs of MCP7B cause this chipset had been cancelled.
Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model=auto to STAC/IDT codecs to use the BIOS default setup
explicitly. It can be used to disable the device-specific model quirk
in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warning:
sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness)
sound/pci/hda/hda_codec.c:1544:19: expected unsigned long *vals
sound/pci/hda/hda_codec.c:1544:19: got long *<noident>
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warnings:
sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness)
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the pseudo device-locking using card->shutdown flag to avoid
the crash via clear/reconfig during operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the upper limit 0dB to the volume of mixer amp 0x20 for
AD1984A HP laptops. The overloaded volume may damage the internal
speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make user_pin overriding even the driver pincfg, e.g. the static / fixed
pin config table in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename from override_pin and cur_pin with user_pin and driver_pin,
respectively, to be a bit more intuitive.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sleeping for 2 seconds before checking for the iobox is not enough
on some systems.
this patch increases the timeout, but polls the card during that
time. it thus speeds up the module loading when the card has already
been initialized, while being more robust on systems, which require
a higher timeout than the predefined 2 seconds.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not
check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver
grabs them as well.
According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the
subsystem ids set to 0, the saa7146 default.
Fix conflicts with DVB devices by checking for subsystem ids = 0
specifically.
Signed-off-by: Anssi Hannula <anssi.hannula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the generic pincfg cache and save/restore functions.
Also introduced the pin-overriding via hwdep sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use -60 dB as the minimum value of the master volume mixer control.
While the DACs would support ranges down to about -120 dB, such
attenuations are not useful in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we
should handle the ADC parameters as we do with the X-Meridian.
Using the code for the wrong ADC does not seem to have any audible
effects, and the Windows driver does it, but it is nonetheless a good
idea to run the AK5385 with an oversampling ratio that is not outside
the documented limits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usage and comments make it clear values of 1/0 were intended
rather than -1/0
Noticed by sparse:
sound/pci/pcxhr/pcxhr.h💯20: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the EEPROM was partially overwritten (which seems to happen before the OS is
booted), restore its entire contents by deducing it from the remaining
information.
This does not have any effect on the Linux driver, which works even with
incomplete information in the EEPROM, but it makes other drivers work again.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Under as yet unknown circumstances, the first word of the sound card's
EEPROM gets overwritten. When this has happened, we cannot rely on the
subsystem IDs that the kernel reads from the PCI configuration
registers. Instead, we read the IDs directly from the EEPROM and do the
ID matching manually.
Because the model-specific driver cannot determine the model before
calling oxygen_pci_probe(), that function now gets a get_model()
callback as parameter. The customizing of the model structure, which
was formerly done by the probe() callback, also has moved into
get_model().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When allocating resources, use a fixed name instead of reading it from
the model structure. This allows us to allocate the resources before
the actual model is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the model-specific data dynamically instead of including it in
the memory block of the card structure. This will allow us to determine
the actual model after the card creation.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the owner field out of the oxygen_model structure and make it
a parameter of oxygen_pci_probe(), because the actual owner module does
not depend on the card model. Furthermore, moving it out of the model
structure allows us to create the card structure before the actual model
is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 7e86c0e685 ("do not
overwrite EEPROM on Xonar D2/D2X") because it did not actually help with
the problem.
More user reports show that the overwriting of the EEPROM is not
triggered by using this driver but by installing Linux, and that the
installation of any other operating system (even one without any CMI8788
driver) has the same effect. In other words, the presence of this
driver does not have any effect on the occurrence of the error. (So
far, the available evidence seems to point to a BIOS bug.)
Furthermore, it turns out that the EEPROM chip is protected against
stray write commands by the command format and by requiring a separate
write-enable command, so the error scenario in the previous commit (that
SPI writes can be misinterpreted as an EEPROM write command) is not even
theoretically possible.
The mixer control that was removed as a consequence of the previous
commit can only be partially emulated in userspace, which also means it
cannot be seen be the in-kernel OSS API emulation, so it is better to
revert that change.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC268 can be configured as digital-only, e.g. for HDMI, on some
machines. Allow the parser to set up the digital-only mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.
Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Martin Olsson
Thadeu Lima de Souza Cascardo
Tim Blechmann
Simos Xenitellis
Takashi Iwai
David Heidelberger
Guido Günther
Alexander Beregalov
Hector Martin
Ozan Çağlayan
Kacper Szczesniak
Jaroslav Kysela
Nickolas Lloyd
Wei Ni
Michal Marek
Tony Vroon
Linus Torvalds
Clemens Ladisch
Andreas Mohr
Adam Williamson
Torben Schulz
Roel Kluin
Wai Yew CHAY
Raphael Doursenaud
Chris Pockelé
James Gardiner
Giuliano Pochini
Subrata Modak
Kailang Yang
Daniel T Chen
Joe Perches
Li Zefan
Yang Hongyang
Matthew Ranostay
Akinobu Mita
Wu Fengguang
Mauro Carvalho Chehab
Pascal de Bruijn
Wolfram Sang
Jiri Slaby
Gregorio Guidi
Christoph Plattner
Herton Ronaldo Krzesinski
peerchen
Hannes Eder
Juan Jesus Garcia de Soria
Luke Yelavich
Anssi Hannula
Michael Schwingen
Harvey Harrison