The option bdl_pos_adj should be provided for each card instance instead of
a global one because the value depends rather on each controller-chip.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
position_fix=3 is the option to correct the DMA position with the
FIFO size. But, it never worked correctly, and we have now more other
workarounds for the DMA position fixes. Thus better to remove it.
Also, change POS_FIX_NONE to POS_FIX_LPIB to represent its real role
better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new option, bdl_pos_adj, to adjust the delay of IRQ-wakeup
timing.
Most HD-audio hardwares have a problem that a BDL IRQ is issued before
actually the data and the DMA pointer are updated.
We have already a mechanism to force to delay snd_pcm_period_elapsed()
calls via workq, but this costs much CPU, and typically the delay is
within one sample. Thus, it's more clever to adjust the BDL entries
instead.
The new option adds the size of the delay in frames. As default,
it's set to 1 -- that is, one sample delay. Even the hardware is
really correct, one sample delay is relatively harmless in comparison
with reporting wrong positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the missing GFP_ATOMIC to page_alloc when called with GFP_DMA.
GFP_KERNEL often results in stalls for ZONE_DMA, so GFP_ATOMIC is more
prgmatic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I have Toshiba dynabook SS RX1 and this patch adds that support.
Signed-off-by: Akio Idehara <zbe64533@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The power_save option was set as boot although it was meant to be a
timeout value like the same option of snd-hda-intel originally.
Now fixed to the same style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds support for Master Left Inv Switch on wm9711.
At least required to drive the mono speaker on the PXA270 platfrom
Signed-off-by: Juergen Beisert <j.beisert@pengutronix.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I think that hda_verb array must have "terminator (empty array)".
But alc262_sony_unsol[] does not have it.
And it causes gcc-4.3's buggy behavior
with snd_hda_sequence_write().
Signed-off-by: Akio Idehara <zbe64533@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the hw volume buttons work correctly on some HP OmniBook laptops.
The original quirk was apparently applied a bit too early and it was
also lacking some critial register writes. This improved sequence was
discovered by trial and error (like the original sequence). Tested and
found working on OB500 and OB6000 laptops.
Signed-off-by: Ville Syrjala <syrjala@sci.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that some ICH9-based boards use SD3 for the audio codec
where the current driver code doesn't probe. Since we have a better
codec slot check now, it must be safe to increase this to 4.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two motherboards's pin configuration are not covered by driver.
I wrote a new model to support them.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the Linux kernel is compiled with CONFIG_DEBUG_SHIRQ=y,
the Soundblaster Audigy2 ZS Notebook PCMCIA card causes the
system hang during boot (udev stage) or when the card is hot-plug.
The CONFIG_DEBUG_SHIRQ flag is by default 'y' with all Fedora
kernels since 2.6.23. The problem was reported as
https://bugzilla.redhat.com/show_bug.cgi?id=326411
The issue was hunted down to the snd_emu10k1_create() routine:
/* pseudo-code */
snd_emu10k1_create(...) {
...
request_irq(... IRQF_SHARED ...) {
register the irq handler
#ifdef CONFIG_DEBUG_SHIRQ
call the irq handler: snd_emu10k1_interrupt() {
poll I/O port // <---- !! system hangs
...
}
#endif
}
...
snd_emu10k1_cardbus_init(...) {
initialize I/O ports
}
...
}
The early access to I/O port in the interrupt handler causes
the freeze. Obviously it is necessary to init the I/O ports
before accessing them. This patch moves the registration of
the irq handler after the initialization of the I/O ports.
Signed-off-by: Jaroslav Franek <jarin.franek@post.cz>
Acked-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use quirk table to assign ALC268_TOSHIBA to COMPAL IFL90/JFL-92 laptops.
No analog output on autoprobe.
Signed-off-by: Tony Vroon <tony@linx.net>
Tested-by: Guri <gurashka@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Audigy2 Platinum, the Analog/Digital mixer switch is inverted.
https://bugzilla.novell.com/show_bug.cgi?id=396204
The patch adds a simple workaround.
There might be another device requiring a similar fix, too (or fix for
audigy2 generically), but right now I fix only the known broken one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mic pins are wrongly assigned on AD1884A mobile model.
The mic handling is fixed for the automatic mic selection, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the page allocation of emu10k1 driver for emux synth support.
Since these pages aren't be necessarily coherent, we can avoid
expensive DMA-coherent routines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ak4531 module is used only by ens1370 driver (and unlikely that
any other will use it ever). Let's make it local to ens1370.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed EAPD and COEF setups for Realtek ALC662/663, 660-VD and 888 codecs.
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show more exact codec chip name in the PCM stream name strings.
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of ALC663 codec, including specific models for
ASUS M51VA, ASUS G71V, ASUS H13 and ASUS G50V.
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing PCI ID for ICH9 controller (8086:2911)
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE to
represent its meaning more better. This config isn't provided only
for the detection but for more verbose debug prints in general.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The kconfig items related with AC97-powersave must be outside the
CONFIG_SND_PCI range. And it'd be better together with CONFIG_SND_AC97.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes CVS keywords that weren't updated for a long time
from comments.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix capture mute widget for STAC9250/9251 codecs. The widget 0x09
has no mute but 0x14 does actually.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Added IDs for the Foxconn P35AX-S mainboard to patch_realtek.c, so
that ALC883_6ST_DIG is used by default.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We get quite noisy output on the right channel on VT1708 codec
when 24bit samples are used. Suppress the 24bit support until any
real fix is found.
https://bugzilla.novell.com/show_bug.cgi?id=390473
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a config table entry for the ASUS P5K-E/WIFI-AP mainboard (ID
1043:8227) to use AD1988_6STACK_DIG
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The VT1724 MIDI port is not MPU-401 compatible; remove the hacks that
try to make the MPU-401 library work with it, and just use some simple
device-specific code.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Pavel Hofman <pavel.hofman@insite.cz>
Corrected the model assignment for the ASUS P5GD1 w/SPDIF after reports of
surround sound not being possible.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many HD-audio controllers seem inaccurate about the IRQ timing of
PCM period updates. This has caused problems on audio quality; e.g.
JACK doesn't work with two periods.
This patch fixes the problem by checking the current DMA position
at IRQ handler and delays the period-update via a workq if it's
inaccurate.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- figured out 'Digital(ly) Enhanced Game Port' functionality,
implemented support for it (eliminating gameport polling overhead)
- removed optional joystick activation, gameport now enabled unconditionally,
since we now support it via the PCI I/O space, not via conflict-prone
legacy I/O (which I was thus able to DISABLE now)!
- fix playback bug (a muted wave output would get unmuted upon start of
playback, of course this is not what we want, thus remember mute state)
- implement partial power management: when idle, lower clock rate and disable
codec (reduced noise!), and disable gameport circuit when unused
- instantiate OPL3 timer, too
- much better implementation of snd_azf3328_mixer_write_volume_gradually()
- slightly optimized interrupt handling
- lots of cleanup
This time, I also found a way to verify proper OPL3 operation
via MIDI file playback (emulation via synth hardware).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a constraint for the period time so that there are less than ten
seconds between interrupts so that ALSA does not assume that the device
is dead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Introduce symbols for the buffer/period size constraints so that their
limits and relationships become clearer.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Move the setting of the output enable GPIO bit to a separate function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Create separate functions for the code that initializes the hardware, as
opposed to initializing internal driver state, so that they can be
reused for resume support.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When initializing the DAC volume registers, we can just use the generic
volume update functions instead of setting the registers manually.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Save the written values of all CMI8788 and AC97 registers and of some of
the DAC/ADC registers so that it is possible to restore the register
state later.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Remove another magic number - add a symbol for the size of the PCI I/O
range.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Adjust the MODULE_LICENSE strings to properly reflect the actual license.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On boards with VT1617A codec, the sound disappears suddenly.
This looks like a problem with HPE-bit control that is supposed to be
set in patch_vt1617a(). However, on such problematic hardwares, the
bit is actually reset mysteriously.
The patch adds a workaround for the wrong quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FM801-tea575x tuner has a reverse selection to V4L1 and this causes
nasty dependency problems.
The patch simplifies the dependency with a normal
"depends on VIDEO_V4L1". This decreases the usability but fixes bugs,
yeah. If any better feature like "requires" is introduced to kbuild
in future, we'll be able to switch it...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sound boards with VT1724 and compatible chips may lock up when
MPU401 is accessed together with the PCM streaming.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The irq handler of PCI drivers must be released before releasing other
resources since the handler for a shared irq can be still called and
may access the freed resource again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
free_irq() calls synchronize_irq() for you, so there is no need for
drivers to manually do the same thing (again). Thus, calls where
sync-irq immediately precedes free-irq can be simplified.
However, during this audit several bugs were noticed, where free-irq is
preceded by a "irq >= 0" check... but the sync-irq call is not covered
by the same check.
So, where sync-irq could not be eliminated completely, the missing check
was added.
Signed-off-by: Jeff Garzik <jgarzik@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct some arguments in calls to snd_ice1712_gpio_write_bits() from
ap192_set_rate_val().
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some calls to snd_ice1712_gpio_write() go wrong, if
snd_ice1712_gpio_write_bits() ran before and changed the gpio mask register.
Read the actual gpio value and combine it with the to be set bits in the cpu
instead.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let "chip reset" become first. Increasement of the "chip reset" related timeout
leads to correctly read eeprom's contents here.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last ALC889A hack may break on some devices with certain model presets
since patch_alc*() have different model tables. So, now it's handled in
the original patch_alc882() but fly to patch_alc883() in model=auto
appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks,
the second headphone jack should be used for the 5.1 surround sound.
Add support for 'Headphone as Line Out' switch, which allows it be used
in 5.1 surround sound.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a pointer for DAC volume TLV data to the model structure so that the
model driver do not need to manually assign it in their control filter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the playback volume controls as being muted and having
minimal volume.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add fields for the DAC volume limits to the module structure so that
model drivers do not need to install their own control info handlers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The empty hifier_mixer_init() function is useless; remove it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of AD1989A and AD1989B codecs.
These codecs can have multiple SPDIF devices, but currently we handle
only one SPDIF. If any real devices with two SPDIF interfaces (likely
one for SPDIF and one for HDMI), we'll fix this rightly.
Otherwise, these codecs are pretty similar with AD1988.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the GPIO 1 mixer control to enable I/O through the front panel
connector of the Xonar DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for Quanta IL1 mini-notebook to alsa, defining a new model
for it. It comes with an ALC267 codec chip. Some notes about this model:
* In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common
amp mute, to avoid conflict with mixer switch (mixer and automute use the
same nid).
* The only connected capture sources in the hardware are the internal mic and
external mic jack. So instead of using an input source selector like on other
ALC268 models, the mic automute automatically switch between captures.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power management support for EAPD enabled laptops, when headphones
are sensed it pulls the EAPD GPIO line low to power it down.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several laptops have have the SPDIF out defined as 'Digital other out'
when it should be 'SPDIF out' in the default config.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy PC speaker signal was not routed to outputs. The codec is not
prevented from powering down in this patch, although I suppose one could
argue that perhaps it should be. Let me know if anyone feels strongly one
way or the other.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Please refer to [0003874] on the alsa mantis.
This patch added the pci quirk.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To mute the output of Pin widget 15 in ALC880, we should use the
HDA_OUTPUT. However, current code looks like :
snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
It may be a misspelling.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
WARNING: braces {} are not necessary for single statement blocks
#40: FILE: sound/pci/es1968.c:1831:
+ if (diff > 1) {
+ __maestro_write(chip, IDR0_DATA_PORT, cp1);
+ }
total: 0 errors, 1 warnings, 35 lines checked
./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
Please run checkpatch prior to sending patches
Cc: Andreas Mueller <andreas@stapelspeicher.org>
Tested-by: Rene Herman <rene.herman@keyaccess.nl>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of
course).
The patch is also incorporated in the *BSD drivers where I "ported" it from.
Without this patch most of the stereo audio gets out of sync and really
distorted (oss-emulation with mplayer at 48000khz worked somehow).
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/rme9652/hdspm.c has unusually large number of static inline
functions - 22.
I looked through them and some of them seem to be too big to warrant inlining.
This patch removes "inline" from these static functions (regardless of number
of callsites - gcc nowadays auto-inlines statics with one callsite).
Size difference on 32bit x86:
text data bss dec hex filename
20437 2160 516 23113 5a49 linux-2.6-ALLYES/sound/pci/rme9652/hdspm.o
18036 2160 516 20712 50e8 linux-2.6.inline-ALLYES/sound/pci/rme9652/hdspm.o
[coding fix by Takashi Iwai <tiwai@suse.de>]
Signed-off-by: Denys Vlasenko <vda.linux@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Please refer to [0003848] on the alsa mantis.
This patch adds the pci quirk and Mic-Int controller.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar DX, initialize all bits of the two-wire control register.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control for switching whatever it is that is connected to
GPIO pin 1 on the Xonar DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the card model does not have a digital input or an AC97 codec,
disable the respective interrupt and mixer controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When selecting the capture source on the Xonar DX, the input jack must
be routed to either the line input or the microphone input by setting a
GPIO pin. This requires an additional callback so that the model driver
can hook into the toggling of AC97 switches.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the card short name to show to show the card name instead of the
chip name.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When playing data at 96 kHz or higher, reduce the DAC oversampling rate
to 32.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the code that is common to all Xonar models to a separate function,
and make it more generic in preparation for another model.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename all CS5381 symbols to CS53x1 because they can also be used for
Xonar models with a CS5361.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use separate model structures for the D2 and D2X so that the init
function does not have to check for the model again.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"GPL 2" does not mean that there have to be two MODULE_LICENSE("GPL")
entries. ;-)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On C-Media cards, the GPIO pin 0 of the CM9780 must be handled exactly
like on Xonar cards, so move the Xonar code to the common mixer code.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for mic automute in clevo-m720r ALC883 model, and rename it
to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt
accordingly.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot one more: 3stack-hp model also have now the same mixer as
3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding
duplicating the same mixer definition.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the
same as alc883_base_mixer. Avoid duplicated code and use
alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer
definition.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After DAC assignment fix in ALC883, the 6stack-hp model is now the same
as 6stack-dig. So just remove 6stack-hp model and replace its use with
6stack-dig.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* moving most of clock-specific code to card-specific routines
* support for ESI Juli
* to-be-researched - monitoring of analog/digital inputs
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sparc64:
sound/pci/pcxhr/pcxhr.c: In function `pcxhr_update_r_buffer':
sound/pci/pcxhr/pcxhr.c:459: warning: cast to pointer from integer of different size
sound/pci/pcxhr/pcxhr.c: In function `pcxhr_trigger_tasklet':
sound/pci/pcxhr/pcxhr.c:628: warning: long int format, different type arg (arg 4)
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/pcxhr/pcxhr_core.c: In function `pcxhr_set_pipe_state':
sound/pci/pcxhr/pcxhr_core.c:899: warning: long int format, different type arg (arg 4)
suseconds_t is int on sparc64.
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sparc32:
sound/pci/aw2/aw2-alsa.c: In function 'snd_aw2_create':
sound/pci/aw2/aw2-alsa.c:282: error: 'DMA_32BIT_MASK' undeclared (first use in this function)
sound/pci/aw2/aw2-alsa.c:282: error: (Each undeclared identifier is reported only once
sound/pci/aw2/aw2-alsa.c:282: error: for each function it appears in.)
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the master clock outputs of any unused I2S inputs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the flag that enables the MIDI port into the model structure instead
of passing it as a separate parameter to oxygen_pci_probe().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the model drivers to specify if the codec communication goes over
SPI or a 2-wire bus.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When specifying which PCM devices to use, model drivers now use flags
that also specify the routing between PCM devices and DMA channels
instead of just DMA channel bits. This simplifies some code that checks
for these flags.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add controls to enable monitoring of the analog and digital inputs.
To allow monitoring after loading the driver when nothing has been
played back or recorded yet, the I2S input and outputs are initialized
to a valid configuration.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the PCM1796 register symbol definitions to their own header file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the WM8786 register symbol definitions to their own header file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep the format verb at closing PCM streams.
Introduced snd_hda_codec_cleanup_stream() for the parcicular purpose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Actually clfe and surround DACs are inverted in alc883_dac_nids array
(see ALC883 datasheet). I discovered this while testing multi-channel
setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that
has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped
in 6 channel mode (also in 4 channel mode you didn't get rear left/right
output). Other models also were affected by this bug, as can be seen by
the mixer layouts that "workaround" this (the real bug was not noticed,
and some other models simply played with mixer and initial verbs). Thus
along with fixing the order of dac nids, also change the models that
relied on previous dac ordering properly.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable is_capture is initialized but never used otherwise.
The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
type T;
identifier i;
constant C;
@@
(
extern T i;
|
- T i;
<+... when != i
- i = C;
...+>
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* removing the hack with NON_AKM ak4xxx type
* support for card-specific flags in ak4114_stats
* definition of the flags for corresponding cards
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AK4114 on Juli@ has the SPDIF input sample rate detection and
causes errors when an incompatible sample rate is chosen.
The patch adds the open hook to check the current rate and limit
the hw constraints.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The docking station headphone output had no audio and jack sense
was not considered.
Jack information from the laptop itself and the dock are combined, as
the dock does not obscure the connector.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hda-intel driver has a problem at power-off on ASUS P5AD2.
It's caused when the position-buffer is enabled -- most likely a
hardware-specific problem.
This patch adds a quirk to avoid the unnecessary enablement of
position-buffer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03,
0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to
mute?). The result is that currently mute for "Front Playback Switch",
"Surround Playback Switch", "Center Playback Switch" and "LFE Playback
Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard
with 3stack-6ch model).
The solution I found for this is to mute the proper inputs in 0x0c,
0x0d, 0x0e audio mixers.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the headphone controls are created as Master wrongly in
some cases, and this prevents the virtual master controls.
The patch fixes the problem by simply using "Headphone" always for
headphone controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed issue on some laptops that if the Master mixer and DAC mixers are
turned all the way up that will cause distortion. This is fixed by limiting
the max volume with the volume knob nid.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improved ALC262 ultra model for Samsung Q1 Ultra series.
- clean up mixers
- support of input from HP jack as a mic
- add quirk for Q1 EL
Signed-off-by: Takashi Iwai <tiwai@suse.de>
aw2-tsl.h should be rather a C file to be included since it's referred
only in aw2-saa6146.c and includes a table data.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default capture source should be the mic which is 0x01 on this model.
In addition to that the change to VREF50 allows for higher capture volume.
Signed-off-by: Michael Gruber <lists.mg@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed so that internal speakers point to the Front mixer instead of Surround.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Toshiba Equium L30 laptop and renames the mixer
controls to match Laptop usages.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the internal mic as a capture source item for ALC268 acer model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the Device IDs of nvidia MCP79 HD audio controller to hda_intel.c
Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the model laptop-hpsense use the 0x12 as ExtMic,
and use 0x14 as Internal IntMic.
But the hp530 only have one ExtMic, the Pin widget is 0x14.
In this patch, I changed the mixer item for them.
I still reserved the IntMic item, it will be helpful if
other machine may use this model.
ALSA bug#3821.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no suitable model for Pi2515.
This model is to support it. ALSA bug#3800.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
NEC S970 has no sound in the internal speakers when autodetection is
used.
With targa-dig model, there is sound in the speakers and it gets
correctly muted when pluging headphones.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no suitable model for M720R (ALSA bug#3781).
This patch is to support HP jack-sensing and mixer.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sparc32:
sound/pci/hda/hda_intel.c: In function 'azx_create':
sound/pci/hda/hda_intel.c:1838: error: 'DMA_64BIT_MASK' undeclared (first use in this function)
sound/pci/hda/hda_intel.c:1838: error: (Each undeclared identifier is reported only once
sound/pci/hda/hda_intel.c:1838: error: for each function it appears in.)
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added core_init[] for several 92hd73xxx laptops.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power management can't be enabled on fixed ports, since the presence
will always return false and prevent output.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the array declaration to hda_codec.c where it is used and add includes
where the individual presets are declared.
Fixes the following sparse warnings:
sound/pci/hda/patch_realtek.c:13744:25: warning: symbol 'snd_hda_preset_realtek' was not declared. Should it be static?
sound/pci/hda/patch_cmedia.c:729:25: warning: symbol 'snd_hda_preset_cmedia' was not declared. Should it be static?
sound/pci/hda/patch_analog.c:3656:25: warning: symbol 'snd_hda_preset_analog' was not declared. Should it be static?
sound/pci/hda/patch_sigmatel.c:3995:25: warning: symbol 'snd_hda_preset_sigmatel' was not declared. Should it be static?
sound/pci/hda/patch_si3054.c:286:25: warning: symbol 'snd_hda_preset_si3054' was not declared. Should it be static?
sound/pci/hda/patch_atihdmi.c:156:25: warning: symbol 'snd_hda_preset_atihdmi' was not declared. Should it be static?
sound/pci/hda/patch_conexant.c:1721:25: warning: symbol 'snd_hda_preset_conexant' was not declared. Should it be static?
sound/pci/hda/patch_via.c:1962:25: warning: symbol 'snd_hda_preset_via' was not declared. Should it be static?
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Temp variable in the loop shadows the second argument (which is otherwise
unused in this function). Change this to defcfg as it is used to hold
the default config.
sound/pci/hda/patch_sigmatel.c:2759:18: warning: symbol 'cfg' shadows an earlier one
sound/pci/hda/patch_sigmatel.c:2734:26: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In both cases we are passing around the substream number, use
sub_num for this.
sound/pci/riptide/riptide.c:1633:6: warning: symbol 'index' shadows an earlier one
sound/pci/riptide/riptide.c:121:12: originally declared here
sound/pci/riptide/riptide.c:1673:6: warning: symbol 'index' shadows an earlier one
sound/pci/riptide/riptide.c:121:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Inner loop redeclares err with u32 rather than int, stupid fix here
is to change the inner err to err2.
sound/pci/pcxhr/pcxhr_core.c:1008:8: warning: symbol 'err' shadows an earlier one
sound/pci/pcxhr/pcxhr_core.c:983:6: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use priv_idx as an identifier.
sound/pci/oxygen/virtuoso.c:277:15: warning: symbol 'index' shadows an earlier one
sound/pci/oxygen/virtuoso.c:56:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In all four case, adding a private value to the iooff index,
call it priv_idx.
sound/pci/ice1712/ice1712.c:1300:6: warning: symbol 'index' shadows an earlier one
sound/pci/ice1712/ice1712.c:85:12: originally declared here
sound/pci/ice1712/ice1712.c:1312:6: warning: symbol 'index' shadows an earlier one
sound/pci/ice1712/ice1712.c:85:12: originally declared here
sound/pci/ice1712/ice1712.c:1338:6: warning: symbol 'index' shadows an earlier one
sound/pci/ice1712/ice1712.c:85:12: originally declared here
sound/pci/ice1712/ice1712.c:1350:6: warning: symbol 'index' shadows an earlier one
sound/pci/ice1712/ice1712.c:85:12: originally declared here
[tiwai - fixed coding issues as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
enable in these contexts refers specifically to intr enable, as
per the two functions it is found in. Use intr_enable instead.
sound/pci/emu10k1/emu10k1x.c:330:15: warning: symbol 'enable' shadows an earlier one
sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here
sound/pci/emu10k1/emu10k1x.c:341:15: warning: symbol 'enable' shadows an earlier one
sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here
instead of shadowing, use cap_voice as we test for the capture
voice in this statement.
sound/pci/emu10k1/emu10k1x.c:798:25: warning: symbol 'pvoice' shadows an earlier one
sound/pci/emu10k1/emu10k1x.c:787:24: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reading regs from the fpga into an int instead of a u32, trivial
fix.
sound/pci/emu10k1/emuproc.c:422:34: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emuproc.c:422:34: expected unsigned int [usertype] *value
sound/pci/emu10k1/emuproc.c:422:34: got int *<noident>
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/au88x0/au88x0_pcm.c:508:15: warning: Using plain integer as NULL pointer
Also some small codingstyle fixes.
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the variable err to _err within the ADD_CTLS macro to avoid
shadowing the local variable.
sound/pci/ca0106/ca0106_mixer.c:710:2: warning: symbol 'err' shadows an earlier one
sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here
sound/pci/ca0106/ca0106_mixer.c:712:3: warning: symbol 'err' shadows an earlier one
sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here
sound/pci/ca0106/ca0106_mixer.c:721:3: warning: symbol 'err' shadows an earlier one
sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
change to intr_enable as per the two functions it is defined in.
sound/pci/ca0106/ca0106_main.c:438:15: warning: symbol 'enable' shadows an earlier one
sound/pci/ca0106/ca0106_main.c:159:12: originally declared here
sound/pci/ca0106/ca0106_main.c:449:15: warning: symbol 'enable' shadows an earlier one
sound/pci/ca0106/ca0106_main.c:159:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
enable is used to test for whether or not spdif should be enabled,
change to spdif_enable.
sound/pci/ali5451/ali5451.c:1812:15: warning: symbol 'enable' shadows an earlier one
sound/pci/ali5451/ali5451.c:63:12: originally declared here
sound/pci/ali5451/ali5451.c:1840:27: warning: symbol 'enable' shadows an earlier one
sound/pci/ali5451/ali5451.c:63:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
err is always assigned before it is used, no need to declare another
inside the if statement.
sound/pci/ac97/ac97_pcm.c:577:7: warning: symbol 'err' shadows an earlier one
sound/pci/ac97/ac97_pcm.c:572:6: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kernel style does assignment outside of if() block
sound/pci/rme96.c:1562:71: warning: Using plain integer as NULL pointer
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kernel style does assignment outside of if() statements.
sound/pci/rme32.c:1353:71: warning: Using plain integer as NULL pointer
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
change id to elem_id as it is used to initialize each mixer element
sound/pci/maestro3.c:2071:25: warning: symbol 'id' shadows an earlier one
sound/pci/maestro3.c:67:13: originally declared here
index is used in each of these places to count over the dsp's memory,
change to the name dsp_index
sound/pci/maestro3.c:2572:9: warning: symbol 'index' shadows an earlier one
sound/pci/maestro3.c:66:12: originally declared here
sound/pci/maestro3.c:2604:9: warning: symbol 'index' shadows an earlier one
sound/pci/maestro3.c:66:12: originally declared here
[tiwai - fixed coding style issues as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
id was only used as a counter in a for loop, move the declaration
to where it is used and change it to i.
sound/pci/fm801.c:1288:6: warning: symbol 'id' shadows an earlier one
sound/pci/fm801.c:51:13: originally declared here
[tiwai - fixed a coding style issue as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
id is used when initializing the mixer elements, use elem_id here
instead.
sound/pci/es1968.c:1963:25: warning: symbol 'id' shadows an earlier one
sound/pci/es1968.c:129:13: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
index is incremented only when AC97_EI_SPDIF and then assigned to
the index field. Change the temporary name to is_spdif.
sound/pci/ens1370.c:1638:10: warning: symbol 'index' shadows an earlier one
sound/pci/ens1370.c:84:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A temporary variable for each mixer element is used in an initialization
loop, use the name elem_id.
sound/pci/cmipci.c:2747:26: warning: symbol 'id' shadows an earlier one
sound/pci/cmipci.c:56:13: originally declared here
[tiwai - fixed a coding style issue as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bits indicating SPDIF frequency in the status register are not the same for
the 9632 than for the other cards, because it also supports 192kHz. A specific
bitmask has thus been added (used in hdsp_spdif_sample_rate()).
The 9632 does not seem to report external sample rates greater than 96kHz. In
this case, the best seems to report spdif rate when autosync reference is
spdif. This also required to move function hdsp_spdif_sample_rate().
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added logic to check if AUTO_PIN_FRONT_MIC is available for output
switch, if AUTO_PIN_MIC isn't.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops have a internal analog microphone that is not setup by the BIOS.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC_DELL_BIOS quirks were setting the association value wrong
for port 0x0f, which prevented it from being included in hp_outs[].
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix issue on STAC927x codecs that first DAC was getting powered down
even if was being used.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the capture source properly for auto model.
It's especially important for cases that only mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the beep volume control to ALC268 codec support code.
Since the codec doesn't return the correct AMP caps, we need to override
the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a private array for TLV entries of virtual master controls instead
of (supposed) static array. This cleans up the existing codes.
Also, now vmaster assumes the simple dB-range TLV that is the only type
it can handle.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the codes for virtual master controls to sound core part so that
not only hda-intel drivers can use it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of 8 channel sound for codecs that are known to work.
So far, only ALC850 is marked as a 8ch-support codec.
This fix is a modified version of the patch on ALSA BTS#2097 by
Martin Ellis:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of new AD codecs: AD1883, AD1884A, AD1984A and AD1984B.
These are almost compatible except for additional digital pins, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC262 must have capsrc_nids defined as well as in ALC882.
Also, add a NULL check in alc882_mux_enum_put to avoid Oops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last patch for fixing the auto-config pin setting breaks the resume
due to a wrong use of snd_hda_codec_amp_stereo(). The code in the init
hook shouldn't touch the amp cache.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new mixer switch to enable/disable the sharing of the default
PCM stream with analog and SPDIF outputs. When "IEC958 Default PCM"
switch is on, the PCM stream is routed both to analog and SPDIF outputs.
This is the behavior in the earlier version.
Turning this switch off has a merit for some codecs, though. Some codec
chips don't support 24bit formats for SPDIF but only for analog outputs.
In this case, you can use 24bit format by disabling this switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes some bugs in the auto-configurator of Realtek codecs:
- add missing pin set-up for speaker pins
- fix the speaker auto-mute function not to conflict with the existing
"Speaker" mixer switch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, the BIOS sets up only the HP pins with different assoc
and sequence numbers, e.g. on FSC Esprimo with ALC262.
This patch adds a fix-up for such a case. When multiple HPs are defined
and no line-outs is found, the configurator tries to re-assign some pins
from HP list to line-out, judging from the sequence number.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implemented the auto-mic jack sensing for Samsung laptops with AD1986A
codec chip (model=laptop-eapd).
The hardware uses pin 0x1d and 0x1f for the internal and external
mics, respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the codes of the capture source selection for Realtek codecs.
Now using common helper functions with the new capsrc_nids field.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reprogram the speaker-pin setting at each HP pin plug to make sure
the spekaer auto-muting on AD1981HD hp model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've just noticed that there are a handful of duplicate controls in the
ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes
them.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
See ALSA bug#3327 for more details. Experimental.
Also fix support for M-Audio Delta 1010E - subdevice check.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The HD-audio hardware usually supports 64bit address for DMA and other
buffers. The patch enables the feature if supported.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On several laptops that have STAC9228 codecs have unused DACs,
this powers them down to a D3 state.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the problems with Midi In on Hoontech/STA dsp24 cards, for example with
DSP2000 box, without restricting the box configurations available. Also adds
mpu_401 name strings.
Signed-off-by: Alan Horstmann <gineera@aspect135.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current scheme, PCM device numbers are assigned incrementally
in the order of codecs. This causes problems when the codec number
is irregular, e.g. codec #0 for HDMI and codec #1 for analog. Then
the HDMI becomes the first PCM, which is picked up as the default
output device. Unfortuantely this doesn't work well with normal
setups.
This patch introduced the fixed device numbers for the PCM types,
namely, analog, SPDIF, HDMI and modem. The PCM devices are assigned
according to the corresponding PCM type. After this patch, HDMI will
be always assigned to PCM #3, SPDIF to PCM #1, and the first analog
to PCM #0, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code doesn't allow multiple SPDIF devices, and causes
errors when multiple SPDIF devices are found (e.g. SPDIF out and HDMI).
This patch allows multiple SPDIF devices by incrementing the index
automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the OQO Model 2 Ultra Mobile PC.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The C99 specification states in section 6.11.5:
The placement of a storage-class specifier other than at the
beginning of the declaration specifiers in a declaration is an
obsolescent feature.
Signed-off-by: Tobias Klauser <tklauser@distanz.ch>
Signed-off-by: Jesper Juhl <jesper.juhl@gmail.com>
snd_es1968_ac97_read() calls snd_es1968_ac97_wait() first outside a locked
area, and later, while holding a lock.
snd_es1968_ac97_wait() has a polling loop with a cond_resched() inside it..
which sleeps, so the second call is invalid.
This patch adds a version of the wait function that just pure polls. While
this is not very elegant in principle, it's very likely the easiest thing to
do here, we already checked if the chip was ready (while yielding) just
before, so it is very unlikely to take a long time here.
[akpm@linux-foundation.org: coding-style fixes]
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Cc: Jaroslav Kysela <perex@suse.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix the line-out volume control of eeepc p701 to be a proper slave of
the virtual master control.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't need to declare a struct when defining a structure layout. Both
of these structs are unused.
sound/pci/ice1712/revo.c:39:3: warning: symbol 'revo51' was not declared. Should it be static?
sound/pci/ice1712/phase.c:54:3: warning: symbol 'phase28' was not declared. Should it be static?
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models like eeepc ep20 have invalid mixer names that aren't
handled properly by virtual master controls. Rename them to the
proper names.
Also fixed some typos in the mixer names but they are not compiled
in right now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a quirk to the Realtek ALC883 table for the Albatron KI690-AM2
motherboard to use the 6stack-dig model.
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to set the module owner for the HiFier/Xonar models.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP dv8000 laptop has a problem with Master volume. It's due to the
connection of the widget 0x13. When it's connected from the analog
amp mixer (0x19), it works as expected mysteriously (ALSA bug#3775):
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3775
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call free_irq() after iounmap() because other devices could trigger our
shared interrupt handler.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The driver gets Oops with ATI HDMI devices due to the wrong calculation
of index for playback streams. This patch fixes it. Reference:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3746
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pin widgets have always one amp-input value regardless of number of
connections. The proc file showed values wrongly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The capture source selection for ADC list with two elements is buggy
becaues of a wrong capture mux list. This patch fixes the starting
index based on spec->num_adc_nids.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The widget list of capture source selection for ALC883 contains the
wrong NIDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Don't create vmaster controls if no slaves are found in the given list.
This prevents the error due to an empty vmaster control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The GPIO pin 0 of the CM9780 must be set when muting the line input even
on non-Xonar cards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fixed the SPDIF output on Conexant Cx5045 codec. Added the missing
pin output setting and fixed the wrong NID for digital audio-out widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Takashi Iwai
Akio Idehara
Juergen Beisert
Ville Syrjälä
Jiang zhe
Jaroslav Franek
Tony Vroon
Kailang Yang
Adrian Bunk
Mauro Carvalho Chehab
Travis Place
Clemens Ladisch
Andreas Mohr
Daniel Jacobowitz
Jacek Luczak
Peter Lienig
Jeff Garzik
Karsten Wiese
Matthew Ranostay
Herton Ronaldo Krzesinski
Frederik Deweerdt
Andrew Morton
Andreas Mueller
Denys Vlasenko
Pavel Hofman
Julia Lawall
Takashi Iwai
Michael Gruber
Tobin Davis
Peer Chen
Pascal Terjan
Harvey Harrison
Remy Bruno
Cedric Bregardis
Libin Yang
Jonathan Woithe
Jaroslav Kysela
Alan Horstmann
Mirco Tischler
Tobias Klauser
Arjan van de Ven
Andrew Paprocki
Serge A. Suchkov