Attached is a revised version of my patch to add AT32 to ASoC. This cleans
most of the style issues associated with the previous patch. Also fixes an
issue with the playpaq_wm8510.c code depending on a non-released patch to th
AT32 portmux support.
Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has
released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also
applied.
[Fixed up minor checkpatch issues and updated for current kernels -- broonie]
Signed-off-by: Geoffrey Wossum <gwossum@acm.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8990 is a highly integrated ultra-low power hi-fi codec designed
for handsets rich in multimedia features such as mobile TV, digital
audio playback and gaming.
The bulk of this driver was written by Liam Girdwood with some
additional development and updates for new ASoC APIs by me.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8510 is a mono CODEC with speaker driver optimised for telephony
applications, featuring:
- 16/20/24/32 bit audio at data rates between 8kHz and 48kHz
- On-chip PLL
- Dual microphone inputs
This driver was originally written by Liam Girdwood with updates from
Brett Saunders, Geoffrey Wossum and myself.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Brett Saunders <breton.saunders@ntlworld.com>
Signed-off-by: Geoffrey Wossum <geoffrey@pager.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
The UDA1380 codec is used by the HTC Magician and a number of Samsung
reference boards.
This driver has had a long out of tree history, having originally been
written by Giorgio Padrin and converted to ASoC by Richard Purdie.
Since conversion to ASoC most of the work on the driver has been done by
Philipp Zabel with some review and updates for new APIs by Liam Girdwood
and Mark Brown.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix and improve the slots option handling. The sound core tries to
find the slot with the given module name first and assign if it's
still available. If all pre-given slots are unavailable, then try
to find another free slot.
Also, when a module name begins with '!', it means the negative match:
the slot will be given for any modules but that one.
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>
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>
Added entry into usbquirks.h to recognize Roland SonicCell sound module by
mostly duplicating the entry for the Roland SH-201. USB MIDI works just fine,
though the USB audio is a little unreliable (but still works well enough).
Signed-off-by: Chris Mennie <camennie@alumni.uwaterloo.ca>
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>
This solves the problem with mixers wrongly displaying the PWM freq.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
There is a race from when a device is created with device_create() and
then the drvdata is set with a call to dev_set_drvdata() in which a
sysfs file could be open, yet the drvdata will be NULL, causing all
sorts of bad things to happen.
This patch fixes the problem by using the new function,
device_create_drvdata().
Cc: Kay Sievers <kay.sievers@vrfy.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@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>
Change HRTIMER_CB_IRQSAFE to HRTIMER_CB_SOFTIRQ,
as suggested by Thomas Gleixner.
That solves the lock dependancy reported in
Bug #10701.
That also allows to call hrtimer_start()
directly, tasklet "stupid hack" removed.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When timer ticks are disabled when calling
sound/soc/s3c24xx/s3c24xx-i2s.c:s3c24xx_snd_lrsync
and the LR signal never happens, we loop forever.
This has been observed in the following call chain:
snd_pcm_common_ioctl1 -> snd_pcm_action_lock_irq ->
snd_pcm_action_single
-> snd_pcm_do_resume -> soc_pcm_trigger -> s3c24xx_i2s_trigger
The patch below changes the timeout mechanism to use udelay, which
doesn't need timer ticks.
Signed-off-by: Werner Almesberger <werner@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.
Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.
Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The overwhelming majority just say 'initial version' anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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>
Use gpiolib since it is now available for OMAPs. Change also references to
HW version RX44 to product name N810.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC codecs and machine drivers that use DAPM routes all cut'n'paste a
loop iterating over a null terminated array of routes. Factor out this
into a bulk registration function, improving the error reporting for
most users, and deprecate the old API to help out of tree users pick up
the changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most SoC drivers cut'n'paste a loop iterating over an array to register
their DAPM controls. Provide a function they can call instead.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
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>
The hack for dma_alloc_coherent() is no longer needed on 2.6.26 since
the base code was improved.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Tim Niemeyer <reddog@mastersword.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This adds a hook to read the power state of a DAPM widget, I use this
in the gta02 driver to expose certain DAPM widgets in the mixer for
ease of audio routing.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds support for AIC3x GPIO lines. They can be configured for
many possible functions as well as be driven manually. I also introduced
i2c read functionality since the GPIO state register has to be read from
hardware every time and can not be served from cache.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch cleans up the clocking setup for aic3x codecs. It drops the
dividers table and determines the PLL control values programatically.
Under certain conditions, the PLL is disabled entirely which could save
some power.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
It appears that alsa allows a sound buffer with size not
evenly devided by the period size. This triggers a warning in
snd-pcsp and floods the log. As a quick fix, the warning should
be disabled.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considering all the feedbacks I got, depending snd-pcsp on
CONFIG_EXPERIMENTAL looks like the only safe way to get out
of all the troubles at one go. :)
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The attached patch adds back the compatibility code, allowing the
driver to work with older alsa-libs.
The removal was premature, it breaks the real-life configs.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the warning text to the help of snd-pcsp about the possible problem
with this driver so that user can know of the problem in advance.
Also, removed the obsoleted text about ancient pc-speaker patch in
CONFIG_SOUND help.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this typo and avoid similar errors by using ARRAY_SIZE macro.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hardware wavetable synthesizer of an Creative SB Audigy or SB
Live! card (with emu10k chip) receives the MIDI SOFT_PEADAL-press event
(?? 67 127) the appropriate voice is attenuted. Unfortunately when the
pedal is released (event ?? 67 0) the voice does not get it's original
volume again.
Boolean MIDI controls should interpret 0..63 as false and 64..127 as true.
Thanks to Clemens Ladisch for review and correction.
Original patch from "Uwe Kraeger" <uwe_debbug@arcor.de>
Submitted to http://bugs.debian.org/474312
Signed-off-by: maximilian attems <max@stro.at>
Cc: uwe_debbug@arcor.de
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
drivers/md/raid10.c:889:17: warning: Using plain integer as NULL pointer
drivers/media/video/cx18/cx18-driver.c:616:12: warning: Using plain integer as NULL pointer
sound/oss/kahlua.c:70:12: warning: Using plain integer as NULL pointer
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Cc: Neil Brown <neilb@suse.de>
Cc: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Found these two bugs while browsing through the code. The first one is
a cut-n-paste bug, instead of disabling the clock when request_irq()
fails, it enabled it once more. The second one fixes a debug printout,
AT91_SSC_IER is write only, AT91_SSC_IMR is readable (the printed string
actually says imr).
Frank Mandarino was busy so he asked me to send these to this list.
/Patrik
Signed-off-by: Patrik Sevallius <patrik.sevallius@enea.com>
Acked-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's more checkpatch stuff to fix in the driver, this just fixes the
minimum required for the following patch to be clean.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix pcspkr dependancies: make the pcspkr platform
drivers to depend on a platform device, and
not the other way around.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
CC: Vojtech Pavlik <vojtech@suse.cz>
CC: Michael Opdenacker <michael-lists@free-electrons.com>
[fixed for 2.6.26-rc1 by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
S3C2410 i2s driver currently manages only i2s protocol (and not left
justified one) and slave mode.
With this small patch, other modes are possible.
Signed-off-by: Davide Rizzo <davide@elpa.it>
Acked-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
S3C2410 pcm doesn't work.
s3c2410_dma_request() now returns the channel number and not 0 if OK.
Signed-off-by: Davide Rizzo <davide@elpa.it>
Acked-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
sound/drivers/pcsp/pcsp.c: In function 'pcsp_suspend':
sound/drivers/pcsp/pcsp.c:201: error: implicit declaration of function 'snd_pcm_suspend_all'
Signed-off-by: Johann Felix Soden <johfel@users.sourceforge.net>
CC: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Patch taken from the openmoko bugtracker
http://bugzilla.openmoko.org/cgi-bin/bugzilla/show_bug.cgi?id=781
This patch adds Suspend/Resume and Shutdown support for the lm4857 to
the driver.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I have moved workplaces since I originally wrote this driver so update
the contact info for new employers.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up a few issues with the file that checkpatch noted, no functionality
changes.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add common OMAP ASoC drivers and machine driver for Nokia N810. Currently
supported features are:
- Covers OMAPs from 1510 to 2420
- Common DMA driver
- DAI link driver using McBSP port in I2S mode
- Basic machine driver for Nokia N810
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers have duplicated unlikely() macros. IS_ERR() already has
unlikely() in itself.
This patch cleans up such pointless code.
Signed-off-by: Hirofumi Nakagawa <hnakagawa@miraclelinux.com>
Acked-by: David S. Miller <davem@davemloft.net>
Acked-by: Jeff Garzik <jeff@garzik.org>
Cc: Paul Clements <paul.clements@steeleye.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Cc: Alessandro Zummo <a.zummo@towertech.it>
Cc: David Brownell <david-b@pacbell.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Michael Halcrow <mhalcrow@us.ibm.com>
Cc: Anton Altaparmakov <aia21@cantab.net>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Carsten Otte <cotte@de.ibm.com>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Use proc_create() to make sure that ->proc_fops be setup before gluing PDE to
main tree.
Signed-off-by: Denis V. Lunev <den@openvz.org>
Cc: Alexey Dobriyan <adobriyan@gmail.com>
Cc: "Eric W. Biederman" <ebiederm@xmission.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>
Remove proc_root export. Creation and removal works well if parent PDE is
supplied as NULL -- it worked always that way.
So, one useless export removed and consistency added, some drivers created
PDEs with &proc_root as parent but removed them as NULL and so on.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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>
Also change some if (x == NULL) to if (!x).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some debug messages for suspend/resume and to add a clear prefix to
s3c24xx-i2s and s3c24xx-pcm.
Signed-off-by: Tim Niemeyer <reddog@mastersword.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Signed-off-by: Tim Niemeyer <reddog@mastersword.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PXA3xx does not support the use of interrupts during reset and access
to the GPIO status requires similar handling to that for PXA27x.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Since unsigned active_offs < 0 is even true when DMAbuf_get_buffer_pointer()
returns negative
Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs4270_hw_params does several times:
ret = snd_soc_write()
if (ret < 0)
...
This only works when ret is signed.
Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the race at reconnection of the device.
The disconnected usb_chip[] must be cleared before the next probe
call properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A little clean up of snd_card_free*().
Removed snd_card_free_prepare() since it's actually almost identical
with snd_card_disconnect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the call of device_unregister() for the card instance in
snd_card_disconnect() to avoid the race of sysfs card entry, which
can be typically found on usb-audio reconnection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer controls and correct headphone detection bits for PowerMac
G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset.
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable port change interrupt while initialising AWACS, Screamer, and
Burgundy chipsets.
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
bps is unsigned, a negative snd_pcm_format_width() return value is not noticed
Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
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 patch for snd_usb_caiaq makes sample rates higher dann 48KHz work
with devices which have more than 2 stereo input/output pairs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch corrects the input channel order of hardware supported by
snd_usb_caiaq.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes potential lockups in snd_usb_caiaq by refining the
locking mechanims and by using usb_kill_urb() in favor to
usb_unlink_urb().
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Leave the power bit for the touch screen alone when suspending the WM9713
so that the touch screen driver can handle it. This allows the touch
screen to be used as a wakeup source when the system is suspended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound: kernel log levels are 0-7
Kernel log levels are 0-7, not 0-9.
Signed-off-by: Nick Andrew <nick@nick-andrew.net>
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>
Since 43cc71eed1, the platform modalias is
prefixed with "platform:". Add MODULE_ALIAS() to the hotpluggable sound
platform drivers, to re-enable auto loading.
[dbrownell@users.sourceforge.net: more drivers, registration fixes]
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
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>
Putting space between ! and variable is a strange coding style, fix
that, also make it fit into 80 columns where that is easy.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb audio contains useful debugging code, protected by #if
0. Unfortunately, it will not compile because variable names changed;
fix it.
Dallas workaround is formatted in a way where it is not quite obvious
what is normal code and what is quirk. Reformat it to make it obvious.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dallas USB speakers are buggy in more than one way. One of configs
they offer does not work at all.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
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>
When logging register changes in DAPM debug output include the register
number.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
With commit 9004acc70e, include/sound/driver.h
is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
sound/drivers/pcsp/pcsp.c: In function 'snd_pcsp_create':
sound/drivers/pcsp/pcsp.c:54: error: 'loops_per_jiffy' undeclared (first use in\ this function)
sound/drivers/pcsp/pcsp.c:54: error: (Each undeclared identifier is reported on\ ly once
sound/drivers/pcsp/pcsp.c:54: error: for each function it appears in.)
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interrupt handler always provide runtime->period_size data, so it
works correctly only if buffer_size was a multiple of period_size.
This patch fixes periodic click noise.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
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>
We want to have snd_card_set_dev() in _probe(), but not a second one in
snd_ml403_ac97cr_create().
Signed-off-by: Joachim Foerster <JOFT@gmx.de>
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>
