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>
snd_pcm_lib_preallocate_free_for_all() is called from snd_pcm_free() just
after calling the private_free routine. So there should be no need to call
it in driver's private_free routine.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
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>
Add support for monaural playback to at73c213 driver. The sound will be apear
on L-channel. Tested on AT91SAM9260-EK.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clk_round_rate() and clk_set_rate() will return int, so not store thier
return value to unsigned long variable. This bug hides real error on these
API.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
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>
if (...) BUG(); should be replaced with BUG_ON(...) when the test has no
side-effects to allow a definition of BUG_ON that drops the code completely.
The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@ disable unlikely @ expression E,f; @@
(
if (<... f(...) ...>) { BUG(); }
|
- if (unlikely(E)) { BUG(); }
+ BUG_ON(E);
)
@@ expression E,f; @@
(
if (<... f(...) ...>) { BUG(); }
|
- if (E) { BUG(); }
+ BUG_ON(E);
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
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>
- make pcsp_start_timer_tasklet static
- remove redundant includes. <asm/i8253.h> is not available on all platforms.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
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>
GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2
needs to be inverted. This patch adds a mixer control for this.
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>
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>
when the time interval for a period is smaller than kernel HZ, then
snd-aloop and snd-dummy cannot call snd_pcm_period_elapsed as fast enough
annymore. this happens for example with games. but the app still needs to
see, that the buffer actually did go further, which is provided by these
patches.
Signed-off-by: Ahmet İnan <ainan <at> mathematik.uni-freiburg.de>
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>
Reduce the number of times to check for a non-empty Tx FIFO from 100 to
2 because there is no MPU-401 implementation that needs more than one or
two reads to determine the actual FIFO status.
Signed-off-by: Clemens Ladisch <clemens@ladisch.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>
This patch adds __devinit to the functions used when probing. Will also reduce
the memory footprint a bit if CONFIG_HOTPLUG is not enabled.
Signed-off-by: Hans-Christian Egtvedt <hcegtvedt@atmel.com>
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>
This patch adds an ASoC driver for the WM9713 AC97 codec.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
