Optimize the timer update routine to look up wall clock once instead of
checking the position of each stream at each timer update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 13f040f9e5 made another
regression, the missing update of runtime->hw_ptr_interrupt.
Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt(),
not in snd_pcm_update_hw_ptr(), it must be updated before the hw_ptr
change check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
vfree() does it's own 'NULL' check,so no need for check before
calling it.
Signed-off-by: Figo.zhang <figo1802@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a typo in the commit 13f040f9e5
ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
which causes obvious problems with PA.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some code analyzer software mistakenly gives
divide by 0 error messages for these lines.
This patch will end its confusion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Use static tables instead of assigining each funciton pointer
- Add __devinit* to appropriate places; pcm, mixer and timer cannot be
marked because they are kept in the function table that lives long
- Move create_alsa_devs function out of struct ct_atc to mark it
__devinit
Signed-off-by: Takashi Iwai <tiwai@suse.de>
emu20k1 has a native timer interrupt based on the audio clock, which
is more accurate than the system timer (from the synchronization POV).
This patch adds the code to handle this with multiple streams.
The system timer is still used on emu20k2, and can be used also for
emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
with BIOS probing only we offer a non functional headphone swith and
volume slider.
Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAA-mode check in hwct20k1.c is implemented with the endian-dependent
codes. Fix to be more portable (and readable).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case the initalization of an soc_device failed, there is no codec
associated with it. soc_suspend() will still dereference the pointer
and cause an Ooops when entering the sleep mode.
This happens on our board with a multi-target kernel image when booted
on a machine without audio circuits.
This patch makes the code bail out very early in this special case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix this build error when CONFIG_PM is not set:
ound/pci/hda/hda_intel.c: In function 'azx_bus_reset':
sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all'
sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend'
sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume'
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
3 ISA sound drivers lack their __devexit_p() markers, which would
cause build failures when the kernel is built without hotplug support.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the (likely cut-n-paste) error by commit
16a30fbb0d, which causes the error below:
sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache':
sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move locking outside snd_card_set_id_internal() function and rename it
to snd_card_set_id_no_lock() for better function description.
User defined id is just copied to card structure at allocation time.
The real unique id procedure is called in snd_card_register() to
ensure real atomicity.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable all three capture channels, including the missing nid 7 which is
the only one capable of capturing DMIC input
Enable Headphone amp for the HP jack. This causes a volume boost for
headphones, but does not cause any noticeable effect for light loads
like other amps, so there is no need to make it configurable.
Add Input Mix capture mux setting to capture the output of the playback
input mux (that is, what goes out the speakers except for PCM)
Hack another coef register because the stereo DMIC for some reason
produces a nonstandard sum/difference signal. I found a bit to make it
just use the sum signal for both channels, which makes it behave like a
standard mono microphone. The stereo is useless anyway (they're 1cm apart).
Tested working: Three capture channels, mic in, line in, DMIC.
Tested not working: CD. Not sure why, might be unconnected in the actual
hardware or a CD drive issue.
Also looked at SPDIF. It appears to work (emitter lights up inside the
HP out jack) but I lack a proper miniTOSLINK cable to test it.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The introduction of snd_card_set_id() added a lock on the card list
to the old choose_default_id() function when using it to implement
the new API call. This lock is needed to allow us to walk the list
and check to see if our new name is a duplicate. Unfortunately this
causes a lockup when called from snd_card_register() (in cases
where no ID is supplied for the card) since the card list is already
locked there.
Fix this fairly hideously by factoring out the implementation and
using a flag to indicate if the lock should be held. A better fix
would probably be to refactor snd_card_register() to move the
_set_id() outside the locking region but I can't immediately see
anything I can convince myself is safe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes crash when shutting down.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
[I am not sure if this is the correct approach as I don't know if any of
this actual hardware or drivers are really hot pluggable.]
Gets rid of these build warnings:
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_new().
If .snd_pmac_new is only used by .snd_pmac_probe then
annotate .snd_pmac_new with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_burgundy_init().
If .snd_pmac_burgundy_init is only used by .snd_pmac_probe then
annotate .snd_pmac_burgundy_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_daca_init().
If .snd_pmac_daca_init is only used by .snd_pmac_probe then
annotate .snd_pmac_daca_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_init().
If .snd_pmac_tumbler_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_post_init().
If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_post_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_awacs_init().
If .snd_pmac_awacs_init is only used by .snd_pmac_probe then
annotate .snd_pmac_awacs_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_pcm_new().
If .snd_pmac_pcm_new is only used by .snd_pmac_probe then
annotate .snd_pmac_pcm_new with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_attach_beep().
If .snd_pmac_attach_beep is only used by .snd_pmac_probe then
annotate .snd_pmac_attach_beep with a matching annotation.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cut'n'paste mistake, whose likely result was nothing at all.
Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC".
Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the limitation of PAGE_SIZE to be 4k by defining the own
page size and macros for 4k. 8kb page size could be natively supported,
but it's disabled right now for simplicity.
Also, clean up using upper_32_bits() macro.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device seems supporting only U8, S16, S24_3LE, S32. Other linear
formats result in bad outputs.
Also, added the support for 32bit float format, which wasn't listed
in the original code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM names for surround streams should be also fixed as well as the mixer
element names. Also, a bit clean up for PCM name setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We usually pick up "Surround" mixer for the rear output, and "Side"
for the extra surround. Fix the channel mapping to follow it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The prepare callback can be called multiple times, thus it needs to
release and acquire the resource again by itself at the second or later
call.
Simply add pcm_release_resources() at the beginning of each prepare
callback in ctatc.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playback
devices, but ca0106 and emu10k1x don't support it (unlike emu10k1).
We shouldn't set that flag to avoid confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If not passed as module option, provide an own card ID with the newly
introduced snd_set_card_id() call.
This will prevent ALSA from calling choose_default_name() which only
takes the last part of a name containing whitespaces. This for example
caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very
descriptive.
The implementation now takes the short name and removes all whitespaces
from it which is much nicer.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce snd_card_set_id() function to allow lowlevel drivers to set
default identification name for card slot. The function checks also
for identification name collisions and tries to create unique name.
Also, the snd_card_create() function is simplified, because this new
function is used. As bonus, proper name collision checks are evaluated
at the card create time.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the vmaster controls are created, they aren't registered thus
they don't appear in the real world. Added the missing snd_ctl_add()
calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Short story: this laptop has 5.1 built-in speakers which you *really*
want to use (the not-so-"sub" woofer is what makes the audio above
average for a laptop), so 6-channel support is important (plus a decent
asound.conf to upmix stereo). It also has the 3 typical jacks that ought
to have a selectable mode. And it's based on ALC889, which sucks.
Rationale/explanations:
The const_channel_count stuff was added because, for a laptop like this,
you always have 6 channels available (internal speakers) but still need
to set the mode for the 3 external jacks. Therefore, the device always
needs to be in 6-channel mode but there still needs to be a mixer
control for the jack mode. You could use line/mic-in at the same time as
the 6 internal speakers, for example. You might be tempted to make it
even smarter by dynamically switching the max channel count when
headphones are plugged in (therefore muting the internal speakers and
reducing the physical channel count to the jack channel mode), but as a
user I consider this to be harmful because I want the audio to blow up
to 6 channels / upmixed as soon as I unplug the headphones, and having
opened the device while in 2-channel mode would prevent this from
working (and always making 6-channel mode available doesn't do any harm).
The hardware needs EAPD turned on and the DACs routed to the internal
speaker pins, so the patch adds those verbs.
The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work
by default, at least here. I wasted much time trying to talk to
Realtek/pshou about this, but they just kept sending me useless updates
to patch_realtek.c that did nothing relevant. In the end I gave up and
brute forced the issue by trying to flip every bit in the proprietary
coefficient registers, and eventually found the two magic registers that
need to be cleared to enable all DACs. I have only heard Acer users
complain, but that might be because ALC889 is pretty new and using 5.1
(and noticing the missing center/lfe channels) might not be that common.
If this is a generalized issue with all ALC889 systems then those verbs
should probably be moved to a common verb array.
The internal mic is untested and probably doesn't work.
These settings will probably work for other Acer Gemstone laptops with
the same 5.1 speaker config. When identified, those should be added to
the PCI subsystem ID list.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The serial number is of no interest in the longname, remove it. This
gives space for the usb path information which is more informative.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reset of a BUS controller during operations is somehow risky and
shouldn't be done inevitably for devices that have apparently no such
codec-communication problems.
This patch adds the check of the hardware and limits the bus-reset
capability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines machine cause a severe CORB/RIRB stall in certain
weird conditions, such as PA access at the start up together with
fglrx driver. This seems unable to be recovered without the controller
reset.
This patch allows the bus controller reset at critical errors so
that the communication gets recovered again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, LG R510 is only able to produce sound on headphones, the
internal speakers are not working.
The user tested and confirmed that with model=Dell headphones,
internal speakers and the microphone are working flawlessly.
Tested-by: Serdar Soytetir <tulliana@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this is a patch against current snapshot that adds:
6 channels support for the MB5 model
Signed-off-by: Kacper Szczesniak <kacper@qwe.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In addition to the operating mode check, also check the
codec's interface format in case of four channel mode.
If the codec is not in TDM (DSP_A) mode, return with error.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix issues for 3 generations of HP workstations.
The modest modifications do the following:
1. Change the second MIC from device 3 to device 1
2. Init the "boost" values to "0" by default
From: John Brown <john.brown3@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes Line In as Out Switch and Mic In as Out Switch to
enums for consistency, and causes all mic and line in ports to be probed
and controls to be added appropriately.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a half-working quirk for Roland/Edirol M-16DX.
This enables the capture on the device but the playback on it seems still
problematic becuase of lack of sync with the capture clock.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII
USB card.
Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ICH6_GCTL_RESET was wrongly set to another bit by the commit
b21fadb9c1. This caused a problem when
the codec needs really a reset (e.g. recovering from the communication
error at probe).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the codec->reg_cache instead of the array directly
in twl4030_init_chip for setting the default values.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of mangling the CONFIG_* variables in the makefiles over and
over, set a few helper variables in Kconfig.
Signed-off-by: Michal Marek <mmarek@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When hardware has large FIFO, it is necessary to lower jiffies margin
by count of queued samples.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some hardware might have bigger FIFOs and DMA pointer value will be updated
in large chunks. Do not update hw_ptr_jiffies and position timestamp when
hw_ptr value was not changed.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For debugging purposes, it is better to separate actions.
Bit-values:
1: show bad PCM ring buffer pointer
2: show also stack (to debug kernel latency issues)
4: check pointer against system jiffies
Example:
5: show bad PCM ring buffer pointer and do jiffies check
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fifo_size assignment to hw->ioctl callback to allow lowlevel
drivers overwrite the default behaviour.
fifo_size is in frames not bytes as specified in asound.h and alsa-lib's
documentation, but most hardware have fixed byte based FIFOs. Introduce
internal SNDRV_PCM_INFO_FIFO_IN_FRAMES.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The period_elapsed() call should be called when position moves.
The idea was taken from ALSA bug#4455.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Appearently, the used mask in the .pointer callback is invalid. It should
be in period_bytes range. The period_bytes is pow(2), so simple bitwise
operation is used.
Idea was taken from ALSA bug#4455.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed.
This patch provides stub codec that can be used in these configurations.
On DM646x EVM the McASP1 is connected to the S/PDIF out.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added some missing register bits definitions to reduce magic numbers.
Also renamed some to follow the names on the datasheet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the hardware cannot store the multiple replies
like on RIRB, thus each verb has to sync and wait for the response no
matter whether the return value is needed or not. Otherwise it may
result in a wrong return value from the previous verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unsigned variables should use `%u' rather than `%d'.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A docking mic control is shown by default. The Compaq Presario
CQ60 laptop has no docking connector, so designate it as a
CXT5051_HP model.
This makes the phantom mixer slider disappear.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function signature for spin_event_timeout() has changed in version V9.
Adjust the mpc5200 AC97 driver to use the new function.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The effect of symmetric_constraints should provide a standard way to
enforce the use of the same sample rate for both directions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
Kernel 2.6.18 broke the MotU Fastlane, which uses duplicate endpoint
numbers in a manner that is not only illegal but also confuses the
kernel's endpoint descriptor caching mechanism. To work around this, we
have to add a separate usb_set_interface() call to guide the USB core to
the correct descriptors.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: David Fries <david@fries.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM hw_ptr jiffies check results sometimes in problems when a
hardware doesn't give smooth hw_ptr updates. So far, au88x0 and some
other drivers appear not working due to this strict check.
However, this check is a nice debug tool, and the capability should be
still kept.
Hence, we disable this check now as default unless the user enables it
by setting the xrun_debug mode to the specific stream via a proc file.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hw_ptr_jiffies has to be reset properly to avoid the invalid
check of jiffies delta in snd_pcm_update_hw_ptr*() functions.
Especailly this patch fixes the bogus jiffies check after the puase
and resume.
This patch is a modified version of the original patch by Jaroslav.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These drivers use spin_event_timeout() which is only present in the
PowerPC tree at present and which is undergoing some API revisions
so temporarily mark them as BROKEN until these issues are sorted
out.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I've implemented retries for when the AC97 hardware doesn't reset on
first try. About 10% of the time both the Efika and pcm030 AC97 codecs
don't reset on first try and need to be poked multiple times. Failure
is indicated by not having the link clock start ticking. Every once in
a while even five pokes won't get the link started and I have to power
cycle.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rewrite the mpc5200 audio DMA code to support both I2S and AC97.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add missing check of pin vref 50 and others in Realtek codecs
ALSA: hda - Add 5stack-no-fp model for STAC927x
ALSA: hda - Add forced codec-slots for ASUS W5Fm
This patch adds a debug mode to make the codec communication
synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c,
and the call of snd_hda_codec_write*() will become synchronous,
i.e. wait for the reply from the codec at each time issuing a verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the current driver code doesn't do any update
for RIRB just for any safety reason. But, actually the RIRB and
single_cmd mode don't conflict. Unsolicited events can be delivered
even while using the single_cmd mode.
This patch allows the handling of unsolicited events with single_cmd
mode, just always checking RIRB independent from single_cmd flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code doesn't handle the multiple gameports properly,
and uses unnecessary global static variables to store the data.
This patch changes the probe / remove routines to use the driver
data assigned to the dedicated pci device, and adds the support of
multiple devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A code clean up, coding style fixes.
The firmware loading routine is split to an own function to improve
the readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec communication error occurs, the CORB/RIRB counters should
be reset first before re-issuing the verb. Simply call azx_free_cmd_io()
and azx_init_cmd_io() to achieve that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DAPM switch for HeadsetL/R mute. Since all bits are are needed
for the HFL/R pop removal to work the switch is using the SW_SHADOW
no HW register for the HandsfreeL/R mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shadow, non HW register for dealing with the HandsfreeL/R
muting.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E
to a more appropriate DAPM_PGA_E widget.
Also fix the power-up sequence to match with the TRM.
The power-down sequence is not described in the TRM, so do it
in a way, which seams like the correct sequence.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Asus Xonar Essence ST and its daughterboard.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Xonar HDAV1.3 has a digital input jack, so enable the corresponding
device.
This is not related to the HDMI stuff, which stays unsupported.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another PCI ID for a second revision of the Xonar DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a PCM stream is closed, flush the corresponding DMA channel.
Otherwise, the DMA controller would continue to output the last sample
which would result in a DC offset on the output.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs like ALC861 seem to support only VREF50 while the
current driver assumes it's only VREF80. Check other VREF bits to set
the correct value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename the functions in the mpc5200 DMA file from i2s based names to dma
ones to reflect the file they are in.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Basic split of mpc5200 DMA code out from i2s into a standalone file.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent fix for the headphone volume control on IDT/STAC codecs
resulted in the removal of invalid "Side" volume eventually. But,
if the front panel doesn't exist, this setup could be regarded as a
sort of regression, as reported in kernel bug #13250.
Now as a workaround, a new model 5stack-no-fp is added so that the user
without the front panel can choose this one explicitly.
Reference: bko#13250
http://bugzilla.kernel.org/show_bug.cgi?id=13250
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS
problem like W5F.
Tested-by: Alp Kılıç <kilic.alp@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fairly minor issues:
- Don't register the DAIs, it's not required for AC97 devices.
- Make unexported functions static.
- Wrap some excessively long lines.
- Undo tab/space breakage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that we always set a new sysclk when using the FLL in master mode
and pick out the correct value for the sample rate in hw_params().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase
Inversion Playback Switch' truncated to 'Sigmatel Surround Phase
Inversion Playback ' bootup message by omitting weird Sigmatel prefix
in this case; also fix up the related ca0106 mixer control removal
part by using identical naming there.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Give unique stream names for the two playback streams so
DAPM can figure out which codec_dai is in use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this patch the initial headset pop-removal related values are
configured for the twl4030 codec (ramp delay and sysclk).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
twl4030_setup_data structure can be passed from platform drivers to
the codec via the snd_soc_device->codec_data pointer.
Currently the setup data has support for the Headset pop-removal
related configuration, which differs from board to board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The second argument of the probe method points to the amba_id
structure, so it's better passed with the correct type. None of the
current in-tree drivers uses the pointer, so they have only been
checked for a clean compile.
Change suggested by Russell King.
Signed-off-by: Alessandro Rubini <rubini@unipv.it>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
The commit 5a641bcd63 changed the
printk format to '%lu', but the value passed seems to be dependent
on the architecture. On x86-64, I got a new warning now because an
int value is passed actaully.
As a workaround, just cast the value always to unsigned long.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ctxfi driver requires explicitly the 4k page size, and gives a
build error on architectures with non-4k pages.
As a workaround, just add the kconfig dependency on X86, which is
the only architecture ever tested.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles
the headset ramp up and down sequences needed for the pop noise
removal.
With this patch the order of the internal components in the twl4030
codec is turned on and off in a correct order.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Restructuring the twl4030 codec's DAPM routing to be able to handle the power
sequences correctly.
The twl4030 codec internal implementation have this order:
DAC -> Analog PGA -> Mixer/Mux
While the ASoC framework expects the following order:
DAC -> Mixer -> Analog PGA
This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and
adds two levels of mixer to handle the digital and analog loopback
functionality.
Now the analog loopback does not powers on any of the DACs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the integrated ACLC of the TXx9 family.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital beep tone is calculated in two different ways depending
on the codec chip. The standard one is using a divider, and another
one is a linear tone for IDT/STAC codecs. Currently, only the
latter type is used for all codecs, which resulted in a wrong tone
pitch.
This patch adds the calculation of the standard HD-audio type.
Also clean-up the fields in hda_beep struct.
Reference: bko#13162
http://bugzilla.kernel.org/show_bug.cgi?id=13162
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper prefix to each kernel message in hda_intel.c.
Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used
together with snd_print*().
Reference: bko#13207
http://bugzilla.kernel.org/show_bug.cgi?id=13207
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver.
Voice DAI init function enables corresponding interface by
writting directly to VOICE_IF codec register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a control for selecting the codec operation mode. TWL4030 codec
has two modes:
- Option 1. Audio only (4 audio DACs)
- Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC)
Control is restricted when a stream is ongoing, since codec's
operation mode cannot be changed on-the-fly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move all the gpio functions out of <mach/hardware.h> as
this file is for defining the generic IO base addresses
for the kernel IO calls.
Make a new header <mach/gpio-fns.h> to take this and
include it via the chain from <linux/gpio.h> which is
what most of these files should be using (and will be
changed as soon as possible).
Note, this does make minor changes to some drivers but
should not mess up any pending merges.
CC: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.
Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI entries of Creative with HD-audio class can be the devices
with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi
driver. With that driver, the device will mutate from HD-audio to
its native class.
This patch adds a simple ifdef to avoid the conflict of device probe
between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still
OK to be added as it has no emu20kx chip, and is a pure HD-audio
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink
and downlink paths, respectively.
This patch also corrects voice 8/16kHz mode selection bit
(SEL_16K) of CODEC_MODE register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai