We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.
Thanks to Guillem Jover for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion the names for the DACs changed:
DACL1 -> DAC Left1
...
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mux switch related texts fits to on line, no need to wrap
them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BUG() should be marked as not returning but for at least some
configurations (including some widely deployed compilers) that's either
not happening or being forgotten by the compiler. Add some extra return
statements to the affected paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The return value of pci_enable_device() must be checked even in resume
callback:
sound/pci/ca0106/ca0106_main.c:1779: warning: ignoring return value of ‘pci_enable_device’, declared with attribute warn_unused_result
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add snd_ prefix to avoid the conflict of symbols in omac-mcbsp.c:
sound/soc/omap/omap-mcbsp.c:503: error: static declaration of 'omap_mcbsp_init' follows non-static declaration
arch/arm/plat-omap/include/mach/mcbsp.h:373: error: previous declaration of 'omap_mcbsp_init' was here
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the function name of module init entry for twl4030.c, which
conflicted with the existing hardware init function:
sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init'
sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here
Also fixed the section type of init function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes use of the support for delayed DAI registration to allow the
WM8900 I2C device to be registered by general platform/architecture code
rather than as part of the ASoC device probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will allow codec drivers to be refactored to allow them to be
registered out of line with the ASoC device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the lists of platforms, platform DAIs and cards to check to see that
everything has registered. Since relationships are still specified by
direct references to the structures in the drivers and the drivers all
register everything at modprobe there should be no practical effect yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently this is done at module probe time since ASoC ties in codec
device probe to the instantiation of the entire ASoC device. Subsequent
patches will refactor the codec drivers to handle probing separately.
Note that the core does not yet use this information.
AC97 is special since the codec is controlled over the AC97 link but
we want to give the machine driver a chance to set up the system before
trying to instantiate since it may need to do configuration before the
AC97 link will operate
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a separate gain control for the Headset output already.
Do not reset the gain to 0 dB at power up.
In power-down, there is no need to set the Headset output gain
to power-down mode, since if the CODECPDZ is in powered off this
setting has no effect.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Handsfree outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Carkit outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Headset outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the PreDrive outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Earpiece output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four APGA switch to DAPM routing and widgets.
Add user control for DA enable for all APGA as normal
control.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four DACs to dapm_widgets with power switch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the license and misc comments at the beginning of the code.
Also, use ns_to_ktime() for simplification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the restoration of the standard PCI configuration registers
in the snd_hda_intel driver to a ->resume_early() callback executed
with interrupts disabled, since doing that with interrupts enabled
may lead to problems in some cases.
This patch addresses the regression from 2.6.26 tracked as
http://bugzilla.kernel.org/show_bug.cgi?id=12121 .
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds basic support for OMAP3 Pandora.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code overrides the event type on input pins always to
PWR_EVENT. Although this still works (PWR_EVENT and INSERT_EVENT
are handled samely), it'd be better to avoid such overrides.
Also, currently the unsol events are registered even for fixed pins
which will never raise the pin-detection event.
This patch fixes both issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin-detection function used in patch_sigmatel.c shouldn't be specific
to HP pin because it's used for input pins in general, too.
This patch fixes the detection function, removes the HP check from it
and moves to stac92xx_hp_detect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce the command timeout to 0.5sec. Should be enough to allow a
working command interface but removes a RCU stall and slow resume on
some revisions where the AC97 revision detection stalls in resume.
Signed-off-by: Peter Gruber <nokos@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add aic3x_set_headset_detection() function to define the headset
detection mode for tlv32aic3x chips
- added aic3x_button_pressed()
- Read from the real-time registers in aic3x_headset_detected() to query
headset presence without an occured interrupt
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TWL4030 codec device has two ADCs. Both of them can have
several inputs routed to them, but TRM says that only one source
can be selected for every ADC, even though every source has a
dedicated bit in the registers.
This patch adds input source controls. It modifies default register
values to have no inputs selected and ADCs disabled. When some
input is selected, control handlers enable apropriate input
amplifier and ADC. If a microphone is selected, bias power is
automatically enabled. When some input is deselected, unused
chip parts are disabled.
Microphone and line input recording tested on OMAP3 pandora board.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a sparse warning caused by the lack of a connection with the
prototype for ac97_bus_type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the conversion (drop) from 24bit in the DSP to 16bit in AC97,
the maximum capture level on Audigy seems lower than it could be.
This patch adds a workaround to enable the artificial capture boost
switch. When this switch is on, the whole analog capature level is
boost up. However, this results in the lower capture resolution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
All outputs have dedicated gain controls except the
HandsFree output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Playback volume controls for all four DACs.
All four paths has three levels of volume controls:
Digital Fine gain, Digital Coarse gain, Analog gain.
The controls are named to reflect their connection to the DACs.
Per DAC volume can be performed, if needed:
amixer sset 'DAC1 Analog' 5,10
DACL1 analog gain to 5
DACR1 analog gain to 10
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital Capture gain control has a range:
0 to 31 dB in 1 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the old-style trigger callback in s3c2443-ac97.c:
sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrong shutdown callback type. Also removed the unused variables
there:
sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
sound/soc/pxa/corgi.c: At top level:
sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9171e5e6a2.
I can't reproduce the compile warnings any more. The warnings
might be some weird cross-compiling set up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig,
thus no more need in Kconfig of each sub directory.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit e669dae614, since it
is incomplete, and clashes with fuller patches and the sparc 32/64
unification effort.
Requested-by: David Miller <davem@davemloft.net>
Acked-by: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The WM9715 is software compatible with the WM9711 and WM9712.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hide annoying uninitialized warnings:
sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function
sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- compatibility issue : change firmware filenames
the pcxhr driver version <= 1.0.18a does not work
with new firmware > 1.0.17. Keep the old firmware files
and add new firmware files with different names
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power-state changes in patch_sigmatel.c are accessed via *_cached()
but they shouldn't be really cached. Fixed to the normal write.
Also, stac92hd71xx_suspend and resume are no longer necessary as the
power-state changes are handled properly in the common routine.
Removed these hacks now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp switch of SPDIF outputs have to be cached in the amp cache
instead of codec cache. Otherwise it conflicts with the IEC958
playback switch control in hda_codec.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'merge' of git://git.kernel.org/pub/scm/linux/kernel/git/paulus/powerpc:
powerpc: Fix system calls on Cell entered with XER.SO=1
powerpc/cell: Fix GDB watchpoints, again
powerpc/mpic: Don't reset affinity for secondary MPIC on boot
powerpc/cell/axon-msi: Retry on missing interrupt
powerpc: Fix boot freeze on machine with empty memory node
powerpc: Fix IRQ assignment for some PCIe devices
powerpc/spufs: Fix spinning in spufs_ps_fault on signal
powerpc/mpc832x_rdb: fix swapped ethernet ids
powerpc: Use generic PHY driver for Marvell 88E1111 PHY on GE Fanuc SBC610
powerpc/85xx: L2 cache size wrong in 8572DS dts
powerpc/virtex: Update defconfigs
powerpc/52xx: update defconfigs
xsysace: Fix driver to use resource_size_t instead of unsigned long
powerpc/virtex: fix various format/casting printk mismatches
powerpc/mpc5200: fix bestcomm Kconfig dependencies
powerpc/44x: Fix 460EX/460GT machine check handling
powerpc/40x: Limit allocable DRAM during early mapping
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Check model for Dell 92HD73xx laptops
ALSA: hda - mark Dell studio 1535 quirk
ALSA: hda - No 'Headphone as Line-out' swich without line-outs
ALSA: hda - Fix AFG power management on IDT 92HD* codecs
ALSA: hda - Fix caching of SPDIF status bits
ALSA: hda - Add a quirk for Dell Studio 15
ALSA: hda: Add STAC_DELL_M4_3 quirk
sound/sound_core: Fix sparse warnings
ALSA: hda: STAC_DELL_M6 EAPD
switch to __init for those; unlike powerpc sparc has no hotplug support
for that stuff and their ->probe() tends to call __init functions while
being declared __devinit.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Gateway notebooks have their ID inside codec vendor ID, not at PCI ID. Due to
that, model auto-detection were not possible with the standard seek method.
This is what is found at lspci -vnn:
00:14.2 Audio device [0403]: ATI Technologies Inc SB450 HDA Audio [1002:437b] (rev 01)
Subsystem: ATI Technologies Inc SB450 HDA Audio [1002:437b]
Yet, autodetection is possible, since the codec properly reflects the vendor at
the Subsystem ID:
$ cat /proc/asound/card0/codec#0 |head -4
Codec: SigmaTel STAC9250
Address: 0
Vendor Id: 0x83847634
Subsystem Id: 0x107b0367
This patch adds a new autodetection function that seeks for codec subsystem ID.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The global functions in hda_codec.c and other core parts are only
for HD-audio codec and controller drivers. When the HD-audio driver
is built in kernel, all stuff have to be statically linked, thus
we don't need any exports.
This patch introduces a conditional macro to do export only
when needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a hook for proc outputs of codec-specific stuff.
Moved realtek-specific coeff output into patch_realtek.c as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved power_save field initialization inside a proper ifdef
to fix a build error without CONFIG_SND_HDA_POWER_SAVE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split the monolithc HD-audio driver into several pieces:
- snd-hda-intel HD-audio PCI controller driver; loaded via udev
- snd-hda-codec HD-audio codec bus driver
- snd-hda-codec-* Specific HD-audio codec drivers
When built as modules, snd-hda-codec (that is invoked by snd-hda-intel)
looks up the codec vendor ID and loads the corresponding codec module
automatically via request_module().
When built in a kernel, each codec drivers are statically hooked up
before probing the PCI.
This patch adds appropriate EXPORT_SYMBOL_GPL()'s and the module
information for each driver, and driver-linking codes between
codec-bus and codec drivers.
TODO:
- Avoid EXPORT_SYMBOL*() when built-in kernel
- Restore __devinit appropriately depending on the condition
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_mixart_create creates a link between mgr and card that
allows snd_mixart_free to free card as well. But if snd_mixart_create
fails, then the link has not been created and card has to be freed explicitly.
The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S,S1;
position p1,p2,p3;
expression E,E1;
type T,T1;
expression *ptr != NULL;
@@
(
if ((x@p1 = snd_card_new(...)) == NULL) S
|
x@p1 = snd_card_new(...);
)
... when != snd_card_free(...,(T)x,...)
when != if (...) { <+... snd_card_free(...,(T)x,...) ...+> }
when != true x == NULL || ...
when != x = E
when != E = (T)x
when any
(
if (x == NULL || ...) S1
|
if@p2 (...) {
... when != snd_card_free(...,(T1)x,...)
when != if (...) { <+... snd_card_free(...,(T1)x,...) ...+> }
when != x = E1
when != E1 = (T1)x
(
return \(0\|<+...x...+>\|ptr\);
|
return@p3 ...;
)
}
)
@ script:python @
p1 << r.p1;
p3 << r.p3;
@@
print "* file: %s snd_card_new: %s return: %s" % (p1[0].file,p1[0].line,p3[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reconfiguration of PCM affected all PCM streams on the bus, but
this this should be done rather only for the target codec.
This patch does the following:
- introduce bitmap indicating the PCM device usages on a hda_bus
- refactor the PCM build functions
- fix __devinit prefix in some fucntions
- add a proper ifdef around HDA-reconfig-specific functions
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move power_save option into hda_intel.c, and make a field in hda_bus,
instead of keeping module parameters in separate files.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor clean ups: move snd_hda_codecs_inuse() into hda_intel.c and
make static. Also, make snd_hda_query_supported_pcm() static
as it's used only in hda_codec.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-usb-caiaq: clean up the control adding code by moving dulpicate code
to a function.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables more routing functions for tlv320aic3x codecs.
It is now possible to
- control the volume of the PGA bypass path for the HPL, HPR, HPLCOM
and HPRCOM outputs individually
- route right line1 input to the left ADC channel
- route left line1 input to the right ADC channel
- route right mic3 input to left DAC channel
- route left mic3 input to right DAC channel
- route left line1 input to right line1 output
- route right line1 input to left line1 output
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the callback mode HRTIMER_CB_IRQSAFE_UNLOCK, the start of the
stream with zero delay doesn't work. Since IRQSAFE mode is removed,
we have to change the pcsp start-up code.
This patch splits the callback function to two parts, the triggering
of the port and the calculation of the expire time, and the update of
the ALSA PCM core. The first part is called both from the trigger-start
and the hrtimer callback while the latter is handled only in the
hrtimer callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no argument named @state in snd_hda_resume,
remove its' comment.
Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no argument named @clk_id in snd_soc_dai_set_fmt,
remove its' comment.
Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add ASoC support for TI SDP3430. It's based on Gumstix
Overo SoC code by Steve Sakoman.
Signed-off-by: Misael Lopez Cruz <mesak82@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes Kconfig dependency of TWL4030 audio codec driver
with TWL4030 core driver on both overo and omap2evm
boards
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Acked-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch adds support for mono audio links so that McBSP DAI can operate with
real mono codecs. In I2S, the signalling remains the same but only first
frame (left channel) is transmitting audio data and second frame having null
data. In DSP_A, only first frame is transmitted.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prepare for upcoming McBSP DAI update adding support for mono links by
restricting number of channels to 2 in N810. This is due tlv320aic3x which
claims channels_min = 1 and playing pure mono audio over I2S would cause
it to be played only from left channel if both cpu and codec DAI's claim to
support mono.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Impact: cleanup, move all hrtimer processing into hardirq context
This is an attempt at removing some of the hrtimer complexity by
reducing the number of callback modes to 1.
This means that all hrtimer callback functions will be ran from HARD-irq
context.
I went through all the 30 odd hrtimer callback functions in the kernel
and saw only one that I'm not quite sure of, which is the one in
net/can/bcm.c - hence I'm CC-ing the folks responsible for that code.
Furthermore, the hrtimer core now calls callbacks directly with IRQs
disabled in case you try to enqueue an expired timer. If this timer is a
periodic timer (which should use hrtimer_forward() to advance its time)
then it might be possible to end up in an inf. recursive loop due to the
fact that hrtimer_forward() doesn't round up to the next timer
granularity, and therefore keeps on calling the callback - obviously
this needs a fix.
Aside from that, this seems to compile and actually boot on my dual core
test box - although I'm sure there are some bugs in, me not hitting any
makes me certain :-)
Signed-off-by: Peter Zijlstra <a.p.zijlstra@chello.nl>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Check the model type instead of PCI SSID for detection of the mic types
on Dell laptops with IDT 92HD73xx codecs. In this way, a new laptop
can be tested via model module option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add support for pcxhr stereo cards mixer controls
- adjust tlv db scales to real dBu values
- fix bug with monitoring volume control pcxhr_monitor_vol_put
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards and their firmware
- autorize sound cards without analog IO
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- minor bugfixes : period and buffer size consraints
- fix PLL register values
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we need to handle many unsolicited events assigned to different
widgets, allocate the event dynamically using the existing events
array, and use the tag appropriately instead of combination of fixed
number and widget nid. (Note that widget nid can be over 4 bits!)
Also, replaced the call of unsol_event handler with a dedicated
function to be more readable.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AFG pin power-mapping isn't properly set for the fixed I/O pins
on IDT 92HD* codecs. This resulted in the low power mode after the
boot until any jack detection is executed, thus no output from the
speaker.
This patch fixes the power mapping for the fixed pins, and also fixes
the GPIO bits and digital I/O pin settings properly in stac92xx_ini().
Reference: Novell bnc#446025
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPDIF status bits controls are written via snd_hda_codec_write()
without caching. This causes a regression at resume that the bits
are lost.
Simply replacing it with the cached version fixes the problem.
Reference:
http://lkml.org/lkml/2008/11/24/324
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Medion MD96630 has ALC268 codec on slot#2 although it's not used
for any purpose. This codec conflicts with the primiary codec ALC888
on slot#0, and gives mixer errors.
This patch adds a corresponding entry to probe_mask blacklist.
Reference: Novell bnc#412528
https://bugzilla.novell.com/show_bug.cgi?id=412528
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
0x0 : Power down (mute)
0x1 : 6dB
0x2 : 0 dB
0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)
After the patch, FGAIN volume control works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the bug 0004240: ALC888 - Intel HDA - Headphone Controlling.
It is made against the 2008-11-23 snapshot.
Added Realtek ALC888 model entry for the Fujitsu-Siemens Amilo Xa3530
laptop. It has 4 jacks: HP out, Mic-in, Line-in and Line-out/Side/SPDIF
(this one is on the laptop side, the other ones are on the rear).
Model detection works.
Headphone jack sense works now.
Front mic works now, was same as Acer Aspire 4930G.
Added channel mode from 2 to 8 channels.
In 2ch and 4ch modes, the front is also sent to the Line-out/side jack
for convenience instead of just muting the Line-out/side jack like other
models do.
When using the Mic-in jack as CLFE, the sound is very low (bug?). To
work it around, in 6ch mode the CLFE channel is duplicated to the
Line-out/side jack because this one has a better amp.
Cc: manu@frogged.de
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix "defined but not used" build warning by moving eld_versoin_names[]
and cea_edid_version_names[] into hdmi_print_eld_info().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DisplayPort is a digital display interface standard put forth by
the Video Electronics Standards Association (VESA). It defines a
new license-free, royalty-free, digital audio/video interconnect,
intended to be used primarily between a computer and its display monitor,
or a computer and a home-theater system.
- From Wikipedia, the free encyclopedia
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- rename ELD proc write routine to hdmi_write_eld_info()
- support modifying WMAPro's profile
Write to some ELD fields (monitor_name, manufacture_id, product_id,
eld_version, edid_version) are deliberately not supported, since that
won't correct wrong behaviors and only leads to confusions.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- make some messages more user friendly
- add message prefix "HDMI:" to indicate the problem's domain
(also easier to do `dmesg | grep HDMI` ;-)
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print some CA selecting info, which could be valuable for debugging when
something goes wrong.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_print_pcm_rates() and snd_print_pcm_bits() are used by both
hda_proc.c and hda_eld.c, thus they have to be defined in the common
place.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warning:
sound/pci/hda/patch_nvhdmi.c:161:25: warning: symbol
'snd_hda_preset_nvhdmi' was not declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.
NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
The logic is flawed (e.g. the slave might startup before the
master's rate and sample_bits are set).
Signed-off-by: Karl Beldan <karl.beldan@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the codec re-configuration feature selectable via Kconfig,
CONFIG_SND_HDA_RECONFIG.
Also mark it as experimental (as it really is).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack instances created in patch_sigmatel.c may be double-freed.
The device management code checks the invalid element, and thus there
is no real breakage, but it spews annoying warning messages.
But, we can't simply remove the release calls of these jack instances
because they have to be freed when the codec is re-configured.
Now, a new flag, bus->shutdown is introduced to indicate that the bus
is really being unloaded, i.e. the objects managed by the device
manager will be automatically deleted. We release these objects only
when this flag isn't set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow users to fix quicks of ELD ROMs by writing new values to the ELD proc
interface. The format is one or more lines of "name hex_value".
Users can add/remove/modify up to 32 SAD(Short Audio Descriptor) entries.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "monitor name" to "monitor_name" to conform with the keyword style.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added STAC_DELL_M4_3 quirk for Dell systems, also reorganized the
board config switch to assign number of digital muxes, microphones,
and SPDIF muxes via the PCI quirk defined.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warnings:
sound/sound_core.c:460:2: warning: returning void-valued expression
sound/sound_core.c:477:2: warning: returning void-valued expression
sound/sound_core.c:510:5: warning: symbol 'soundcore_open' was not
declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The changes specific for Samsung laptops seem unapplicable to other
hardware models like ASUS. The mic inputs are lost on such hardware
by the change 5d5d5f43f1.
This patch adds back the old laptop-eapd model, and create a new
model "samsung" for the new one specific to Samsung laptops with
automatic mic selection feature.
Reference: kernel bugzilla #12070http://bugzilla.kernel.org/show_bug.cgi?id=12070
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for EAPD on system suspend and disabling EAPD on headphone jack
detection for STAC_DELL_M6 laptops.
This patch fixes the regressions, the silent output on HP of some Dell
laptops (see Novell bnc#446025):
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added Realtek ALC888 model entry for the Acer Aspire 4930G laptop that
fixes the following features:
- internal microphone
- heaphone jack sense
- channel mode
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace 5 free-and-return-err blocks with goto-out-free ones.
This makes the main logic more outstanding.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds twl4030 audio support on omap2evm
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock
[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).
[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Strip out some ELD printk messages that end user won't care,
and make the output compact.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a global function snd_print_pcm_bits() and use it in the ELD code.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved support for EAPD mute on suspend from stac92hd71xx_suspend
to the generic stac92xx_suspend function.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactor the channel mapping code for consistent naming
and make it more informed about channel allocations.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To play a 3+ channels LPCM/DSD stream via HDMI,
- HDMI sink must tell HDMI source about its speaker placements
(via ELD, speaker-allocation field)
- HDMI source must tell the HDMI sink about channel allocation
(via audio infoframe, channel-allocation field)
(related docs: HDMI 1.3a spec section 7.4, CEA-861-D section 7.5.3 and 6.6)
This patch attempts to set the CA(channel-allocation) byte in the audio infoframe
according to
- the number of channels in the current stream
- the speakers attached to the HDMI sink
A channel_allocations[] line must meet the following two criteria to be
considered as a valid candidate for CA:
1) its number of allocated channels = substream->runtime->channels
2) its speakers are a subset of the available ones on the sink side
If there are multiple candidates, the first one is selected. This simple
policy shall cheat the sink into playing music, but may direct data to the
wrong speakers.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
code refactor: make a global function snd_print_channel_allocation().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
code refactor: make a standalone function hdmi_fill_audio_infoframe().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not ignore the return of 'device_create_file' in
'snd_card_register' and thereby fixing the following warnings:
sound/core/init.c: In function 'snd_card_register':
sound/core/init.c:640: warning: ignoring return value of
'device_create_file', declared with attribute warn_unused_result
sound/core/init.c:641: warning: ignoring return value of
'device_create_file', declared with attribute warn_unused_result
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A probe function should have a clean return 0 path.
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clean up redudent code and correct building problem in non-mmap mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the multiple imux instances for matrix-type mixers like ALC882.
So far, only ALC260 used this feature, but other codecs may need a
similar stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the GPIO mask and co initialization in patch_stac92hd71bxx()
so that the gpio_maks for HP_M4 model is set properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the restore of pin configs at resume for some STAC/IDT codec
models. These models set explicitly the pin configs after the default
init configs, and these aren't restored properly at resume.
This patch introduces two changes:
- Allocate always pin_configs array in stac_spec so that the driver
can overwrite the value freely
- Introduce stac_change_pin_config() to change the pin config value
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call the init callback and remove duplicated codes in stac92xx_resume().
This also fixes the missing initialization such as digital I/O pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack detection input elements should be created in build_controls
callback instead of init callback because init can be called multiple
times by suspend/resume and power-saving.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create /proc/asound/card<card_no>/eld#<codec_no> to reflect the audio
configurations and capabilities of the attached HDMI sink.
Some notes:
- Shall we show an empty file if the ELD content is not valid?
Well it's not that simple. There could be partially populated ELD,
and there may be malformed ELD provided by buggy drivers/monitors.
So expose ELD as it is.
- The ELD retrieval routines rely on the Intel HDA interface,
others are/could be universal and independent ones.
- How do we name the proc file?
If there are going to be two HDMI pins per codec, then the current naming
scheme (eld#<codec no>) will fail. Luckily the user space dependencies should
be minimal, so it would be trivial to do the rename if that happens.
- The ELD proc file content is designed to be easy for scripts and human reading.
Its lines all have the pattern:
<item_name>\t[\t]*<item_value>
where <item_name> is a keyword in c language, while <item_value> could be any
contents, including white spaces. <item_value> could also be a null value.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ELD handling routines can be shared by all HDMI codecs,
and they are large enough to make a standalone source file.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to share some code with print_pcm_rates(),
so extract a common routine snd_print_pcm_rates() from it.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove get_amp_nid(): it duplicates the one defined in hda_local.h
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only fully documented registers are cached in the WM8990 but additional
registers exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reorder HDMI audio enabling sequence so that
1) the sink knows about the coming audio stream
2) unmute
3) start transferring audio samples
The theory is that in the path A=>B=>C, we first make C ready, and then
enable B, and lastly allow A to send audio samples.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for EAPD on system suspend and disabling EAPD on headphone jack
detection for STAC_DELL_M6 laptops.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new alc883 model ALC1200_ASUS_P5Q for ASUS P5Q-EM boards.
It is the same as ALC883_6ST_DIG except that the SPDIF digital
output nid is 0x10.
Tested-by: Andrei Tanas <andrei@tanas.ca>
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check board preset model instead of codec->subsystem_id in
patch_92hd71bxx() so that other hardwares configured via the model
option work like the given model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Error handling code following a kzalloc should free the allocated data.
The error handling code is adjusted to call pci_disable_device(pci); as
well, as done later in the function
The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,l;
position p1,p2;
expression *ptr != NULL;
@@
(
if ((x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...)) == NULL) S
|
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
)
<... when != x
when != if (...) { <+...x...+> }
x->f = E
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed value for STAC_VREF_EVENT from 0x40 to 0x00 because the
unsol response value is only 6-bits width and the former value
was 1<<6 which is an overrun.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8728 is a high performance stereo DAC designed for applications
such as DVD, home theatre and digital TV.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 8dc840f88d. Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue. Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Split the bound Master control to individual Headphone and Speaker
volume controls for VAIO with STAC982x codecs.
The Master controls is still created as a vmaster.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch it is possible to select drivers which require
bestcomm support without bestcomm support being selected. This
patch reworks the bestcomm dependencies to ensure the correct
bestcomm tasks are always enabled.
Reported-by: Hans Lehmann <hans.lehmann@ritter-elektronik.de>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
The digital beep widget may have no mute control, and always enabling
the beep is ofen pretty annoying, especially on laptops.
This patch adds a mixer control "PC Beep Playback Switch" when there
is no mixer amp mute is found, and controls it on software.
Reference: Novell bnc#444572
https://bugzilla.novell.com/show_bug.cgi?id=444572
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).
Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch uses HD Audio PCI class code to detect AMD HD Audio cards.
Signed-off-by: Libin Yang <libin.yang@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the handling of SiI1392 HDMI codec from patch_atihdmi.c to
patch_intelhdmi.c, which makes our ASUS P5E-VM HDMI board work.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card->id (card text identification) can be changed at runtime.
It might be confusing to have old text identification in device name.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For udev, we need a way to rename soundcard names. The soundcard numbers
(indexes) are hardwired but we have a text identification which can be
changed at run-time. The ALSA user space tools already allow using of
this text identification.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the recent change for multiple HP as line-out switch, only
one of the multiple headphons (usually a wrong one) is toggled
and the other pins are still disabled. This causes the silent output
problem on some Dell laptops.
Also, the hp_switch check is screwed up when a line-in or a mic-in
jack exists. This is added as an additional output, but hp_switch
check doesn't take it into account.
This patch fixes these issues: simplify hp_switch check by using
the NID instead of bool, and clean up / fix the toggle of HP pins
in unsol event handler code.
Reference: Novell bnc#443267
https://bugzilla.novell.com/show_bug.cgi?id=443267
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input pins are sometimes not initialized properly because
of the optimization check of the current pinctl code.
Force to initialize the mic input pins so that they can be set up
properly even if they were in a weird state. But keep other input
pins if already set up as input, since this could be an extra mic
pin.
Reference: Novell bnc#443738
https://bugzilla.novell.com/show_bug.cgi?id=443738
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The creation of analog-mux mixer element is missing in
patch_stac9200() due to the dynamic allocation patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Make the HP EliteBook 8530p use AD1884A model laptop
ALSA: gusextreme: Fix build errors
ALSA: hdsp: check for iobox and upload firmware during ioctl
ALSA: HDSP: check for io box before uploading firmware
ALSA: hda - Add another HP model (6730s) for AD1884A
alsa: fix snd_BUG_on() and friends
ALSA: hda - Add a quirk for MEDION MD96630
ALSA: hda - Limit the number of GPIOs show in proc
Added a QUIRK to patch_analog.c for the HP Elitebook 8530p
(IDs 0x103c:0x30e7) to use AD1884A model 'laptop' by default.
Playback and Capture confirmed working.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
currently, the error message when trying to run hdspmixer or hdspconf
if the breakout box is not connected is somehow misleading, since it
asks the user to upload the firmware.
this patch adds a test, whether the breakout box is connected and
tries to upload the firmware in the case, that it is not present, e.g.
because of power failures of the breakout box.
[Minor coding-style fixes by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
currently the hdsp driver tries to upload the firmware, even if the
io box is not connected. this patch adds a check for the io box
before trying to upload the firmware.
thus instead of messages complaining about the fifo status and firmware
loading failure, the driver gives a message that no multiface or
digiface is connected.
[A minor coding-style fix by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added model=laptop for another HP machine (103c:3614) with AD1884A
codec.
Signed-off-by: Michel Marti <mma@objectxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on Dell Studio Desktop tells wrong codec probe masks.
This patch gives the preset mask value to avoid invalid access.
Reference: Novell bug#440907
https://bugzilla.novell.com/show_bug.cgi?id=440907
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The master switch doesn't influence on NID 0x15, the headphone jack
on HP 3013 model because alc260_hp_master_update() ignores the passed
arguments.
Also, corrected the wrong arguments of hp3013 (0x10 and 0x15) although
this doesn't change any behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for more sample rates, different crystals
and split playback/capture rates.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The widget layout of the Fujitsu Lifebook S6420 (which is ICH9M-based
and uses an ALC269) is similar but not identical to the Lifebook
S6410/E8410 (which are ICH8M-based and use an ALC262).
It is named lifebook as fujitsu is in use for Amilo machines. This builds
on the Quanta FL1 work and supports all analog inputs & outputs that I am
aware of. Microphone autoswitch is implemented. The laptop mic port takes
precedence over the dock mic port if both happen to have a jack plugged in.
This made sense to me as a design decision (imagine a presentation
environment with the dock fully wired in and the presenter quickly wanting
to override the mic with a headset).
There is mention of a digital audio path on the codec graph, so perhaps
the headphone socket is dual-function analog/digital. I will follow up
with another patch if I can acquire equipment to test this.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an error occurs during the codec probing, typically accessing to an
non-existing codec slot, the controller chip gets often screwed up and
can no longer communicate with the codecs.
This patch adds a preparation phase just to probe codec addresses before
actually creating codec instances. If any error occurs during this
probing phase, the driver resets the controller to recover.
This will (hopefully) fix the famous "single_cmd" errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hda_bus ops callback take struct hda_bus pointer.
Also, the command callback takes the composed command word, instead of
each small bits in arguments.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to TRM, 256*Fs clock output should be enabled
when TWL4030 is in slave mode, not master.
This allows sound to work on OMAP3 Pandora, which uses
256*Fs clock.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C24xx dma does not allow more than one buffer to be enqueue prior to
the dma transfers starting. This patch adds an additional parameter to
s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load
value.
Signed-off-by: David Anders <danders at amltd.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.
[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are actually two variants of ALC268 Acer implementation, one
with an analog built-in mic (pin 0x19) and another with a digital
mic (pin 0x12). Created a new model, acer-dmic, for the latter case
now.
So far, all known models are assigned to be analog-mic, according to
the BIOS setup. If this doesn't match with the actual case, one needs
to try model=acer-dmic, and fix the entry to point ALC268_ACER_DMIC
if it works.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Asus V1s series laptops have an ALC660VD with PCI id: 0x1043, 0x1633.
1.) remove the previous behaviour of mapping that to the ALC861VD_LENOVO
device.
2.) add a new ALC660VD_V1S device based on ALC861VD_LENOVO, with an
added digital out.
Signed-off-by: Tristan Aston <astrotris@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a proper ifdef to shut out a compile warning:
CC [M] sound/pci/hda/patch_intelhdmi.o
sound/pci/hda/patch_intelhdmi.c:286: warning: ‘hdmi_get_dip_index’ defined but \
not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION.
sound/soc/codecs/ad73311.c
This patch removes the said #include <version.h>.
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the hda-reconfiguration patches, the check of empty stream
is gone, and this results in an error with the codec setup with empty
streams.
This patch adds the check again to avoid the error at probing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add NULL-check of the return value of snd_kctl_new1() before
accessing it. Also, make a sanity NULL check to snd_BUG_ON()
for debugging only.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call device_create_file only once in snd_soc_dapm_sys_add function.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda: make a STAC_DELL_EQ option
ALSA: emu10k1 - Add more invert_shared_spdif flag to Audigy models
ALSA: hda - Add a quirk for another Acer Aspire (1025:0090)
ALSA: remove direct access of dev->bus_id in sound/isa/*
sound: struct device - replace bus_id with dev_name(), dev_set_name()
ALSA: Fix PIT lockup on some chipsets when using the PC-Speaker
ALSA: rawmidi - Add open check in rawmidi callbacks
ALSA: hda - Add digital-mic for ALC269 auto-probe mode
ALSA: hda - Disable broken mic auto-muting in Realtek codes
Add support for explicitly enabling the EQ distortion hack for
systems without software biquad support.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk for another Acer Aspier laptop (1025:0090) with ALC883
codec. Reported in Novell bnc#426935:
https://bugzilla.novell.com/show_bug.cgi?id=426935
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed the direct accesses of dev->bus_id in sound/isa/* by replacement
with dev_err() or dev_warn() functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andres Salomon