Don't use the internal I/O functions directly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Recent enhancements in the bias management means that we might not be
in standby when the CODEC is idle and can have active widgets without
being in full power mode but the shutdown functionality assumes these
things. Add checks for the bias level at each stage so that we don't
do transitions other than the ON->PREPARE->STANDBY->OFF ones that the
drivers are expecting.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Add a stereo volume control for analog input channel pair 1/2.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mute control for every analog output channel.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the soft log-conversion and add a dB scale according to
the DAC documentation instead.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused driver version information from the individual files.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Early revisions of several of the WM8994 variants have register updates
to improve performance. Move these over to using the regmap patch system
instead of open coding them in the audio driver. Since the regmap init
is done by the MFD the code is moved there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Request the DMA channel in the PCM open callback instead of the hwparams
callback, this allows us to let open fail if no dma channel is available. This
also fixes a bug where the channel will be requested multiple times if hwparams
is called multiple times.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Request the DMA channel in the pcm open callback. This allows us to let open
fail if there is no dma channel available.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the call to snd_soc_dai_set_dma_data from the hw_params callback to the
startup callback. This allows us to use the dma data in the pcm driver's open
callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are few existing user drivers of dw_dmac. They will break as soon as we
remove unused fields from struct dw_dma_slave. This patch focuses to fix these
user drivers to use dma_slave_config() routine.
Signed-off-by: Viresh Kumar <viresh.kumar@st.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
.device_fc is added in struct dma_slave_config recently. All user drivers, which
want DMA to be the flow controller must pass this field as false. As earlier
driver don't look to use this feature, mark it false for now.
Signed-off-by: Viresh Kumar <viresh.kumar@st.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Added sanity checks in a few places not to assume the pins having the
certain amp caps or the input-source being always assigned to a mux.
No actual bugs have been triggered by these, but surely better to be a
bit more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver may create duplicated mic boost controls when there are
multiple mics with the very same type / location, and this leads to
the error at actual kcontrol creation. It needs to check the validity
of the created control and add a proper index if it's duplicated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 3702b08 added a lock, but did not account for the case of
SNDRV_PCM_POS_XRUN, which would get immediately overwritten.
This could be bundled into one if-else-if statement, but the goto
helps to clarify the 'exceptional' case.
Thanks to Andreas Pape for spotting this.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
on.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
A couple of small, driver specific fixes - nothing too exciting going
on.
Chip designers frequently include things like the enable and disable
controls for algorithms in the register blocks which also hold the
coefficients. Since it's desirable to split out the enable/disable
control from userspace the plain SND_SOC_BYTES() isn't optimal for
these devices.
Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word
of the block to be excluded from the control. This supports the needs
of devices I've looked at and lets us have a reasonably simple API.
Further controls can be added in future if that's needed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow devices to export blocks of registers to the application layer,
intended for use for reading and writing coefficient data which can't
usefully be worked with by the kernel at runtime (for example, due to
requiring complex and expensive calculations or being the results of
callibration procedures). Currently drivers are using platform data to
provide configurations for coefficient blocks which isn't at all
convenient for runtime management or configuration development.
Currently only devices using regmap are supported, an error will be
generated for any attempt to work with a byte control on a non-regmap
device. There's no fundamental block to other devices so support could
be added if required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow us to build infrastructure which needs to know the size of a value
without requiring regmap based drivers to supply this information to both
ASoC and regmap by asking regmap for the value.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When an external capacitor is connected to MICBIAS2 on devices with
jack detection (which is not required but may be done in some systems)
then the loading may mean that better performance is obtained when
the microphone bias is enabled normally rather than using the low power
mode. Provide platform data allowing systems to indicate if they require
this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This minimises the chance of any external capacitors that are fitted
being discharged into headphones as they insert.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few clean-ups for post-static-quirk time:
- Call alc_auto_init_std() statically in alc_init() instead of setting
spec->init_hook in each caller. spec->init_hook field is left
unused for any future use.
- Move the call of set_capture_mixer() to to alc_parse_auto_config()
instead of each caller.
- Get rid of the ADC-filling and imux check in each parser function.
This is no longer needed since the auto-parser always check ADCs and
imux. It was only for the static quirks.
- Kill unused defines
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the machine has two speakers but wants to put more multi-io
jacks, the parser shouldn't consider about the shared DAC but try to
assign the individual DACs. Otherwise the channel mapping would be
fairly confused and lead to the wrong DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the multi-io is added to the two speaker output configuration,
the parser would try to add yet another "Bass Speaker" control since
it checks only cfg->line_outs. Add a workaround for it by simply
passing the channel name in the case of multi-io outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the outputs are using shared DACs, we can still create individual
mute siwtches since they are assigned per pin. This allows to create,
e.g. Speaker and Bass Speaker mute switches while the single volume is
used for these outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When BIOS is damn crazy and gives no pin-config at all, the driver might
lead to a NULL dereference. Let's add a NULL check for such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The below patch fixes some typos "aditional" to "additional", and also fixes
a comment with another word mispelled.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Similarly in patch_via.c, parse the active analog-loopback connections
and create a list dynamically rather than static arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Resitance is futile. The remaining static model quirks for Apple
machines with ALC882-compatible codecs are converted to the auto-parser
now. We can remove all alc*_quirks.c finally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ak4642 driver had wrong dapm settings for headphone L/R.
If you select headphone L, and select R after that,
headphone L setting was removed by R settings.
This patch fixes it up.
It provides just "Headphone Enable" to user side
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the printk(KERN_ERR* instances with dev_err in the driver.
While we are here clean up some of the debug messages as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8775 is register compatible with the WM8776 so can be supported with
the same driver though it is an ADC only part. Add the device ID to the
WM8776 driver, further updates will be added in the future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If anyone wants to debug the driver and avoid the existing fix-ups,
pass model=nofixup option. Then the driver will skip to pick up the
fixup list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally the all static quirks for ALC880 are converted to the
auto-parser. Since we are never sure whether the BIOS on so many old
machines are really correct, the quirk table entries are copied as
they are, but just providing the proper pin-config values
accordingly.
Since alc880_quirks.c is removed, alc882_quirks.c has to be adjusted
slightly to be built again. There might be some compile warnings due
to the remaining alc882 quirks, but these shall be killed sooner or
later, I don't care it much at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that BIOS on most of ASUS mobo's set the pin-config tables
reasonably well for the auto-parser. We'd need GPIO setups, but should
work as is other than that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z71V has a totally broken BIOS setup (at least the info I got),
thus we need to override the whole pin-config table to make the
auto-parser working correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=uniwill would work almost as is, but a couple of adjustments
are needed to make the mutli-io working correctly. The headphone and
speaker pins have to be marked properly in pin configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Uniwill p53 has a sane BIOS setup but just needs the volume-knob handling
like Fujitsu laptops with ALC880.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar as the previous patch for model=fujitsu, we can now move the
static quirk for F1734 to the auto-parser. The only difference is the
default pin configurations: F1734 has less pins than Amilo's.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SND_PCM is already selected by SND_SOC, there is no need for
SND_IMX_SOC and SND_MXS_SOC to select it again.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now adding the support for the volume-knob widget, we can move the static
quirk for ALC880 model=fujitsu to the auto-parser completely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the call of alc_apply_fixup() with ALC_FIXUP_ACT_PROBE after the
whole setups of patch_ops & co, so that the fix-up function may override
the default setup. This will be needed for installing the own unsol
event handler (e.g. for volume-knob controls).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a patch to the sound/core/misc.c file that
fixes up a trailing white space issue found by the
checkpatch.pl tool.
Signed-off-by: Jeffrin Jose <ahiliation@yahoo.co.in>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch supports DMAEngine to FSI driver.
It supports only Tx case at this point.
If platform/cpu doesn't support DMAEngine, FSI driver will
use PIO transfer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Merge tag 'v3.3-rc4' into for-3.4 in order to resolve the conflict
resolved below within the FSI driver and allow the application of the
dmaeengine conversion that depends on this resolution.
Linux 3.3-rc4
Conflicts:
sound/soc/sh/fsi.c
This fixes below build warning:
WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe()
The function pxa2xx_ac97_probe() references
the function __devinit pxa2xx_ac97_hw_probe().
This is often because pxa2xx_ac97_probe lacks a __devinit
annotation or the annotation of pxa2xx_ac97_hw_probe is wrong.
Also rename pxa_ac97_dai to pxa_ac97_dai_driver to fix below build warning:
LD sound/soc/pxa/built-in.o
WARNING: sound/soc/pxa/built-in.o(.data+0x18c): Section mismatch in reference from the variable pxa_ac97_dai to the function .devinit.text:pxa2xx_ac97_probe()
The variable pxa_ac97_dai references
the function __devinit pxa2xx_ac97_probe()
If the reference is valid then annotate the
variable with __init* or __refdata (see linux/init.h) or name the variable:
*driver, *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Showing the returned values on error messages is useful information.
While at it, use pr_err/pr_warn whenever possible.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While it matches the current code only bringing the device out of reset
isn't actually doing what the function says so make sure we set the GPIO
high before we pull it low.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clevo machines with ALC880 are all well with proper BIOS setup.
It seems still requiring the additional COEF setup for the EAPD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactor the DAC filling function to be used for both the primary
line outputs and extra outputs using the individual badness tables.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Medion W810 with ALC880 has a typical BIOS bug, copying the
pin-defaults without disabling the unused pins. At least, the pin
0x17 must be disabled. Also, it requires GPIO-2 setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This means we don't need to walk through every single widget in the system
for each stream event which is a bit less silly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In order to allow us to do smarter things with DAI links create DAPM
widgets which directly represent the DAIs in the DAPM graph. These are
automatically created from the DAIs as we probe the card with references
held in both directions between the widget and the DAI.
The widgets are not made available for direct instantiation by drivers,
they are created automatically from the DAIs. Drivers should be updated
to create stream routes using DAPM maps rather than by annotating AIF
and DAC widgets with streams.
In order to ease transition to this model from existing drivers we
automatically create DAPM routes between the DAI widgets and the existing
stream widgets which are started and stopped by the DAI widgets, though
the old stream handling mechanism is still in place. This also has the
nice effect of removing non-DAPM devices as any device with a DAI
acquires a widget automatically which will allow future simplifications
to the core DAPM logic.
The intention is that in future the AIF and DAI widgets will gain the
ability to interact such that we are able to manage activity on
individual channels independantly rather than powering up and down the
entire AIF as we do currently.
Currently we only generate these for CODECs, mostly as I have no systems
with non-CODEC DAPM to integrate with. It should be a simple matter of
programming to add the additional hookup for these.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Neater and avoids warnings when used in other places where const strings
are desired.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In order to allow us to do something smarter than iterate through widgets
doing strcmp() to work out what to power up for stream events change the
interface used to generate them to be based on the combination of a DAI
and a stream direction rather than just a simple string identifying the
stream.
At some point we'll probably want a set of channels too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Let the caller fiddle with the widget after we're done in order to
facilitate further refactoring.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Everything now uses snd_soc_dapm_new_controls() instead so we don't need
to make it part of the external API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This fixes below build warning:
CC sound/soc/pxa/spitz.o
sound/soc/pxa/spitz.c: In function 'spitz_startup':
sound/soc/pxa/spitz.c:116: warning: passing argument 1 of 'spitz_ext_control' from incompatible pointer type
sound/soc/pxa/spitz.c:47: note: expected 'struct snd_soc_card *' but argument is of type 'struct snd_soc_codec *'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALC880 model=lg could work fine with the auto-parser due to the recent
rewrite, but it still needs the manual adjustment; namely, the BIOS leaves
unused pins as some real active jacks. This confuses the parser.
Thus we just cover these pins and override the pin-configs as a fix-up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Try harder to fit the multi-io pins also by checking the hard-wired
connections for multi-ios. Also, the badness values are adjusted to
prioritize the multi-ios as more valuable. These changes will enable
the multi-io on some machines without losing the current capability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix for the speaker on Acer Aspire 59135 introduced
another problem for surround outputs. It changed the connections on
the line-in/mic pins for limiting the routes, but it left the modified
connections. Thus wrong connection indices were written when set to
4ch or 6ch mode.
This patch fixes it by restoring the right connections just after
parsing the tree but before the initialization.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows userspace control of final power off, allowing policy decisions
for register configuration retention.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only enable and disable the FLL when explicitly told to, supporting some
additional use cases and making the driver behaviour more standard.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we update for any changes in cases where we reconfigure while
live (eg, for analogue bypass).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, the Realtek driver tires to assign the single-connected routes
for all pins only once at the beginning. However, since some DACs have
been already mapped, the rest pins might have also single conections.
In this patch, the driver does the single-connection assignment in a
loop until all possbile single-connections are checked. This will
improve the DAC assignment, e.g. for ASUS G72.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the Realtek auto-parser for assigning the DACs and
mixers in more suitable ways by evaluating the assignment with "badness"
calculations.
When assigning a DAC hinders the assignment of individual DACs for
other pins, some badness point is given. Similarly, when it blocks the
assignment of unique mixer controls, another badness point is added.
Also, if no DAC, even shared DAC, can be assigned, more badness is
pointed. Finally, comparing the accumulated badness, the best route is
chosen among several trials.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the connections from the pin selector contain only two
widgets, a route to DAC and the aa-mixer, it's certainly a
single connection. In such a case, get_dac_if_single() should
return the connected DAC, too.
This will improve the detection of the individual DAC
assignment for each pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Machine kcontrols now use card instead of codec for thier "chip".
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The bitmap introduced in the commit [527e4d73: ALSA: hda/realtek - Fix
missing volume controls with ALC260] is too narrow for some codecs,
which may have more NIDs than 0x20, thus it may overflow the bitmap
array on them.
Just double the number to cover all and also add a sanity-check code
to be safer.
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've enabled the static fixups for ASUS machines with ALC269 codec,
just for making things compatible during the transition to the auto-
parser. However, it seems that the static configurations do more harmful
than good, as some of entries don't match with the actual hardware setups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we can clean up all static quirks for ALC260.
Also many codes in alc_quirks.c can be ripped off since they have been
used only by ALC260 static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Presario B1900 needs a similar hack like Replacer, toggling GPIO1
per the jack state, in addition to the COEF setup used for other Acer
laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The support for Replacer 627V in the auto-parser needs the unique unsol
event handling: although the machine has a single output pin 0x0f, it's
used for both the headphone and the speaker, and the driver needs to
toggle the output route via GPIO 1.
In addition, it needs a special COEF setup with 0x3050.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALC260 model=acer needs GPIO1 setup. It could be selected well
if the codec SSID is set properly by BIOS, but to make sure, enable it
forcibly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=will for ALC260 requires the pin 0x0f to be a headphone and
some special verbs for the COEF to turn on the amp. Now added these as
fixup entries and removed the static model quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show the id we read when the id mismatch is detected.
This is useful for debugging.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform widgets to be visible in debugfs like codec widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the addition of the non-CODEC control adds card controls like the
DAPM pin switch have been broken as they are expecting the private data
for the control to be the CODEC but it's now the card. Fix that for the
pin switches, an audit of other drivers is required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Update the codec debugfs initialisation to use dev_warn() instead of
printk(KERN_WARNING).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when DAPM widgets are power sequenced the stream_event()
completion callback is only called for the stream_event originator
DAPM context. Other components in the card may also be interested so
make sure they are also notified of any widget power events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's useful to export the DAPM reset as a static function for future use
by other DAPM functions. e.g. The dynamic PCM query widgets resets the DAPM
graph before working out active paths.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A malicious USB device could feed in a large nr_rates value. This would
cause the subsequent call to kmemdup() to allocate a smaller buffer than
expected, leading to out-of-bounds access.
This patch validates the nr_rates value and reuses the limit introduced
in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow
in parse_uac2_sample_rate_range()").
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling "propably" to "probably" and "activ" to "active"
in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The main role of the WM9081 is as a class D speaker amplifier so there
is no concern about pops. There are also very few registers and a fast
power up time so we can happily mark the driver as idle_bias_off.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://bugs.launchpad.net/bugs/930842
The reporter states that audio is inaudible by default without muting
'External Amplifier'. Add a quirk to handle his SSID so that changing
the control is not necessary.
Reported-and-tested-by: Benjamin Carlson <elderbubba0810@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
driver.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
A simple fix from Morimoto-san for the pointer() operation in the FSI
driver.
ALC5632 codec supports digital microphone. This patch adds DAPM widget.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Conflicts:
sound/pci/hda/patch_realtek.c
Merged back the fix for Acer Aspire 6935 with ALC889 codec.
The fix commit was based on 3.2 kernel so that it can be applied to
stable kernel cleanly.
Since 3.2 kernel, the driver starts trying to assign the multi-io DACs
before the speaker, thus it assigns DAC2/3 for multi-io and DAC4 for
the speaker for a standard laptop setup like a HP, a speaker, a mic-in
and a line-in. However, on Acer Aspire 6935, it seems that the
speaker pin 0x14 must be connected with either DAC1 or 2; otherwise it
results in silence by some reason, although the codec itself allows
the routing to DAC3/4.
As a workaround, the connection list of each pin is reduced to be
mapped to either only DAC1/2 or DAC3/4, so that the compatible
assignment as in kernel 3.1 is achieved.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1705 codec has two ADCs where the secondary ADC has no MUX but only
a fixed connection to the mic pin. This confused the driver and it
tries always overriding the input-source selection by assumption of
the existing MUX for the secondary ADC, resulted in resetting the
input-source at each time PM (including power-saving) occurs.
The fix is simply to check the existence of MUX for secondary ADCs in
the initialization code.
Tested-by: Anisse Astier <anisse@astier.eu>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few machines with ALC861 & co are reported not to work properly with
the auto-mute feature in software. The auto-mute feature is implemented
in the hardware level, and the jack-detection never works with them.
Also, rename the fixup index as ALC861_FIXUP_* to follow the standard.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to indicate that the codec has no jack-detection cap.
This flag should be set for hardwares that have no jack-detect
implementation although the codec chip itself supports it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Raise the ramp time to 50ms to cover corner cases, use the startup bias
generator, explicitly reset the ramp circuit when complete and reorder
things all of which should improve performance somewhat for systems that
are sensitive to noise from VMID.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The enable of the single ended line outputs on wm_hubs devices performs
better if the output is enabled prior to starting VMID. Since inactive
outputs are held at VMID anyway there is little cost to doing this for
unused outputs. Add callbacks into wm_hubs and keep track of which outputs
are really active so we can disable them once we're active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the mixer path diagram input sources' attenuators logically
belong to the playback path and DMIC boost only affects capture.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This brings the TLVs in sync with the documentation and allows to properly
manipulate mic boost controls with alsamixer.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch should fix output through speakers using HP mixer.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix I2S digital interface power for recording.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DMIC switches and controls to ALC5632 codec.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no Phone Mix<->Mono Mix route in datasheet.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename capture switches to common scheme XXX2REC.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove extra zero from volume step in DECLARE_TLV_DB_SCALE macro.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses these functions for data that is allocated in
the probe function of a platform device and is only freed in the remove
function.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix below build error which is introduced by
commit 022658 "ASoC: core: Add support for DAI and machine kcontrols".
CC [M] sound/soc/omap/n810.o
sound/soc/omap/n810.c: In function 'n810_set_input':
sound/soc/omap/n810.c:194: error: 'codec' undeclared (first use in this function)
sound/soc/omap/n810.c:194: error: (Each undeclared identifier is reported only once
sound/soc/omap/n810.c:194: error: for each function it appears in.)
sound/soc/omap/n810.c:188: warning: unused variable 'card'
make[3]: *** [sound/soc/omap/n810.o] Error 1
make[2]: *** [sound/soc/omap] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Correct spelling "memroy" to "memory" in
sound/soc/codecs/twl4030.c
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
.. the number of the half-beast?
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Merge tag 'v3.3-rc3' as we've got several bugfixes in there which are
colliding annoyingly with development.
Linux 3.3-rc3
.. the number of the half-beast?
Conflicts:
sound/soc/codecs/wm5100.c
sound/soc/codecs/wm8994.c
current fsi_pointer() calculation was not correct for FSI driver.
This patch fix it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's set unconditionally.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
This is usually not a use case dependant flag anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since we've already got logic to special case immediate teardown of the
stream we may as well use it if the pmdown_time has been set to zero by
the application layer instead of scheduling a work item with zero delay.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The new HP laptops turns off the mute LED with VREF50 or VREF80, but
not in HIZ unlike the previous models. Since VREF50 (also 80) works
with the previous models, let's use VREF50 for all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new ASoC dynamic PCM core needs to create PCMs and substreams that are
for use by internal ASoC drivers only and not visible to userspace for
direct IO. These new PCMs are similar to regular PCMs expect they have no
device nodes or procfs entries. The ASoC component drivers use them in exactly
the same way as regular PCMs for PCM and DAI operations.
The intention is that a dynamic PCM based driver will register both regular
PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace
however the internal PCMs will be used by the driver to route digital audio
through numerous back end DAI links (with potentially a DSP providing different
hw_params, DAI formats based on the regular front end PCM params) to devices
like CODECs, MODEMs, Bluetooth, FM, DMICs, etc
This patch adds a new snd_pcm_new_internal() API call to create the internal PCM
without device nodes or procfs. It also adds adds a new internal flag to snd_pcm.
[fixed minor coding-style issues by tiwai]
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
that were sent. They collide with some -next work so I'd really like to
get them into 3.3-rc3 if possible to merge back up into the -next code.
All driver specific and unexciting in the grand scheme of things.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
A few small WM8994 updates to go on top of the previous lot of things
that were sent. They collide with some -next work so I'd really like to
get them into 3.3-rc3 if possible to merge back up into the -next code.
All driver specific and unexciting in the grand scheme of things.
The VMID ramp rate is supposed to be 0x3, not 11b. Fix that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Make sure we generate enough BCLKs for I2S style modes by always having
a minimum of two channels worth of clocks for the BCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver accidentally exchanged the left/right fields for stereo AC'97
mixer registers. This affected only the aux and CD inputs because the
line input bypasses the AC'97 codec and the mic input is mono; cards
without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected.
Reported-and-tested-by: Abby Cedar <abbycedar@yahoo.com.au>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.31+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio DeviceIDs for the Intel Lynx Point PCH.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No practical impact but now we have the control type we may as well use
it for the slightly nicer sequencing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This hardware requires same fixup for the node 0x0f like Asus A6Rp.
More information: https://bugzilla.redhat.com/show_bug.cgi?id=785417
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we're not implementing GPIO or microphone detection support
there's no users for the read() function any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
In patch_ca0132.c, the error returned from chipio_write() isn't checked
always. Also, the power-up/down sequence isn't tracked properly in some
error paths.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the Android suspend ignore code for idle_bias_off CODECs. That one is
actually a regression fix as some of the new power savings that have
been introduced confused the suspend ignore code, making devices that
are active for non-audio reasons look like they are idle causing them to
be suspended instead of being kept active.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
The only particularly remarkable change here is the one for handling of
the Android suspend ignore code for idle_bias_off CODECs. That one is
actually a regression fix as some of the new power savings that have
been introduced confused the suspend ignore code, making devices that
are active for non-audio reasons look like they are idle causing them to
be suspended instead of being kept active.
Make the mechanism used for WM8994 more like that for WM1811 and WM8958:
provide the logic to distinguish between headphone and headset and hard
code the reporting of sensible SND_JACK values. Should integration with
other detection mechanisms be required we can add appropriate callbacks
(though some integrations should be able to use the subsystem ones).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise we might get an error if the GPIO is configured as an input
since that makes the register volatile and a suspended device can't be
read from.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the default state that the runtime PM infrastructure expects so
instead just kick the runtime PM core to suspend us if we're not doing
anything (as is default).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build breakage by using the correct API call to add kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Looks like nobody is or will be using this code.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix some spelling mistakes in the header and remove the todo items. Most
todo items are now available as kcontrol options now anyway.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow for the operation of custom mixer and mux DAPM widgets that can call
snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly
after updating their status. This is useful with complex DAPM Mixer operations
where we need to do additional work in addition to setting a few mixer register
bits.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In cases where we should enter STANDBY DAPM will power us up, otherwise
there is no need to power up and we can remain in OFF.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 has a different accessory detect architecture, call its setup
function too. We ignore the errors and the driver will check for chip type
so it's safe to call the setup functions for both architectures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can enable VMID independently of the bias in some use cases so we need
to ensure that the core device is powered up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
sparse complains that "spec->multiout.dac_nids" is a pointer.
sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>)
sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids
sound/pci/hda/patch_realtek.c:2321:37: right side has type int
It was meant to be num_dacs instead of dac_nids.
Although the current code still works as expected (when num_dacs is zero,
dac_nids should be NULL, too), better to fix now, of course.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The header 'linux/moduleparam.h' is included twice in
'sound/isa/sb/emu8000_patch.c'. Once is enough.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.
This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-
snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()
This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().
It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().
Finally, it updates the existing machine drivers that register controls to either :-
1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.
In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In most cases, the slave strings for vmaster are identical between
volumes and switches except for "xxx Volume" and "xxx Switch" suffix.
Now snd_hda_add_vmaster() takes the optional suffix argument so that
each string can be composed with the given suffix, and we can share the
slave name strings in both volume and switch calls nicely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes in Realtek auto-parser added the new "Bass Speaker"
and "CLFE" mixer elements which should be tracked as vmaster slaves,
too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42720
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI got each PortA/B parameter by porta_flags/portb_flags from platform.
And .set_rate function was shared for PortA/B.
This structure was not readable and not flexible.
This patch adds sh_fsi_port_info, and its own settings was added on each platform.
it is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
is_play should be kept as local valuable.
it prepare cleanup for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds struct fsi_stream_handler and defined fsi_pio_push/pop_handler.
these are controled by fsi_steam_xxx() function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_get_current_fifo_samples() uses fsi_stream instead of is_play.
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch re-define fsi_is_play() and fsi_stream_is_play().
fsi_data_pop/push() function keeps direct value of "is_play" at this point,
but it will be removed soon.
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify fsi_pio_get_area() parameter to use
struct fsi_stream, and used it on fsi_fifo_data_ctrl().
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_fifo_init() is called only from fsi_hw_startup()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Next transfer data size of "push" and "pop" had calculated on shared function.
But it was not readable code.
This patch divided it into for push, and for pop.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch gathered fsi_stream_xxx() functions in order to make it readable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Lenovo laptops with cx5066 chips seem to work better with
model=auto. Let's get rid of the fallback to the wrong model.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738397
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All DAPM routes are configured via device tree, and there is no need in
DAPM route structures in board file.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding device tree support of headset autodetection on PAZ00 board.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the standard logging macros and use dev_ variants where we can, also
reporting error codes whenever we report an error. These changes (the
error codes in particular) make it noticeably easier to figure out what
went wrong just from the basic dmesg output.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since the dynamic pin power-control and the analog low-current mode
may lead to pop-noise, it's safer to set it off as default.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an attempt to fix S3-resume problems reported for a few
laptops with different Conexant codecs. They show the communication
stalls at some time in S3, and the driver falls back into the
single-cmd mode. This leads to the silent output or the lack of
auto-mute feature.
As a workaround, here enables the sync_write and the bus-reset flags
to make the communication more stable.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740115
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738397
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs have several different power-saving features, and one of
them is the analog low-current mode. But it turned out that the ALC
mode causes pop-noises at each on/off time on some machines. As a
quick workaround, disable the ALC when another power-saving feature,
the dynamic pin power-control, is turned off, too, since the dynamic
power-control is already exposed as a mixer enum element so that user
can turn it on/off freely.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The analog low-current mode must be enabled when the no stream is
running but the current detection checks it in a wrong way.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of always writing AC_VERB_SET_POWER_STATE, check the current
power-state and don't write again if the value is already set.
This may reduce the click noise upon the dynamic power-state change
(e.g. in analog-input mixer).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The second line output mixer has the controls for the line input bypasses
in the opposite order.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing
this manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This typo caused the wrong codec's nid to be checked for wcaps type.
As a result, sometimes speakers would duplicate the output sent to
HDMI output.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/924320
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we get an indeterminate impedance with both headset polarities then
give up and report the accessory as a headphone rather than continually
retrying.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
.. several days delayed. No reason, I just didn't think of it.
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Merge tag 'v3.3-rc2' into for-3.4
A reasonable amount of new development is causing fiddly merge conflicts
between different resource management changes (mostly fixing bugs in
resource management due to noticing things while doing enhancements in
the same area).
Linux 3.3-rc2
.. several days delayed. No reason, I just didn't think of it.
If cs_automic is called twice (like it is during init) while the mic
is present, it will over-write the last_input with the new one,
causing it to switch back to the automic input when the mic is
unplugged. This leaves the driver in a state (cur_input, last_input,
and automix_idx the same) where the internal mic can not be selected
until it is rebooted without the mic attached.
Check that the mic hasn't already been switched to before setting
last_input.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements device tree support for Tegra boards with ALC5632
codec.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For later wm_hubs devices we have much less need to keep the biases up
even when using single ended line outputs so flag idle_bias_off for
everything except the WM8993 and WM8994.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the same pm_ops for all system suspend and resume paths. This isn't
ideal for suspend to disk with older CODECs as we'll suspend and then
resume the CODEC before powering off all of which takes a long time due
to VMID ramps but it's very simple to implement and for modern CODECs the
overhead should be minimal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If an idle_bias_off device is in any state other than off then it is still
active for some reason (typically a low power function such as accessory
detection). This wasn't an issue when the feature was implemented as we
always went to _ON for any active function, subsequent power improvements
have changed things.
With the modern way of doing things we should overhaul the infrastructure
to allow devices to explicitly take references for these functions but
that's a much more invasive change and will require driver updates to
deploy, this will bring the framework into line with the existing driver
set before we do that work.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we've switched over to regmap the ASoC level cache sync will be
ineffectual and potentially harmful as there is no longer an ASoC level
cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a codec has both a front and a rear Line In, two controls both
named "Line Jack" will be created, which causes parsing to fail.
While a long term solution might be to name the jacks differently,
this extra check is consistent with what is already being done in many
auto-parsers, and will also protect against other cases when two
inputs have the same label.
BugLink: https://bugs.launchpad.net/bugs/923409
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1
and IN2 to LINEOUT2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The runtime PM core no longer requires any callbacks so don't provide
empty ones for it any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Mute Switch should be called Playback Switch to match the volume
and the regmap core has been updated so we don't need to mark all the
registers as non-volatile.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Add a driver supporting the volume control and the mute pin. Shdn pin
and DAPM are not taken care of yet.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It turned out that other ASUS laptops require the similar fix to
enable the VREF on the pin 0x0f for the secret output amp, not only
ASUS A6Rp. Moreover, it's required even when the pin is being used
as the output. Thus, writing a fixed value doesn't work always.
This patch applies the VREF-fix for all ASUS laptops with ALC861/660
in a fixup function that checks the current value and turns on only
the VREF value no matter whether input or output direction is set.
The automute function is modified as well to keep the pin VREF upon
muting/unmuting via pin-control; otherwise the pin VREF is reset at
plugging/unplugging a jack.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Openmoko GTA01 machine has been removed from the machine ID database,
so we need to remove references to it as well.
Without that fix we have:
sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’:
sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’
Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8962 has ground referenced and class D outputs so can happily go
down to BIAS_OFF without a large startup time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix silent output on Haier W18 laptop
ALSA: hda: set mute led polarity for laptops with buggy BIOS based on SSID
ALSA: hda - Fix silent output on ASUS A6Rp
ALSA: Fix memory leak on error in snd_compr_set_params()
ALSA: ymfpci - Don't create invalid PCM & mixers when AC97 doesn't support
This patch converts at73c213 to use the module_spi_driver()
macro which makes the code smaller and a bit simpler.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch converts onyx and tas codec drivers to use the module_i2c_driver()
macro which makes the code smaller and a bit simpler.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user reports that he needs to add model=auto for audio to
work properly. In fact, since node 0x15 is not even a pin node,
the existing fixup is definitely wrong. Relevant information can
be found in the buglink below.
Cc: stable@kernel.org (3.2+)
BugLink: https://bugs.launchpad.net/bugs/918254
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC
drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver
for devices can unconditionally support idle_bias_off.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For optimal performance the single ended line outputs require that the
line output VMID buffer be enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Report any accessory with a low impedance as a headphone, previously
anything other than a short or microphone would not be reported at all.
The most likely reason is a microphone with incorrect polarity.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we get an indeterminate impedance with both headset polarities then
give up and report the accessory as a headphone rather than continually
retrying.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using pm_runtime to decide if the device should go into full power down
has the dual advantage of allowing easier integration with non-DAPM
reasons to power on the device (like the FLL) and allowing userspace to
control the final power down which is useful for tuning retention of
DSP firmware.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A signal generator has no power control itself and so shouldn't cause a
power up of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We don't really care if any action is taken immediately so let the PM
core defer things if it wants to.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM2200 is a low power mobile CODEC with enhanced Wolfson myZone
Ambient Noise Cancellation (ANC) intended for mobile telephony
applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modern devices allow systems to enable and disable individual supplies on
the device, allowing additional power saving by switching off regulators
which power portions of the device which are not currently in use. Add a
new SND_SOC_DAPM_REGULATOR_SUPPLY widget type factoring out the code for
managing such widgets from individual drivers.
The widget name will be used as the supply name when requesting the
regulator from the regulator API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The very same problem is seen on Haier W18 laptop with ALC861 as seen
on ASUS A6Rp, which was fixed by the commit 3b25eb69.
Now we just need to add a new SSID entry pointing to the same fixup.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42656
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable ASoC audio support for OMAP4 based machines with
twl6040 codec via the omap-abe-twl6040 machine driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoid using hardwired configuration for MCLK frequency.
Different board design might use other MCLK frequency.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Disable the not connected pins on the board based on the
received platform data.
DO not register the jack function on boards, which does not have
means to detect it (jack is always connected).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can have machines without DMIC connected. In this case there is
no need to create amother (unusable) capture PCM on the card.
The existence of the DMIC connection can be checked via
pdata->has_dmic.
Select the correct dai_link structure for the card based on
pdata->has_dmic.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SDP4430 is a reference platform, and as such it has all possible
audio routing implemented.
Correct the DAPM routing to be complete.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to avoid breakage change the omap-abe-twl6040
machine driver's routing.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert the OMAP4 ABE/TWL6040 machine driver to platform
driver.
For the card name use the string provided via platform data.
The card's name for OMAP4 SDP4430 has been changed:
SDP4430 -> OMAP4-SDP4430
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the internal prefixes within the driver from sdp4430.
At he same time correct the Kconfig text as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The same machine driver will support other boards
with similar audio configuration (OMAP4, ABE, twl6040).
Rename the driver to have more generic name.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error
ASoC: wm8958: Use correct format string in dev_err() call
ASoC: wm8996: Call _POST_PMU callback for CPVDD
ASoC: mxs: Fix mxs-saif timeout
ASoC: Disable register synchronisation for low frequency WM8996 SYSCLK
ASoC: Don't go through cache when applying WM5100 rev A updates
ASoC: Mark WM5100 register map cache only when going into BIAS_OFF
ASoC: tlv320aic32x4: always enable analouge block
ASoC: tlv320aic32x4: always enable dividers
ASoC: sgtl5000: Fix wrong register name in restore
We need to write the configuration for each microphone to a different
register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
When we have identified an accessory make sure we've flagged that we've
done so in order to make sure we always report buttons and don't continue
to polarity flip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As per discussion we can safely ignore the 8 and 16 bit sample
sizes when applying the msbits constraint.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
HP laptop models with buggy BIOS are apparently frequent, including
machines with different codecs. Set the polarity of the mute led based
on the SSID and include an entry for the HP Mini 110-3100.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: Predrag Ivanovic <predivan@open.telekom.rs>
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The refactoring of Realtek codec driver in 3.2 kernel caused a
regression for ASUS A6Rp laptop; it doesn't give any output.
The reason was that this machine has a secret master mute (or EAPD)
control via NID 0x0f VREF. Setting VREF50 on this node makes the
sound working again.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hardware is configured with one or both of the IN4 inputs used
for DC measurement (with no DC blocking capacitor connected) then we can
improve power consumption slightly in idle modes by applying a register
write sequence. Provide platform data to enable this, implemented using
a regmap patch.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If copy_from_user() does not return 0 we'll leak the memory we
allocated for 'params' when that variable goes out of scope.
Also a small CodingStyle cleanup: Use braces on both branches of
if/else when one branch needs it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- check SDAC bit of AC97 primary codec when create "rear" device 3,
"4ch" device 2 and "4ch Duplication" switch as the card need a four channels
AC97 codec to support surround40.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add "PCM Playback Volume" controls for 16 playback subdevices
This allow application to change the volume of each subdevice
by using hardware mixer of au88x0 and default is zero gain/attenunation.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
