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>
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>
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>
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>
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
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>
In wm2000_i2c_probe(), if we take the true branch in
"
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
NULL, 0);
if (ret != 0)
goto err_fw;
"
then we'll release_firmware(fw) at the 'err_fw' label. But we've already
done that just a few lines above. That's a use-after-free bug.
This patch restructures the code so that we always call
release_firmware(fw) before leaving the function, but only ever call
it once.
This means that we have to initialize 'fw' to NULL since some paths
may now end up calling it without having called request_firmware(),
but since request_firmware() deals gracefully with NULL pointers, we
are fine if we just NULL initialize it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To print a value of type size_t one should use %zd, not %d.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent change of the power-widget handling for IDT codecs caused
the silent output from the docking-station line-out jack. This was
partially fixed by the commit f2cbba7602
"ALSA: hda - Fix the lost power-setup of seconary pins after PM resume".
But the line-out on the docking-station is still silent when booted
with the jack plugged even by this fix.
The remainig bug is that the power-widget is set off in stac92xx_init()
because the pins in cfg->line_out_pins[] aren't checked there properly
but only hp_pins[] are checked in is_nid_hp_pin().
This patch fixes the problem by checking both HP and line-out pins
and leaving the power-map correctly.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42637
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user reports that model=auto works fine for him. Using
model=auto bring in new features such as jack detection notification
to userspace.
Alsa info is available at http://paste.ubuntu.com/805351/
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 2ae66c2655
ALSA: hda: option to enable arbitrary buffer/period sizes
introduced a regression on machines with Intel controller and Nvidia
HDMI. The reason is that the driver modifies the global variable
align_buffer_size when an Intel controller is found, and the Nvidia
HDMI controller is probed after Intel although Nvidia chips require
the aligned buffers.
This patch fixes the problem by moving the flag into the local struct
so that it's not affected by other controllers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42567
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix below build warning when CONFIG_HOTPLUG is not set.
CC sound/soc/codecs/alc5623.o
sound/soc/codecs/alc5623.c:1062: warning: ‘alc5623_i2c_remove’ defined but not used
CC sound/soc/codecs/alc5632.o
sound/soc/codecs/alc5632.c:1112: warning: ‘alc5632_i2c_remove’ defined but not used
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to table based DAPM and control init.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should be allowing a 5ms delay after the charge pump is started in
order to ensure it has finished ramping.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Visstrim_M10 has an external class D amplifier.
This patch provides support for controlling the 4
possible gain levels and per channel muting.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The problem was introduced due to the obscure formatting some of the
older drivers use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a driver is using regmap directly ensure that we're coherent with
non-ASoC register updates by using the regmap API directly to do our
read/modify/write cycles. This will bypass the ASoC cache but drivers
using regmap directly should not be using the ASoC cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On a mx28evk board the following errors happens on mxs-sgtl5000 probe:
[ 0.660000] saif0_clk_set_rate: divider writing timeout
[ 0.670000] mxs-sgtl5000: probe of mxs-sgtl5000.0 failed with error -110
[ 0.670000] ALSA device list:
[ 0.680000] No soundcards found.
This timeout happens because clk_set_rate will result in writing to the DIV bits
of register HW_CLKCTRL_SAIF0 with the saif clock gated (CLKGATE bit set to one).
MX28 Reference states the following about CLKGATE:
"The DIV field can change ONLY when this clock gate bit field is low."
So call clk_prepare_enable prior to clk_set_rate to fix this problem.
After this change the mxs-saif driver can be correctly probed and audio is functional.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is more what the device model wants us to do and will allow use by
non-audio functions before the audio part of the device has come up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will allow us to move the interrupt allocation out of the ASoC part
of the driver and simplifies the locking by removing any reliance in the
bulk of the interrupt path on the big CODEC lock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McPDM internal FIFO is 24 bit wide. From the 32 bit sample
8 bit is discarded. Let application know about this
via msbits constraint.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Core can set the msbits constraint in behalf of the dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Core can set the msbits constraint in behalf of the dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Core can set the msbits constraint in behalf of the dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Throughout the sgtl5000 driver source code and also in the sgtl5000 datasheet
the revision code is shown in hexadecimal.
Print it hex format, for consistency.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If an interrupt is supplied then use it for thermal warning and FLL lock
notifications. When using the interrupt raise the timeout for the FLL lock
substantially to reduce the chances of spurious warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Most devices accept data in formats that don't correspond directly to
their internal format. ALSA allows us to set a msbits constraint which
tells userspace about this in case it finds it useful (for example, in
order to avoid wasting effort dithering bits that will be ignored when
raising the sample size of data) so provide a mechanism for drivers to
specify the number of bits that are actually significant on a DAI and
add the appropriate constraints along with all the others.
This is done slightly awkwardly as the constraint is specified per sample
size - we loop over every possible sample size, including ones that the
device doesn't support and including ones that have fewer bits than are
actually used, but this is harmless as the upper layers do the right thing
in these cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This must be a leftover from a previous driver.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.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>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Much more compact, both in terms of source and especially in terms of
RAM used at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Visstrim_M10 board uses an external microphone
that can be enabled/disabled by the user
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Both the Snapper CL15 and EDB93xx audio drivers set the same audio
configuration in ep93xx_i2s_acquire. Remove the arguments to
ep93xx_i2s_acquire so that the audio drivers no longer need the
EP93XX_SYSCON defines exported.
Cc: Hartley Sweeten <hsweeten@visionengravers.com>
Cc: Mika Westerberg <mika.westerberg@iki.fi>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
remove unnecessary inclusion of machine specific header
file mach/dm365.h from cq93vc.c voice codec driver
which comes in the way of platform code consolidation.
Signed-off-by: Manjunath Hadli <manjunath.hadli@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is useful to create dummy codec devices where we need to have some
DAI links without a real Codec. e.g. could be used to represent dumb FM,
MODEM, etc
This is also used by dynamic PCM for DAI links that have no codec.
Signed-off-by: Liam Girdwood <lrg@ti.com>
[Fixed the indentation -- broonie]
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a method for mach drivers to query the HS DC offset step size in mV.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Print out the offending DAI link entry when a naming error occurs. Makes
thing easier to debug for machines with a large number of DAI links.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
