These scales should be regular, not linear.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

These should be regular rather than linear scales.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

It's not needed and the version number never gets updated anyway.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NBarry Song <21cnbao@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.36

These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.

Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: NIlkka Koskinen <ilkka.koskinen@nokia.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add support for NUC900 AC97
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix build warning about unused ops and add ops
to the sdp4430 DAI link.
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Enable earphone speaker in sdp4430 machine driver.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add ASoC support for TI SDP4430.
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
Signed-off-by: NFelipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
Signed-off-by: NArnaud Patard <apatard@mandriva.com>
Acked-by: NLiam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

McBSP module in OMAP4 needs to be able to set its tx/rx threshold
and enable the transmitter/receiver when starting an audio stream.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

In OMAP4, there is only one irq line for TX and RX paths. Use
the correct irq line to avoid errors at runtime.

Also, request irq line only once (instead of requesting for TX
and RX).
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.
Signed-off-by: NSergey Lapin <slapin@ossfans.org>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Log the values we're getting back from the DC servo and the values we
write to it.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Make dev_() prints much prettier.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This allows more flexible integration with subsystem features.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked.  This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link.  It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.

Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.

When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.

Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This reverts commit 6f399115.

Since core has now support for limiting the volume on controls this
patch is not needed.  Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)

If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:

snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);

This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35

The S3C DMA API doesn't make use of hw_addr.to/from and also
the FIFO addresses are provided from the I2S drivers. So these
fields are redundant.
This patch removes the hw_addr.to/from fields for I2S and the
inclusion of header, paving way for the header to be moved closer
to the I2S controller drivers.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc