McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.

Sidetone feature can be tested with following commands:

	(set up codec input and output paths)
	# Enable and configure sidetone
	amixer -D hw:0 set 'McBSP2 Sidetone' on
	amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
	echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
	# Do not loop audio via CPU
	arecord -f dat >/dev/null |aplay /dev/zero
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out
signals of TLV320AIC34 than TPA6130 headphone amplifier.

This patch adds route to transmitter and "FM Transmitter" control to keep
route active when needed.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
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>

It is more usefull to report headset instead of video out cable in response
to jack insertion as this is more usual use-case and because now the headset
feature is supported. Automatic accessory detection is not possible at the
moment so most sensible static accessory type have to be used.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds support for headset microphone in Nokia RX-51/N900. The mic
signal from audio jack is routed to codec A LINE1L via two switches and the
mic bias is coming from codec B part.

First switch is the tv-out switch that is already supported and the second
switch selects between voltage detection circuit and codecs. As there is
no use for voltage detection at the moment the second switch is connected
statically to codecs in rx51_soc_init.

Headset can be active when control "Jack Function" is set to "Headset".
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Just slight cleanup to be sync with upcoming change.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The Amstrad Delta ASoC driver used to override the digital_mute()
callback, expected to be not provided by the on-board CX20442 CODEC
driver, with its own implementation. While this is still posssible when
substituting the whole empty snd_soc_dai_driver.ops member (the CX20442
case), replacing snd_soc_dai_ops.digital_mute only is no longer correct
after the snd_soc_dai_driver.ops member has been constified, and results
in build error.

Drop this actually not used code path in hope the CX20442 driver never
provides its own snd_soc_dai_ops structure.

Created and tested against linux-2.6.38-rc2
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Earphone in Nokia RX-51/N900 is connected to left HP output of B part of the
TLV320AIC34 dual codec. In RX-51 the codec A is used as a traditional codec
and the codec B as an auxiliary device.

Audio from codec A goes via the codec B to earphone:
MONO_LOUT of A -> LINE2R of B (B interconnects) -> HPLOUT of B -> Earphone.

Take earphone into use by utilizing the recent ASoC auxiliary and
cross-device support.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier.
This patch adds support for it and stereo output can be active when
"Jack Function" == "TV-OUT" || "Headphone".

As the TPA6130 can output very high volume levels the output is limited
with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds McBSP support for the OMAP4 CPU
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: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Use jack framework to enable detection for the headset microphone
and stereo output in the sdp4430.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NDavid Anders <x0132446@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 FM stereo pins to the machine driver and add them as a
dapm widget.
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>

Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.

Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.

Thanks to Tony Lindgren <tony@atomide.com> for noticing.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Tony Lindgren <tony@atomide.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 current implementation, there are resources leak in the error path.
This patch properly reclaims the allocated resources in the error path.

Also adds a missing clk_put in osk_soc_exit.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Return PTR_ERR(omap3pandora_dac_reg) instead of 0 if regulator_get failed.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In the new code introduced with commit cf4c87ab,
"OMAP: McBSP: implement McBSP CLKR and FSR signal muxing via mach-omap2/mcbsp.c",
the way omap1 build is supposed to bypass omap2 specific functionality doesn't
optimize out all omap2 specific stuff. This breaks linking phase for omap1
machines, giving "undefined reference to `omap2_mcbsp1_mux_clkr_src'"
and "undefined reference to `omap2_mcbsp1_mux_fsr_src'" errors. Fix it.

Created and tested against linux-2.6.37-rc1.
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NPaul Walmsley <paul@pwsan.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Commit 346a5c890 (OMAP: control: move plat-omap/control.h
to mach-omap2/control.h) in the linux-omap tree removed
plat/control.h and most of its callers. This one slipped
through - breaking the build as below when
CONFIG_SND_OMAP_SOC_MCPDM is defined. Fix this.

  CC      sound/soc/omap/omap-mcpdm.o
sound/soc/omap/omap-mcpdm.c:35: fatal error: plat/control.h: No such file or directory
compilation terminated.
make[3]: *** [sound/soc/omap/omap-mcpdm.o] Error 1
make[2]: *** [sound/soc/omap] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: NAnand Gadiyar <gadiyar@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Paul Walmsley <paul@pwsan.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NPaul Walmsley <paul@pwsan.com>
Signed-off-by: NTony Lindgren <tony@atomide.com>

It is currently completely normal to execute these machine drivers code on
different boards if the kernel includes support for multiple boards so no
error message should be printed if the machine_is_xxx does not match with
the machine driver.

Therefore remove these pr_err and pr_debug prints in those cases.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Previously the OMAP McBSP ASoC driver implemented CLKS switching by
using omap_ctrl_{read,write}l() directly.  This is against policy; the OMAP
System Control Module functions are not intended to be exported to drivers.
These symbols are no longer exported, so as a result, the OMAP McBSP ASoC
driver does not build as a module.

Resolve the CLKS clock changing portion of this problem by creating a
clock parent changing function that lives in
arch/arm/mach-omap2/mcbsp.c, and modify the ASoC driver to use it.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, this symbol must be exported for use by
sound/soc/omap/omap-mcbsp.c.

Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/* and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are.  These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversions.
Signed-off-by: NPaul Walmsley <paul@pwsan.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The OMAP ASoC McBSP code implemented CLKR and FSR signal muxing via
direct System Control Module writes on OMAP2+.  This required the
omap_ctrl_{read,write}l() functions to be exported, which is against
policy: the only code that should call those functions directly is
OMAP core code, not device drivers.  omap_ctrl_{read,write}*() are no
longer exported, so the driver no longer builds as a module.

Fix the pinmuxing part of the problem by removing calls to
omap_ctrl_{read,write}l() from the OMAP ASoC McBSP code and
implementing signal muxing functions in arch/arm/mach-omap2/mcbsp.c.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, these symbols must be exported for use by
sound/soc/omap/omap-mcbsp.c.

Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/*, and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are.  These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversion.
Signed-off-by: NPaul Walmsley <paul@pwsan.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch fixes sparse warning due non declaration of static function

sound/soc/omap/omap-mcbsp.c:783:5: warning: symbol 'omap_mcbsp_st_info_volsw' was not declared. Should it be static?
Signed-off-by: NG, Manjunath Kondaiah <manjugk@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Tony Lindgren <tony@atomide.com>
Cc: Nishanth Menon <nm@ti.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Speaker amplifier is controlled by TWL4030 GPIO which may sleep. Therefore
use gpio_set_value_cansleep to get rid of runtime warning that is introduced
after the commit 9c4ba946 and to get a stack trace if ever executing this
code in atomic context.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.

Use a better name, "cx20442-voice", as suggested by Liam Girdwood.
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: NTimur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: NJassi Brar <jassi.brar@samsung.com>
Signed-off-by: NSeungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: NTimur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.

Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.

The patch extends the original THRESHOLD mode by doing the
following:

if (period_words <= max_threshold)
Current THRESHOLD mode configuration

Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size

With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only
SND_JACK_VIDEOOUT type is reported. More types could be reported after
getting more audio features supported and necessary drivers integrated for
implementing automated accessory detection.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used
as headphone, headset or audio-video connector. This patch implements the
control 'Jack Function' which is used to select the desired function.
At the moment only TV-out without audio is supported.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.

Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer

This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)

Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Fix build warning about unused ops and add ops
to the sdp4430 DAI link.
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>

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>

This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.htmlSigned-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NEduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>