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

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

Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

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

Add support for adding "status = disabled" to an SSI node to incidate that it
is not wired on the board.  This replaces the not-so-intuitive previous method
of omitting a codec-handle property.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Acked-by: NKumar Gala <galak@kernel.crashing.org>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The error handling code in the OF probe function of the SSI driver is not
freeing all resources correctly.

Since the machine driver no longer calls the DMA driver to provide information
about the SSI, we don't need to keep a list of DMA objects any more.  In
addition, the fsl_soc_dma_remove() function is incorrectly removing *all*
DMA objects when it should only remove one.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.

The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture.  This function is just a front-end for
dma_alloc_coherent().  Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs.  Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NKumar Gala <galak@kernel.crashing.org>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch add sound support for the Goni board based on S5PV210.

The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.

The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
  : When TV-OUT cable is inserted on Goni board,
  the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
  : When TV-OUT cable is inserted on Goni board,
  the TV-OUT cable is connected to television.
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>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch add sound support for the Aquila board based on S5PC110.

The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.

The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
  : When TV-OUT cable is inserted on Aquila board,
  the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
  : When TV-OUT cable is inserted on Aquila board,
  the TV-OUT cable is connected to television.
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>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board

This patch modify the wrong field name on Aquila board.
Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

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>

Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
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 fixes the error path in wm9081_register to properly free resources.
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 a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Reviewed-by: NGuennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).
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>

This patch fixes the error path in wm8961_register to properly free resources.
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>

This patch fixes the error path in wm8955_register to properly free resources.
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>

This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.
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>

This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
   snd_soc_register_dai() fail to properly free resources.
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>

wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).
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>

This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
   handling if snd_soc_register_dai 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>

da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.
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>

ak4642 should be kfreed if ak4642_init() return error.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Reviewed-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NBarry Song <21cnbao@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz.  The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.

TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.
Signed-off-by: NIan Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

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>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current FSI driver id is not only 0
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
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>

Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
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>

There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.
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>

HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.

Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.

Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.

Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).
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>

AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>