ASoC: da7210: Add HeadPhone Playback Volume control

HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.

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

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

ASoC: TWL4030: Add configurable delay after digimic enable

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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode

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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: davinci: let platform data define edma queue numbers

Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.

This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.

platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.

Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.

Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.

This patch has been tested on DM644x and OMAP-L138 EVMs.

Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC:Support Samsung SoC(S5P) in I2Sv2

This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' into for-2.6.36

ASoC: Select wm_hubs automatically for WM8994

Otherwise all machine drivers need to do so.

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Implement WM8994 AIF1ADC2 paths

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

ASoC: Explicitly disable DC servo on WM hubs headphone powerdown

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

ASoC: eukrea-tlv320: add support for cpuimx35sd

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: imx: check kzalloc() result and fix memory leak

If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: imx: check kzalloc() result and fix memory leak

If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: davinci: check kzalloc() result (typo)

The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: fsi: Add specified ID for soc-audio

Specified ID is necessary, when some codecs are used with FSI.

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

Merge branch 'for-2.6.35' into for-2.6.36

ASoC: Remove duplicate AUX definition from WM8776

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org

ASoC: TWL6040: Correct widget handling for drivers

In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.

Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: Handle read failures in codec_reg

When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.

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

Merge branch 'for-2.6.35' into for-2.6.36

ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register

snd_soc_unregister_codec is called twice if snd_soc_register_dai fail.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: wm8727: add a missing return in wm8727_platform_probe

otherwise the error path will always be executed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: au1x: fix section mismatch in psc-i2s.c

Annotate platform probe callback with __devinit instead of plain __init.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: kirkwood-i2s: Handle mute/unmute playback/record

The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Fix kirkwood i2s mono playback

Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: fsi: Fixup for master mode

This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.

This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.

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

ASoC: fsi: Add pr_err for noticing unsupported access

This patch didn't use dev_err,
because it is difficult to get struct device here.

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

ASoC: fsi: Change struct fsi_regs to fsi_core

Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.

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

ASoC: fsi: remove noisy CR_FMT macro

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

ASoC: fsi: remove un-used variable on fsi_dai_startup

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

ASoC: fsi: modify format area definition on flags

There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence number

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

ASoC: Remove unnecessary casts of private_data

Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Remove unnecessary casts of private_data

Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' into for-2.6.36

ASoC: fsi: fixup wrong value setting order of TDM

channel size should be set before setting register value

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

ASoC: fsi: fixup clock inversion operation

Clock inversion should be specified by each flags bit.

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

ASoC: TWL4030: DAC power optimization

Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: TWL4030: Fix for digital loopback gain range

When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: Implement WM8994 OPCLK support

The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK.  The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.

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

ASoC: Include WM8994 GPIO and interrupt registers in codec_reg

Very handy for debug.

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

ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)

    i2s_accurate_sck switch can be used to have a better approximate
    sampling frequency.
    The clock is an externally visible bit clock and it is named
    i2s continuous serial clock (I2S_SCK).
    The trade off is between more accurate clock (fast clock)
    and less accurate clock (slow clock).
    The waveform will be not symmetric.
    Probably it is possible to get a better algorithm for calculating
    the divider, trying to keep a slower clock as possible.

    This patch has been developed against the
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: DaVinci: Added selection of clk input pin for McBSP

    When McBSP peripheral gets the clock from an external pin,
    there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
    and MCBSP_CLKS.
    evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
    hardware connection and I use MCBSP_CLKS, so I have added
    this possibility.

    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm)

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)

    Added two clocking options for dm365 McBSP peripheral when used
    with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
    clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
    from external pin and generates frame sync).
    A slave clock management can be important when the external codec needs
    the system clock and the bit clock synchronized (tested with uda1345).
    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Invert speaker enabling behaviour in SmartQ sound driver

The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Automatically manage ALC coefficients for WM8960

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

ASoC: Automatically manage DAC deemphasis rate for WM8960

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

ASoC: Remove current WM8960 deemphasis control

It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.

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

ASoC: Fix sorting of Makefile and Kconfig

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

ASoC: Add SmartQ sound driver

This adds sound support for the SmartQ board.

The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: codec: Add WM8987 device id to WM8750 driver

The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: ak4642: Add Digital Playback Volume control

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: uda134x: correct bias level setup for codecs family

For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: uda134x: add DATA011 register found in codecs family

In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge remote branch 'takashi/topic/asoc' into for-2.6.36

ASoC: clean i.MX Kconfig

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: uda134x: fix bias level setup on initialization

On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: uda134x: replace a macro with a value in platform struct.

This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

ASoC: RX-51: Add basic jack detection

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: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: RX-51: Add Jack Function kcontrol

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: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

codecs/tlv320aic23: fix bias management for suspend/resume

in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.

in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF

Signed-off-by: Eric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: JZ4740: Add qi_lb60 board driver

This patch adds ASoC support for the qi_lb60 board.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Add JZ4740 codec driver

This patch adds support for the JZ4740 internal codec.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Add JZ4740 ASoC support

This patch adds ASoC support for JZ4740 SoCs I2S module.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: pandora: fix CLKX polarity

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: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: Fix sorting of DA7210 entries in Kconfig

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Fix overflow bug in SOC_DOUBLE_R_SX_TLV

When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.

The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.

Solution: Mask the result so that it "wraps around", emulating
sign-extension.

Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: eukrea-tlv320: add support for our i.MX25 board

* tdm slot has to be configured to get sound working on i.MX25

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210

The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.

Consequently when I2C is not set, the compilation fails [1]

This patch fixes this issues, by adding a depencdency on the related HW-
controller.

Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Pay attention to write errors in volsw_2r_sx

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

ASoC: Default WM2000 ANC and speaker to enabled

The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.

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

Merge branch 'for-2.6.35' into for-2.6.36

ASoC: DaVinci: Fix McASP hardware FIFO configuration

On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at

http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf

Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)

During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.

https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).

The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.

Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: header cleanup for da7210

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

ASoC: header cleanup for ak4642

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

ASoC: header cleanup for FSI-DA7210

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

ASoC: header cleanup for FSI-AK4642

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

ASoC: header cleanup for FSI

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

ASoC: Remove unused header from MPC5200 PSC driver

The header contains an extern that isn't used by anything.  Remove.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: NUC900: patch for fix build error

This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: EP93xx: Add Snapper CL15 i2s audio support

Add support for i2s audio on Bluewater Systems Snapper CL15 module

Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'ep93xx' into for-2.6.36

Trivial add/add fixup required in the clock table.

Conflicts:
arch/arm/mach-ep93xx/clock.c

EP93xx: Add i2s core support

Add core support for EP93xx i2s audio

Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.36' of git://git./linux/kernel/git/broonie/sound-2.6 into topic/asoc

ASoC: patch for the useless 'break' removal in kirkwood

This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: atmel: trivial code cleanup

Remove break after return, it is not needed.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: ep93xx i2s audio driver

Add ep93xx i2s audio driver

Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: tlv320dac33: Add support for changing upper threshold

Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: imx: add eukrea-tlv320

Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: imx-ssi.c: add new choices to platform configuration

* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
IMX_SSI_NET : enable Network Mode
IMX_SSI_SYN : enable Synchronous Mode
IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: s3c: patch for the unnecessary variable 'state' removal

The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: atmel: patch for the unnecessary variable removal

The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: use resource_size for au1x

Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: omap-mcbsp: Place correct constraints for streams

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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: omap-mcbsp: Save, and use wlen for threshold configuration

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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

OMAP3: McBSP: Use the port's buffer_size when calculating tx delay

Sicne the platform data's buffer_size now holds the full size
of the FIFO, there is no longer need to handle the ports
differently.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

OMAP3: McBSP: Change the way how the FIFO is handled

Use the actual FIFO size in words as buffer_size on OMAP3.
Change the threshold configuration to use 1 based numbering, when
specifying the allowed threshold maximum or the McBSP threshold value.
Set the default maximum threshold to (buffer_size - 0x10) intialy.
>From users of McBSP, now it is expected to use this method.
Asking for threshold 1 means that the value written to threshold registers
are going to be 0, which means 1 word threshold.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

OMAP: McBSP: Function to query the FIFO size

Users of McBSP can use the omap_mcbsp_get_fifo_size function to query
the size of the McBSP FIFO.
The function will return the FIFO size in words (the HW maximum).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: s6000: use resource_size for {request/release}_mem_region and ioremap

The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' of git://git./linux/kernel/git/broonie/sound-2.6 into fix/asoc