Add Voice DAI to support the PCM voice interface of the twl4030 codec.

The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.

The PCM voice interface has two modes
 - PCM mode1 : This uses the normal FS polarity and the rising edge of
               the clock signal.
 - PCM mode2 : This uses the FS polarity inverted and the falling edge
               of the clock signal.

If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.

But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...

The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false

This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.

The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a.

Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with  FWID(0).
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Fix accidental change of <mach/regs-gpio.h> to
<plat/regs-gpio.h> in s3c2412-i2s.c
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Fix the build error in s3c-i2s-v2.c caused by
a change to the snd_soc_dai ops field.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The definition of s3c_i2sv2_iis_calc_rate was never
renamed from s3c2412_iis_calc_rate, so rename this
to allow the build to work.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Fix build errors in sound/soc/s3c24xx/jive_wm8750.c
from changes to ASoC.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

pxa_ssp_set_dai_fmt() currently has an early exit if the desired format
equals the current configuration. This is correct behaviour unless this
function is called with a zero value parameter for the first time.
Zero is a valid value for this function, but the early exit is bogus in
this case.

Hence, set priv->dai_fmt to -1 in the beginning so we can configure the
port.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

It has a shared LRCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Also make sure we're checking for the right operation while we're here.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.

Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.

The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.

Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.

The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.

This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)

SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).

This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless.  Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Eric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

* 'core/softlockup' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
  softlockup: make DETECT_HUNG_TASK default depend on DETECT_SOFTLOCKUP
  softlockup: move 'one' to the softlockup section in sysctl.c
  softlockup: ensure the task has been switched out once
  softlockup: remove timestamp checking from hung_task
  softlockup: convert read_lock in hung_task to rcu_read_lock
  softlockup: check all tasks in hung_task
  softlockup: remove unused definition for spawn_softlockup_task
  softlockup: fix potential race in hung_task when resetting timeout
  softlockup: fix to allow compiling with !DETECT_HUNG_TASK
  softlockup: decouple hung tasks check from softlockup detection

Merge branch 'tracing-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip

* 'tracing-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
  branch tracer, intel-iommu: fix build with CONFIG_BRANCH_TRACER=y
  branch tracer: Fix for enabling branch profiling makes sparse unusable
  ftrace: Correct a text align for event format output
  Update /debug/tracing/README
  tracing/ftrace: alloc the started cpumask for the trace file
  tracing, x86: remove duplicated #include
  ftrace: Add check of sched_stopped for probe_sched_wakeup
  function-graph: add proper initialization for init task
  tracing/ftrace: fix missing include string.h
  tracing: fix incorrect return type of ns2usecs()
  tracing: remove CALLER_ADDR2 from wakeup tracer
  blktrace: fix pdu_len when tracing packet command requests
  blktrace: small cleanup in blk_msg_write()
  blktrace: NUL-terminate user space messages
  tracing: move scripts/trace/power.pl to scripts/tracing/power.pl

* 'irq/threaded' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
  genirq: fix devres.o build for GENERIC_HARDIRQS=n
  genirq: provide old request_irq() for CONFIG_GENERIC_HARDIRQ=n
  genirq: threaded irq handlers review fixups
  genirq: add support for threaded interrupts to devres
  genirq: add threaded interrupt handler support

Commit c2ec175c ("mm: page_mkwrite
change prototype to match fault") exposed a bug in the NFS
implementation of page_mkwrite.  We should be returning 0 on success...
Signed-off-by: Trond Myklebust <Trond.Myklebust@netapp.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci-2.6:
  PCI: pci_slot: grab refcount on slot's bus
  PCI Hotplug: acpiphp: grab refcount on p2p subordinate bus
  PCI: allow PCI core hotplug to remove PCI root bus
  PCI: Fix oops in pci_vpd_truncate
  PCI: don't corrupt enable_cnt when doing manual resource alignment
  PCI: annotate pci_rescan_bus as __ref, not __devinit
  PCI-IOV: fix missing kernel-doc
  PCI: Setup disabled bridges even if buses are added
  PCI: SR-IOV quirk for Intel 82576 NIC

* 'for-linus' of git://git.kernel.dk/linux-2.6-block:
  loop: mutex already unlocked in loop_clr_fd()
  cfq-iosched: don't let idling interfere with plugging
  block: remove unused REQ_UNPLUG
  cfq-iosched: kill two unused cfqq flags
  cfq-iosched: change dispatch logic to deal with single requests at the time
  mflash: initial support
  cciss: change to discover first memory BAR
  cciss: kernel scan thread for MSA2012
  cciss: fix residual count for block pc requests
  block: fix inconsistency in I/O stat accounting code
  block: elevator quiescing helpers

The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.

The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.

The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.

A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

The code that enables branch tracing for all (non-constant) branches
plays games with the preprocessor and #define's the C 'if ()' construct
to do tracing.

That's all fine, but it fails for some unusual but valid C code that is
sometimes used in macros, notably by the intel-iommu code:

	if (i=drhd->iommu, drhd->ignored) ..

because now the preprocessor complains about multiple arguments to the
'if' macro.

So make the macro expansion of this particularly horrid trick use
varargs, and handle the case of comma-expressions in if-statements.  Use
another macro to do it cleanly in just one place.

This replaces a patch by David (and acked by Steven) that did this all
inside that one already-too-horrid macro.
Tested-by: Ingo Molnar <mingo@elte.hu>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Steven Rostedt <rostedt@goodmis.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>