Reduce our stack consumption by moving the params off the stack, they
are reasonably large and might be an issue on platforms with small stacks.
Reported-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Ackeded-by: NLiam Girdwood <lrg@ti.com>

This can be helpful to users when tuning their systems.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.

This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.

A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.

This patchs adds/changes the following :-

 o Adds DPCM runtime update core.
 o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
   to return if a change has occured rather than 0. No other users check
   atm.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add debugFS files for DPCM link management information.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.

Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.

e.g. pcm:0,0 routing digital data to 2 external codecs.

FE pcm:0,0  ----> BE (McBSP.0) ----> CODEC 0
             +--> BE (McPDM.0) ----> CODEC 1

e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.

FE pcm:0,0 ---
             +--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---

The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.

DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.

Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.

This patch adds the core DPCM code and contains :-

 o The FE and BE PCM operations.
 o FE and BE DAI link support.
 o FE and BE PCM creation.
 o BE support API.
 o BE and FE link management.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

commit 4183eed2 (ASoC: core: Add signed multi register control) introduced
the variable 'min',but it is not used.

Remove it to fix the following build warning:

sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx':
sound/soc/soc-core.c:2990: warning: unused variable 'min'
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

We have never really updated that version number and probably never will, so
just remove it.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The sysclock is fixed, so just set it up once in the init callback instead of
setting it repeatably in the hw_params callback.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.
Signed-off-by: NKyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).

This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.

Added convenience macro.

SOC_SINGLE_STROBE

Added accessor implementations.

snd_soc_get_strobe
snd_soc_put_strobe
Signed-off-by: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.

Added convenience macro.

SOC_SINGLE_XR_SX

Added accessor implementations.

snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx
Signed-off-by: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Fixes the following build warning on x86_64.

In file included from include/trace/ftrace.h:567:0,
                 from include/trace/define_trace.h:86,
                 from include/trace/events/asoc.h:410,
                 from sound/soc/soc-core.c:45:
include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_output_path':
include/trace/events/asoc.h:246:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_input_path':
include/trace/events/asoc.h:275:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Springbank can support stereo, though it is primarily intended for mono
use cases.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In preparation for ASoC DSP support.

Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.

This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: NAshish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.

This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,

(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled

This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.

Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: NAshish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.

Conflicts:
	sound/soc/soc-core.c
	sound/soc/tegra/tegra_i2s.c
	sound/soc/tegra/tegra_spdif.c

Fix the following build warning:

sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'

'%llx' should be used with 'u64' type.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct.  If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.

This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.

This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there.  It is expected that the bias
level callbacks will be used for clock configuration.

Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute().  This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here.  We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.

At present we are also restricted to a single DAPM link for the entire
DAI.  Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.

This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Pull ARM fixes from Russell King:
 "Nothing too disasterous, the biggest thing being the removal of the
  regulator support for vcore in the AMBA driver; only one SoC was using
  this and it got broken during the last merge window, which then
  started causing problems for other people.  Mutual agreement was
  reached for it to be removed."

* 'fixes' of git://git.linaro.org/people/rmk/linux-arm:
  ARM: 7386/1: jump_label: fixup for rename to static_key
  ARM: 7384/1: ThumbEE: Disable userspace TEEHBR access for !CONFIG_ARM_THUMBEE
  ARM: 7382/1: mm: truncate memory banks to fit in 4GB space for classic MMU
  ARM: 7359/2: smp_twd: Only wait for reprogramming on active cpus
  ARM: 7383/1: nommu: populate vectors page from paging_init
  ARM: 7381/1: nommu: fix typo in mm/Kconfig
  ARM: 7380/1: DT: do not add a zero-sized memory property
  ARM: 7379/1: DT: fix atags_to_fdt() second call site
  ARM: 7366/3: amba: Remove AMBA level regulator support
  ARM: 7377/1: vic: re-read status register before dispatching each IRQ handler
  ARM: 7368/1: fault.c: correct how the tsk->[maj|min]_flt gets incremented

The 'max' range needs to be unsigned, since the size of the user address
space is bigger than 2GB.

We know that 'count' is positive in 'long' (that is checked in the
caller), so we will truncate 'max' down to something that fits in a
signed long, but before we actually do that, that comparison needs to be
done in unsigned.

Bug introduced in commit 92ae03f2 ("x86: merge 32/64-bit versions of
'strncpy_from_user()' and speed it up").  On x86-64 you can't trigger
this, since the user address space is much smaller than 63 bits, and on
x86-32 it works in practice, since you would seldom hit the strncpy
limits anyway.

I had actually tested the corner-cases, I had only tested them on
x86-64.  Besides, I had only worried about the case of a pointer *close*
to the end of the address space, rather than really far away from it ;)

This also changes the "we hit the user-specified maximum" to return
'res', for the trivial reason that gcc seems to generate better code
that way.  'res' and 'count' are the same in that case, so it really
doesn't matter which one we return.
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>