Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This binding doesn't include the nvidia,model or nvidia,audio-routing
properties the other Tegra audio DT bindings have, because this binding
is targetted at a single machine, rather than for any machine using the
tlv320aic23 codec.
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Makes the code more standard and prepares for better framework usage.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In the conversion to module_init_i2c() the original open coded module
exit function was left.  Remove it.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: NBrian Austin <brian.austin@cirrus.com>
Signed-off-by: NGeorgi Vlaev <joe@nucleusys.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

We should check dailess before dereferencing.
Reported-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

It tries to clk_get the clock. And if it failed, it assumes the clock
by default enabled.
Signed-off-by: NRichard Zhao <richard.zhao@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NRichard Zhao <richard.zhao@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.

In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.

Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove writable debugFS permission, use simple_open() and
fix indentation.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.
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>

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>