Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.
Signed-off-by: NOla Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds a function getting the stream-name as a string for
a specific stream.
Signed-off-by: NOla Lilja <ola.o.lilja@stericsson.com>
Reviewed-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

SupherH FSI2 can use special data transfer,
but it depends on CPU-FSI2 connection style.

We can use 16bit data stream mode if it was valid connection,
and it is required for 16bit data DMA transfer / SPDIF sound output.
We can use 24bit data transfer if it was invalid connection.

We can select connection type if CPU is SH7372,
and it is always valid connection if latest SuperH.

This patch adds new bus_option and fsi_bus_setup()
for supporting these feature.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There's no space for the sign bit.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.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>

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>

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>

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>

Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
Signed-off-by: NRicardo Neri <ricardo.neri@ti.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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>

This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.

This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Also remove two warnings when CONFIG_SND_DEBUG is not set:

sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Supports larger register maps, not using unsigned ints for the full 32
bit as we rely on checking for negative registers.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle.

Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros.
Add single register macro : SOC_SINGLE_SX_TLV.
Use snd_soc_info_volsw for .info
Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double.

kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros.

The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet.
Signed-off-by: NBrian Austin <brian.austin@cirrus.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There are no users any more and new drivers should be using supply widgets
which fully replace it anyway.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NZeng Zhaoming <zengzm.kernel@gmail.com>

This change adds the logic to support using the jack detect mechanism built
in to the codec to detect both when a jack was inserted and what type of
jack is present.

This change also supports the use of an external mechanism for headphone
detection. If this mechanism exists, when the max98095_jack_detect function
is called, the hp_jack is simply passed NULL.

This change supports both simple headphones, powered headphones, microphones
and headsets with both headphones and a mic.
Signed-off-by: NRhyland Klein <rklein@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In version 3.4 the driver core acquired probe deferral which is a core way
of doing essentially the same thing as ASoC has been doing since forever
to make sure that all the devices needed to make up the card are present
without needing open coding in the subsystem.

Make basic use of this probe deferral mechanism for the cards, removing the
need to handle partially instantiated cards. We should be able to remove
even more code than this, though some of the checks we're currently doing
should stay since they're about things like suppressing unneeded DAPM runs
rather than deferring probes.

In order to avoid robustness issues with our teardown paths (which do need
quite a bit of TLC) add a check for aux_devs prior to attempting to set
things up, this means that we've got a reasonable idea that everything will
be there before we start. As with the removal of partial instantiation
support more work will be needed to make this work neatly.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Currently operations on jack reporting take the CODEC mutex both to protect
the current jack status and also to protect the DAPM run which is triggered
on status updates. Since the addition of a DAPM-specific lock we no longer
need to worry about locking DAPM as it has its own finer grained lock so
create a per jack lock to take care of the jack status.

This is both cleaner where the jack isn't specifically associated with a
CODEC and clearer as it's much more obvious what the lock is protecting.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Currently DAPM widgets use the private data for their regulator.
Add a regulator * for widgets to use instead of private data.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Rename SND_SOC_DAPM_CLASS_PCM to SND_SOC_DAPM_CLASS_RUNTIME to
better match the usage and align with card mutex too.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better
describe all uses for this mutex subclass and align with DAPM too.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Currently stream events are only perfomed on codec stream widgets only.
There is now a need to be able to perform stream events on platform
widgets too.

e.g. we have the ABE platform driver with several DAI links
to dummy codecs. We need to be able to perform stream events on any
of the dummy codec DAI links.

This patch also removes the snd_soc_dai * parameter since it's already
contained within the rtd * parameter.

Finally makle stream event return void since no one checks it anyway.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add platform driver support for CPU DAI DAPM widgets.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

It has now become necessary to use a DAPM mutex instead of the codec
mutex to lock the DAPM operations. This is due to the recent multi
component support and forth coming Dynamic PCM updates.

Currently we lock DAPM operations with the codec mutex of the calling
RTD context. However, DAPM operations can span the whole card context
and all components.

This patch updates the DAPM operations that use the codec mutex to
now use the DAPM mutex PCM subclass for all DAPM ops.

We also add a mutex subclass for DAPM init and PCM operations.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This is the first part of a change that is intended to improve
ASoC locking protection for DAPM and PCM operations.

This part of the series adds a mutex class for the soc_card mutex. The
SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the
SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic
PCM operations. The new mutex classes are required otherwise we will see a false
positive mutex deadlock warning between the card initialisation and the PCM
operations (something that would never deadlock in real life).
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This addresses some header check warnings.  DRM headers which include
"drm.h" have been excluded, as they indirectly include types.h.
Signed-off-by: NBobby Powers <bobbypowers@gmail.com>
Cc: Chris Ball <cjb@laptop.org>
Cc: Dave Airlie <airlied@linux.ie>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

The tea575x-tuner module has been updated to use the latest V4L2 framework
functionality. This also required changes in the drivers that rely on it.

The tea575x changes are:

- The drivers must provide a v4l2_device struct to the tea module.
- The radio_nr module parameter must be part of the actual radio driver,
  and not of the tea module.
- Changed the frequency range to the normal 76-108 MHz range instead of
  50-150.
- Add hardware frequency seek support.
- Fix broken rxsubchans/audmode handling.
- The application can now select between stereo and mono.
- Support polling for control events.
- Add V4L2 priority handling.

And radio-sf16fmr2.c now uses the isa bus kernel framework.
Signed-off-by: NHans Verkuil <hans.verkuil@cisco.com>
Thanks-to: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: NMauro Carvalho Chehab <mchehab@redhat.com>

The <linux/device.h> header includes a lot of stuff, and
it in turn gets a lot of use just for the basic "struct device"
which appears so often.

Clean up the users as follows:

1) For those headers only needing "struct device" as a pointer
in fcn args, replace the include with exactly that.

2) For headers not really using anything from device.h, simply
delete the include altogether.

3) For headers relying on getting device.h implicitly before
being included themselves, now explicitly include device.h

4) For files in which doing #1 or #2 uncovers an implicit
dependency on some other header, fix by explicitly adding
the required header(s).

Any C files that were implicitly relying on device.h to be
present have already been dealt with in advance.

Total removals from #1 and #2: 51.  Total additions coming
from #3: 9.  Total other implicit dependencies from #4: 7.

As of 3.3-rc1, there were 110, so a net removal of 42 gives
about a 38% reduction in device.h presence in include/*
Signed-off-by: NPaul Gortmaker <paul.gortmaker@windriver.com>

Allows the constraint lists to be declared const by drivers which seems
reasonable; there's plenty of other constification we could do if we were
being complete but this was easy and quick.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

We need to resume two legacy registers to recover MIDI/FM functionality
on S3/S4 resume, too.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a hook to vmaster control to be called at each time
when the master value is changed.  It'd be handy for an additional
mute LED control following the Master switch, for example.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add mutex support for platform IO operations. e.g. can be used
for platform DAPM widget IO ops.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There's now core code which falls back to global CODEC operations for
DAI calls that needs to be able to tell if it's dealing with a CPU or
CODEC DAI and given the small number of DAIs in a typical system and
overall memory usage pattern saving a pointer per DAI is really not
worth the effort.
Reported-by: NIan Lartey <ian@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>