This is in preparation for the removal of struct snd_soc_device.

The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card.  Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Currently ASoC card initialisation is completed by a function called
snd_soc_register_card().  As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There is no argument named @clk_id in snd_soc_dai_set_fmt,
remove its' comment.
Signed-off-by: NQinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.

This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.

Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it.  It was much more useful
when ASoC was out of tree.
Signed-off-by: NMark brown <broonie@opensource.wolfsonmicro.com>

[stripped sound/isa/* changes, replaced with the next patch -- tiwai]
Signed-off-by: NKay Sievers <kay.sievers@vrfy.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

Since we can query the playback stream power state directly we do not
need to infer if it is powered up from the timer being scheduled.  Doing
this avoids problems that previously existed with streams being
incorrectly determined to be powered up caused when the timer is
scheduled when streams are closed after being partially set up.
Reported-by: NNobin Mathew <nobin.mathew@gmail.com>
Reported-by: NJukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Update the contact information for Liam Girdwood in ASoC core and
drivers as my old email address is no longer valid.
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Muting the DAC masks artefacts introduced as the digital stream shuts
down, for example when the input stops being clocked.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

ASOC: convert use of uint to unsigned int
Signed-off-by: NJon Smirl <jonsmirl@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Most of the ASoC controls refer to the maximum value that can be set for
a control as mask but there is no actual requirement for all bits to be
set at the highest possible value making the name mask misleading.
Change the code to use max instead.
Signed-off-by: NJon Smirl <jonsmirl@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Convert bitfields in ASoC into full int width. This is a
simple mechanical conversion. Two places in the DAPM code
were fixed to properly use mask.
Signed-off-by: NJon Smirl <jonsmirl@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Some codecs have unusual features in their register maps such as very
large registers representing arrays of coefficients. Support these
codecs in the register cache sysfs file by allowing them to provide a
function register_display() overriding the default output for register
contents.

Also ensure that we don't overflow PAGE_SIZE while writing out the
register dump.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.

This merger removes duplication in both DAI structures and simplifies
the API for other users.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Fix sound/soc build failure when CONFIG_PM=n:

linux-next-20080617/sound/soc/soc-core.c:829: error: 'soc_resume_deferred' undeclared (first use in this function)
soc3.out:make[3]: *** [sound/soc/soc-core.o] Error 1
Signed-off-by: NRandy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

On OpenMoko soc-audio resume is taking 700ms of the whole resume time of
1.3s, dominated by writes to the codec over I2C.  This patch shunts the
resume guts into a workqueue which then is done asynchronously.

The "card" is locked using the ALSA power state APIs as suggested by
Mark Brown.

[Added fix for race with resume to suspend and fixed a couple of nits
from checkpatch -- broonie.]
Signed-off-by: NAndy Green <andy@openmoko.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.
Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.

Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.

Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

The overwhelming majority just say 'initial version' anyway.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Since 43cc71ee, the platform modalias is
prefixed with "platform:".  Add MODULE_ALIAS() to the hotpluggable sound
platform drivers, to re-enable auto loading.

[dbrownell@users.sourceforge.net: more drivers, registration fixes]
Signed-off-by: NKay Sievers <kay.sievers@vrfy.org>
Signed-off-by: NDavid Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Don't take the codec mutex during machine probe until we have registered
with ALSA, fixing a lockdep warning reported by Dmitry Baryshkov.
Cc: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Add TLV support to ASoC.
Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

This header file exists only for some hacks to adapt alsa-driver
tree.  It's useless for building in the kernel.  Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it.  This should be really killed in
future.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Fix the various misspellings of "system", controller", "interrupt" and
"[un]necessary".
Signed-off-by: NRobert P. J. Day <rpjday@mindspring.com>
Signed-off-by: NAdrian Bunk <bunk@kernel.org>

Clean up codes using the new common snd_ctl_boolean_*_info() callbacks.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>