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>

This patch fixes a bug whereby AC97 bus device data was being clobbered
when AC97 codecs using the generic ac97_codec.c driver were being
registered. Codecs that didn't use the generic driver were unaffected
(e.g. WM9712, WM9713).
Changes:-
 o Add new AC97 codec class for custom (or need bus dev registration)
AC97 codecs.
 o Only register/deregister this custom codec device with the AC97 bus.
The generic AC97 driver already does this for generic codec devices.
This may be related to bug #3038 :-
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch updates the ASoC core to the new DAI matching and clocking
API in version 0.13
Changes:-
 o Removed DAI capabilities matching code in favour of manual matching
in the machine drivers.
 o Added DAI operations for codec and CPU interfaces.
 o Removed config_sysclk() function and struct snd_soc_clock_info. No
longer needed as clocking is now configured manually in the machine
drivers. Also removed other clocking data from structures.
 o Added machine driver prepare callback.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This very minor patch fixes the snd_soc_new_pcms() function to comply
with the kernel coding style.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch fixes a bug whereby the power management delayed work would
never be run at driver suspend() or module remove(). Delayed work would
be created (after audio had finished) with a long delay (~5 secs) and
was sometimes never queued before flush_scheduled_work() was being
called at suspend or module remove. This caused the delayed work to
queued after the module had been removed or after resume.
This patch forces any delayed work to complete by cancelling it (timer
cannot fire and add it to queue later), scheduling it for now and
waiting on it's completion.
This is something I probably would like to add to workqueue.c in the
next merge window, however it's here atm because it can oops.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch fixes a bug whereby some resources were not being freed when
codec probe() failed.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

Use global workqueue for simplicity instead of own workqueue
in SoC core and wm8750 codes.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch adds copyright and credit for my good friend Richard Purdie
from OpenedHand for his help and code contribution throughout the
development of the core code. Many thanks Richard (I guess we overlooked
this in trying to get everything working well).
It also adds some extra comments wrt to DAI clock matching.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

From: Andrew Morton <akpm@osdl.org>
I converted the workqueues to per-device while I was there.  It seems
strange to create a new kernel thread (on each CPU!) and to then only
have a single global work to ever be queued upon it.
Plus without this, I'd have to use the _NAR stuff, gawd help me.
Does that workqueue really need to be per-cpu?
Does that workqueue really need to exist?  Why not use keventd?
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: NAndrew Morton <akpm@osdl.org>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch by Philipp Zabel fixes a bug whereby the BCLK matching fails
when the Codec BCLK is constant and the CPU BCLK is based upon a
divider.
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@suse.cz>

This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
 1. BCLK = Rate * x
 2. BCLK = MCLK / x
 3. BCLK = Rate * Chn * Word Size.  (New)
Changes:-
 o Add support for RCW generation of BCLK
 o Update Documentation to include RCW.
 o Update DAI documentation for label = value DAI modes.
 o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Signed-off-by: NLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch fixes some build warnings in soc-core.c
Changes:-
 o Check the return value of soc_ac97_dev_register()
 o Check return value of calls to device_create_file()
Signed-off-by: NLiam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch fixes a build failure when ASoC debug is enabled.
Signed-off-by: NLiam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

Properly quote a string that had an embedded newline.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>

This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o  Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
 o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.
Signed-off-by: NFrank Mandarino <fmandarino@endrelia.com>
Signed-off-by: NRichard Purdie <rpurdie@rpsys.net>
Signed-off-by: NLiam Girdwood <liam.Girdwood@wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@suse.cz>