Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
0x0 : Power down (mute)
0x1 : 6dB
0x2 : 0 dB
0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)
After the patch, FGAIN volume control works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.
NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
The logic is flawed (e.g. the slave might startup before the
master's rate and sample_bits are set).
Signed-off-by: Karl Beldan <karl.beldan@gmail.com>
Signed-off-by: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added STAC_DELL_M4_3 quirk for Dell systems, also reorganized the
board config switch to assign number of digital muxes, microphones,
and SPDIF muxes via the PCI quirk defined.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warnings:
sound/sound_core.c:460:2: warning: returning void-valued expression
sound/sound_core.c:477:2: warning: returning void-valued expression
sound/sound_core.c:510:5: warning: symbol 'soundcore_open' was not
declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for EAPD on system suspend and disabling EAPD on headphone jack
detection for STAC_DELL_M6 laptops.
This patch fixes the regressions, the silent output on HP of some Dell
laptops (see Novell bnc#446025):
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds twl4030 audio support on omap2evm
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock
[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).
[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A probe function should have a clean return 0 path.
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clean up redudent code and correct building problem in non-mmap mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed the GPIO mask and co initialization in patch_stac92hd71bxx()
so that the gpio_maks for HP_M4 model is set properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only fully documented registers are cached in the WM8990 but additional
registers exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check board preset model instead of codec->subsystem_id in
patch_92hd71bxx() so that other hardwares configured via the model
option work like the given model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Error handling code following a kzalloc should free the allocated data.
The error handling code is adjusted to call pci_disable_device(pci); as
well, as done later in the function
The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,l;
position p1,p2;
expression *ptr != NULL;
@@
(
if ((x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...)) == NULL) S
|
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
)
<... when != x
when != if (...) { <+...x...+> }
x->f = E
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed value for STAC_VREF_EVENT from 0x40 to 0x00 because the
unsol response value is only 6-bits width and the former value
was 1<<6 which is an overrun.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8728 is a high performance stereo DAC designed for applications
such as DVD, home theatre and digital TV.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 8dc840f88d. Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue. Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this patch it is possible to select drivers which require
bestcomm support without bestcomm support being selected. This
patch reworks the bestcomm dependencies to ensure the correct
bestcomm tasks are always enabled.
Reported-by: Hans Lehmann <hans.lehmann@ritter-elektronik.de>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
The digital beep widget may have no mute control, and always enabling
the beep is ofen pretty annoying, especially on laptops.
This patch adds a mixer control "PC Beep Playback Switch" when there
is no mixer amp mute is found, and controls it on software.
Reference: Novell bnc#444572
https://bugzilla.novell.com/show_bug.cgi?id=444572
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).
Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the recent change for multiple HP as line-out switch, only
one of the multiple headphons (usually a wrong one) is toggled
and the other pins are still disabled. This causes the silent output
problem on some Dell laptops.
Also, the hp_switch check is screwed up when a line-in or a mic-in
jack exists. This is added as an additional output, but hp_switch
check doesn't take it into account.
This patch fixes these issues: simplify hp_switch check by using
the NID instead of bool, and clean up / fix the toggle of HP pins
in unsol event handler code.
Reference: Novell bnc#443267
https://bugzilla.novell.com/show_bug.cgi?id=443267
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input pins are sometimes not initialized properly because
of the optimization check of the current pinctl code.
Force to initialize the mic input pins so that they can be set up
properly even if they were in a weird state. But keep other input
pins if already set up as input, since this could be an extra mic
pin.
Reference: Novell bnc#443738
https://bugzilla.novell.com/show_bug.cgi?id=443738
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The creation of analog-mux mixer element is missing in
patch_stac9200() due to the dynamic allocation patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Make the HP EliteBook 8530p use AD1884A model laptop
ALSA: gusextreme: Fix build errors
ALSA: hdsp: check for iobox and upload firmware during ioctl
ALSA: HDSP: check for io box before uploading firmware
ALSA: hda - Add another HP model (6730s) for AD1884A
alsa: fix snd_BUG_on() and friends
ALSA: hda - Add a quirk for MEDION MD96630
ALSA: hda - Limit the number of GPIOs show in proc
Added a QUIRK to patch_analog.c for the HP Elitebook 8530p
(IDs 0x103c:0x30e7) to use AD1884A model 'laptop' by default.
Playback and Capture confirmed working.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
currently, the error message when trying to run hdspmixer or hdspconf
if the breakout box is not connected is somehow misleading, since it
asks the user to upload the firmware.
this patch adds a test, whether the breakout box is connected and
tries to upload the firmware in the case, that it is not present, e.g.
because of power failures of the breakout box.
[Minor coding-style fixes by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
currently the hdsp driver tries to upload the firmware, even if the
io box is not connected. this patch adds a check for the io box
before trying to upload the firmware.
thus instead of messages complaining about the fifo status and firmware
loading failure, the driver gives a message that no multiface or
digiface is connected.
[A minor coding-style fix by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added model=laptop for another HP machine (103c:3614) with AD1884A
codec.
Signed-off-by: Michel Marti <mma@objectxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for more sample rates, different crystals
and split playback/capture rates.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to TRM, 256*Fs clock output should be enabled
when TWL4030 is in slave mode, not master.
This allows sound to work on OMAP3 Pandora, which uses
256*Fs clock.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C24xx dma does not allow more than one buffer to be enqueue prior to
the dma transfers starting. This patch adds an additional parameter to
s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load
value.
Signed-off-by: David Anders <danders at amltd.com>
Signed-off-by: Mark 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: Mark brown <broonie@opensource.wolfsonmicro.com>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.
[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION.
sound/soc/codecs/ad73311.c
This patch removes the said #include <version.h>.
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call device_create_file only once in snd_soc_dapm_sys_add function.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda: make a STAC_DELL_EQ option
ALSA: emu10k1 - Add more invert_shared_spdif flag to Audigy models
ALSA: hda - Add a quirk for another Acer Aspire (1025:0090)
ALSA: remove direct access of dev->bus_id in sound/isa/*
sound: struct device - replace bus_id with dev_name(), dev_set_name()
ALSA: Fix PIT lockup on some chipsets when using the PC-Speaker
ALSA: rawmidi - Add open check in rawmidi callbacks
ALSA: hda - Add digital-mic for ALC269 auto-probe mode
ALSA: hda - Disable broken mic auto-muting in Realtek codes
Add support for explicitly enabling the EQ distortion hack for
systems without software biquad support.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk for another Acer Aspier laptop (1025:0090) with ALC883
codec. Reported in Novell bnc#426935:
https://bugzilla.novell.com/show_bug.cgi?id=426935
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed the direct accesses of dev->bus_id in sound/isa/* by replacement
with dev_err() or dev_warn() functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[stripped sound/isa/* changes, replaced with the next patch -- tiwai]
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drivers (e.g. mtpav) may call rawmidi functions in irq handlers
even though the streams are not opened. This results in Oops or panic.
This patch adds the rawmidi state check before actually operating the
rawmidi buffers.
Tested-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We broke O_NONBLOCK handling in OSS dmasound_core in 2.3.11-pre3 - the
original code copied f_flags to open_mode and then checked for
O_NONBLOCK in there, but that got changed to copying f_mode and
O_NONBLOCK has not reached that field in any kernel version.
Since we do not care for any other bits, the fix is obvious...
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
As it is, all instances of ->release() for files that have ->fasync()
need to remember to evict file from fasync lists; forgetting that
creates a hole and we actually have a bunch that *does* forget.
So let's keep our lives simple - let __fput() check FASYNC in
file->f_flags and call ->fasync() there if it's been set. And lose that
crap in ->release() instances - leaving it there is still valid, but we
don't have to bother anymore.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The digital mic wasn't detected properly for ALC269 auto-probing mode
because of its widget number. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of automatic mic-muting is broken in some cases.
The code assumes that the pin nids <= 0x18, but the digital pins can
be less than 0x18.
Also, it assumes the front-mic being the internal mic, but it depends
on the hardware implementation actually.
Instead of complex case-fixes, better to disable the code as now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-2.6:
sparc64: Add missing null terminating entry to bq4802_match[].
sparc: use the new byteorder headers
rtc-m48t59: shift zero year to 1968 on sparc (rev 2)
dbri: check dma_alloc_coherent errors
sparc64: remove byteshifting from out* helpers
Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
It is based on the former eti_b1_wm8731.c file, using the atmel scc API.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Ateml AT91 and AVR32 SoC share common IP for audio and can share the
same driver code using the atmel-ssc API provided for both architectures.
Do this, creating a new unified atmel ASoC architecture to replace the
previous at32 and at91 ones.
[This was contributed as a patch series for reviewability but has been
squashed down to a single commit to help preserve both the history and
bisectability. A small bugfix from Jukka is included.]
Tested-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
