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Without this change, the HDMI/DP codec will be recognised as a
generic codec, and there is no sound when playing through this codec.
As suggested by NVidia side, after adding the new ID in the driver,
the sound playing works well.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When a sequencer queue is created in snd_seq_queue_alloc(),it adds the
new queue element to the public list before referencing it. Thus the
queue might be deleted before the call of snd_seq_queue_use(), and it
results in the use-after-free error, as spotted by syzkaller.
The fix is to reference the queue object at the right time.
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit f6a0dd107ad0c8b59d1c9735eea4b8cb9f460949.
The commit caused a regression on LINE6 Transport that has no control
caps. Although reverting the commit may result back in a spurious
error message for some device again, it's the simplest regression fix,
hence it's taken as is at first. The further code fix will follow
later.
Fixes: f6a0dd107ad0 ("ALSA: line6: Only determine control port properties if needed")
Reported-by: Igor Zinovev <zinigor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_seq_pool_done() syncs with closing of all opened threads, but it
aborts the wait loop with a timeout, and proceeds to the release
resource even if not all threads have been closed. The timeout was 5
seconds, and if you run a crazy stuff, it can exceed easily, and may
result in the access of the invalid memory address -- this is what
syzkaller detected in a bug report.
As a fix, let the code graduate from naiveness, simply remove the loop
timeout.
BugLink: http://lkml.kernel.org/r/CACT4Y+YdhDV2H5LLzDTJDVF-qiYHUHhtRaW4rbb4gUhTCQB81w@mail.gmail.com
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.10
As well as the usual smattering of driver specific fixes collected since
the merge window this has one particularly important fix to the core for
handling of aux_devs which was broken during the merge window by some of
the componentization refactoring.
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'asoc/fix/tlv320aic3x' and 'asoc/fix/topology' into asoc-linus
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kcontrol->private_value is being kfree'd after kcontrol has been freed
(in previous call to snd_ctl_remove). Instead, fix this by kfreeing
the private_value before kcontrol.
CoverityScan CID#1388311 "Read from pointer after free"
Fixes: eea3dd4f1247a ("ASoC: topology: Only free TLV for volume mixers of a widget")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The RESET register only have one self clearing bit and it should not be
cached. If it is cached, when we sync the registers back to the chip we
will initiate a software reset as well, which is not desirable.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The i2s clock pre-divider 1 is used for both i2s1 and sysclk.
The i2s1 is usually used for the main i2s and the pre-divider
will be set in hw_params function.
However, if i2s2 is used, the pre-divider is not set in the hw_params
function and the default value of i2s clock pre-divider 1 is too high
for sysclk and DMIC usage. Fix by overriding default divider value to 2.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The clk_ref_div is not configured in the correct position of the
register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned
the wrong value.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Change to correct name of the register function.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/fix/dwc', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi-codec' into asoc-linus
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The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e.
activate DMA on FIFO when only 2 spaces are left.) This means the
DMA must service the fifo within 2 audio samples, which is just not
enough time for many use cases with high data rate. In many
configurations the audio channel slips (causing l/r swap in stereo
configurations, or channel slipping in multi-channel configurations).
This patch gives more breathing room and allows the SSI to operate
reliably by changing the fifio refill watermark to 8.
There is no change in behavior for older chips (with an 8-deep fifo).
Only the newer chips with a 15-deep fifo get the new behavior. I
suspect a new fifo depth setting could be optimized on the older
chips too, but I have not tested.
Signed-off-by: Caleb Crome <caleb@crome.org>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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We can no longer rely on the return value of
devm_snd_dmaengine_pcm_register(...) to check if the DMA
handle is declared in the DT.
Previously this check activated PIO mode but currently
dma_request_chan returns either a valid channel or -EPROBE_DEFER.
In order to activate PIO mode check instead if the interrupt
line is declared. This reflects better what is documented in
the DT bindings (see Documentation/devicetree/bindings/sound/
designware-i2s.txt).
Also, initialize use_pio variable which was never being set
causing PIO mode to never work.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When multiple front-ends are using the same back-end, putting state of a
front-end to STOP state upon receiving pause command will result in backend
stream getting released by DPCM framework unintentionally. In order to
avoid backend to be released when another active front-end stream is
present, put the stream state to PAUSED state instead of STOP state.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Protect against corrupt firmware files by ensuring that the length we
get for the data in a region actually lies within the available firmware
file data buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one
to free it again leads to a double free error. Fix this by removing
the extraneous snd_ctl_free_one call.
Issue found using static analysis with CoverityScan, CID 1372908
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Saved firmware ctx was not never released, so release Firmware
ctx in cleanup routine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Frequency value of zero did not make sense, use same 24.576MHz
setting and only change the clock source in idle mode
Suggested-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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If a module is not available in a pipeline, fail safely rather than
causing oops.
Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support")
was merged but the corresponding clock framework patches have not,
after being bumped from audio to clock to x86 domains. The missing
clock-related patches result in a regression starting with 4.9 with
the audio card not being created.
Rather than reverting this commit and all following updates already
queued up for 4.10, handle run-time dependency on MCLK and fall back
to the previous bit-clock mode. This provides the same functionality
as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain
silent with this fallback but additional patches are needed anyway.
As suggested by Mark Brown, the fallback is only allowed with -ENOENT,
all other run-time errors, including -EPROBE_DEFER, will stop the probe
with no sound card registered.
This patch should be applied to -stable as well as ASoC 4.10 fixes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently binding of auxiliary devices doesn't work as in
soc_bind_aux_dev() function a bound component is not being added
to any list and in soc_probe_aux_devices() we are trying to walk
the component_dev_list list to probe auxiliary components but
at that time this list doesn't contain any auxiliary components
since they are being added to the card only in soc_probe_component().
This patch adds a list to the card where are stored bound but not
probed auxiliary devices, so that all aux devices can be probed.
Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit 6b7e95d1336b9eb0d4c6db190ce756480496bd13. This commit
is based on a concern about value of the given parameter. It's expected
to be ORed value with some enumeration-constants, thus often it can not be
one of the enumeration-constants. I understood that this is out of
specification and causes implementation-dependent issues.
In C language specification, enumerated type can be interpreted as an
integer type, in which all of enumeration-constants in corresponding
enumerator-list can be stored. Implementations can select one of char,
signed int and unsigned int as its type, and this selection is
implementation-dependent.
In GCC, a signed integer is selected when at least one of
enumeration-constants has negative value, else an unsigned integer is
selected. This behaviour can be switched by -fshort-enums to short type.
Anyway, the type can be decided after scanning all of
enumeration-constants.
Totally, there's no rules to constrain the value of enumerated type to
be one of enumeration-constants. In short, in enumerated type, decision
of actual type for the type is the most important and
enumeration-constants are just used for the decision, thus it's permitted
to have an integer value in a range of enumeration-constants. In our case,
actual type for the type is currently deterministic to be either char or
unsigned int. Under GCC, it's unsigned int.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 16200948d83 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583bb ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Although the old quirk table showed ASUS X71SL with ALC663 codec being
compatible with asus-mode3 fixup, the bugzilla reporter explained that
asus-model8 fits better for the dual headphone controls. So be it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=191781
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS ROG Ranger VIII with ALC1150 codec requires the extra GPIO pin to
up for the front panel. Just use the existing fixup for setting up
the GPIO pins.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189411
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The 'amdtp_stream' structure is initialized by a call of
'amdtp_stream_init()'. Although a parameter of this function is for bit
flags of packet attributes, its type is enumerator.
This commit changes the type so that it's proper for a bit flags.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This module has a bug not to return error code in a case that data
structure for transmitted packets fails to be initialized.
This commit fixes the bug.
Fixes: 35efa5c489de ("ALSA: firewire-tascam: add streaming functionality")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ALSA fireworks driver has a bug not to call an API to destroy
'cmp_connection' structure for input direction. Currently this causes no
issues because it just destroys 'mutex' structure, while it's better to
fix it for future work.
Fix: d23c2cc4485d ("ALSA: fireworks/bebob/dice/oxfw: allow stream destructor after releasing runtime")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ktime_set(S,N) was required for the timespec storage type and is still
useful for situations where a Seconds and Nanoseconds part of a time value
needs to be converted. For anything where the Seconds argument is 0, this
is pointless and can be replaced with a simple assignment.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
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ktime is a union because the initial implementation stored the time in
scalar nanoseconds on 64 bit machine and in a endianess optimized timespec
variant for 32bit machines. The Y2038 cleanup removed the timespec variant
and switched everything to scalar nanoseconds. The union remained, but
become completely pointless.
Get rid of the union and just keep ktime_t as simple typedef of type s64.
The conversion was done with coccinelle and some manual mopping up.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
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There is no point in having an extra type for extra confusion. u64 is
unambiguous.
Conversion was done with the following coccinelle script:
@rem@
@@
-typedef u64 cycle_t;
@fix@
typedef cycle_t;
@@
-cycle_t
+u64
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: John Stultz <john.stultz@linaro.org>
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This was entirely automated, using the script by Al:
PATT='^[[:blank:]]*#[[:blank:]]*include[[:blank:]]*<asm/uaccess.h>'
sed -i -e "s!$PATT!#include <linux/uaccess.h>!" \
$(git grep -l "$PATT"|grep -v ^include/linux/uaccess.h)
to do the replacement at the end of the merge window.
Requested-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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This reverts commit 16200948d8353fe29a473a394d7d26790deae0e7.
The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.
As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.
Fixes: 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- support for stereo DAPM controls
- some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- a new version of the topology ABI; this should be backward
compatible
- updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- yet another race fix at disconnection
- tolerated packet size calculation for some Android devices
- quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- improvement of Dell pin fixup mapping
- quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- replace all kthread usages with simple works"
* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
ALSA: usb-audio: Eliminate noise at the start of DSD playback.
ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
ASoC: topology: avoid uninitialized kcontrol_type
ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
ASoC: zte: spdif and i2s drivers are not zx296702 specific
ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
ASoC: rsnd: enable/disable ADG when suspend/resume timing
ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
ALSA: cs46xx: add a new line
ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
ASoC: nau8825: disable sinc filter for high THD of ADC
ALSA: usb-audio: more tolerant packetsize
ALSA: usb-audio: avoid setting of sample rate multiple times on bus
ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
...
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Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.
Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).
- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track
[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.
[Solution]
This patch adds DSD silence pattern to the endpoint settings.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds native DSD support for the following devices.
- TEAC NT-503
- TEAC UD-503
- TEAC UD-501
(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
DSD mode, same as Denon/Marantz devices.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.10
There's been a few bits of framework work this time around and quite a
lot of cleanups and improvements to existing code:
- Support for stereo DAPM controls from Chen-yu Tsai.
- Some initial work on the of-graph sound card from Morimoto-san, the
main bulk of this is currently in binding review.
- Lots of Renesas cleanups from Morimoto-san and sunxi work from
Chen-yu Tsai.
- regmap conversions of the remaining AC'97 drivers from Lars-Peter
Clausen.
- A new version of the topology ABI from Mengdong Lin.
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek
RT5665.
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'asoc/topic/zte' into asoc-next
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The macro ZX_SPDIF_CLK_RAT should be 2 instead of 4. With this
fix, we can get correct audio output on HDMI through SPDIF interface.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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ZTE ZX SPDIF and I2S drivers can work on not only ZX296702 but also
other ZTE ZX family SoCs like ZX296718, which is an arm64 platform.
Let's make a few renaming and tweak the Kconfig a bit to get the drivers
available for other ZTE ZX platforms.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Reviewed-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used for matching instead and the
defines are unused. The last user of these defines was removed in commit
f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So
remove the defines as well.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used for matching instead and the
defines are unused. The last user of these defines was removed in commit
f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So
remove the defines as well.
This also means the wm9712.h file no longer has any content and can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the regmap conversion ac97_read/ac97_write are just simple wrappers
around snd_soc_read/snd_soc_write. Use those instead directly and remove
the wrappers. Also use snd_soc_update_bits() were appropriate.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Marek Vasut <marex@denx.de>
Acked-by: Marek Vasut <marex@denx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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