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'asoc/fix/ssm2602' and 'asoc/fix/topology' into asoc-linus
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Structure i2c_driver does not need to set the owner field, as this will
be populated by the driver core.
Generated by scripts/coccinelle/api/platform_no_drv_owner.cocci.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now we can replace Codec to Component. Let's do it.
Note:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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commit aeec6cc08215 ("ASoC: Intel: bytcr_rt5651: Configure PLL1 before
using it") is using codec->dev, but codec is replaced to component.
Let's use component
Fixes: aeec6cc08215 ("ASoC: Intel: bytcr_rt5651: Configure PLL1 before using it")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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commit c88d31153356 ("ASoC: amd: Enable da7219 master clock using common
clock framework") is using rtd->codec, but codec is replaced to component.
Let's use component
Fixes: c88d31153356 ("ASoC: amd: Enable da7219 master clock using common clock framework")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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With the current exit return path of the ctl_info allocation failure
cval is not being freed resulting in a memory leak. Fix this by kfree'ing
it on the return.
Detected by CoverityScan, CID#1466878 ("Resource Leak")
Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The previous fix 40cab6e88cb0 ("ALSA: pcm: Return -EBUSY for OSS
ioctls changing busy streams") introduced some mutex unbalance; the
check of runtime->oss.rw_ref was inserted in a wrong place after the
mutex lock.
This patch fixes the inconsistency by rewriting with the helper
functions to lock/unlock parameters with the stream check.
Fixes: 40cab6e88cb0 ("ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Smatch complains that "tmp" can be uninitialized if we do a zero size
write.
Fixes: 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix the topology kcontrol string handling so that string pointer
references are strdup()ed instead of being copied. This fixes issues
with kcontrol templates on the stack or ones that are freed. Remember
and free the strings too when topology is unloaded.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
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Show paused ALSA aloop device as inactive, i.e. the control
"PCM Slave Active" set as false. Notification sent upon state change.
This makes it possible for client capturing from aloop device to know if
data is expected. Without it the client expects data even if playback
is paused.
Signed-off-by: Robert Rosengren <robert.rosengren@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When trying to use the driver (e.g. aplay *.wav), the 4MiB DMA buffer
will get mmapp'ed in 16KiB chunks. But this fails with the 2nd 16KiB
area, as the page offset is outside of the VMA range (size), which is
currently used as size parameter in snd_pcm_lib_default_mmap(). By
using the DMA buffer size (dma_bytes) instead, the complete DMA buffer
can be mmapp'ed and the issue is fixed.
This issue was detected on an ARM platform (TI AM57xx) using the RME
HDSP MADI PCIe soundcard.
Fixes: 657b1989dacf ("ALSA: pcm - Use dma_mmap_coherent() if available")
Signed-off-by: Stefan Roese <sr@denx.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some functions definitions have either the initial open brace and/or
the closing brace outside of column 1.
Move those braces to column 1.
This allows various function analyzers like gnu complexity to work
properly for these modified functions.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Paul Moore <paul@paul-moore.com>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Acked-by: Dave Chinner <dchinner@redhat.com>
Reviewed-by: Darrick J. Wong <darrick.wong@oracle.com>
Acked-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Acked-by: Martin K. Petersen <martin.petersen@oracle.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
Acked-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Acked-by: Martin K. Petersen <martin.petersen@oracle.com>
Acked-by: Steven Rostedt (VMware) <rostedt@goodmis.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
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Make the "clock valid" control a global control instead of a mixer
so that it doesn't appear in mixer applications.
Additionally, remove the check for writeability prohibited by spec, and
Use common code to read the control value.
Tested with a UAC2 Audio device that presents a clock validity
control. The control still shows up in /proc usbmixer but not
in alsamixer.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This implements UAC2 jack detection support, presenting
jack status as a boolean read-only mono mixer.
The presence of any channel in the UAC2_TE_CONNECTOR
control for a terminal will result in the mixer saying
the jack is connected.
Mixer naming follows the convention in sound/core/ctljack.c,
terminating the mixer with " Jack".
For additional clues as to which jack is being presented,
the name is prefixed with " - Input Jack" or " - Output Jack"
depending on if it's an input or output terminal.
This is required because terminal names are ambiguous
between inputs and outputs and often duplicated -
Bidirectional terminal types (0x400 -> 0x4FF)
"... may be used separately for input only or output only.
These types require two Terminal descriptors. Both have the same type."
(quote from "USB Device Class Definition for Terminal Types")
Since bidirectional terminal types are common for headphone adapters,
this distinguishes between two otherwise identically-named
jack controls.
Tested with a UAC2 audio device with connector control capability.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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OSS PCM stream management isn't modal but it allows ioctls issued at
any time for changing the parameters. In the previous hardening
patch ("ALSA: pcm: Avoid potential races between OSS ioctls and
read/write"), we covered these races and prevent the corruption by
protecting the concurrent accesses via params_lock mutex. However,
this means that some ioctls that try to change the stream parameter
(e.g. channels or format) would be blocked until the read/write
finishes, and it may take really long.
Basically changing the parameter while reading/writing is an invalid
operation, hence it's even more user-friendly from the API POV if it
returns -EBUSY in such a situation.
This patch adds such checks in the relevant ioctls with the addition
of read/write access refcount.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Although we apply the params_lock mutex to the whole read and write
operations as well as snd_pcm_oss_change_params(), we may still face
some races.
First off, the params_lock is taken inside the read and write loop.
This is intentional for avoiding the too long locking, but it allows
the in-between parameter change, which might lead to invalid
pointers. We check the readiness of the stream and set up via
snd_pcm_oss_make_ready() at the beginning of read and write, but it's
called only once, by assuming that it remains ready in the rest.
Second, many ioctls that may change the actual parameters
(i.e. setting runtime->oss.params=1) aren't protected, hence they can
be processed in a half-baked state.
This patch is an attempt to plug these holes. The stream readiness
check is moved inside the read/write inner loop, so that the stream is
always set up in a proper state before further processing. Also, each
ioctl that may change the parameter is wrapped with the params_lock
for avoiding the races.
The issues were triggered by syzkaller in a few different scenarios,
particularly the one below appearing as GPF in loopback_pos_update.
Reported-by: syzbot+c4227aec125487ec3efa@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Integrate the native DSD support quirk codes of "ITF-USB DSD" based DACs.
Now, "is_itf_usb_dsd_2alts_dac()" and "is_itf_usb_dsd_3alts_dac()" is
integrated into one function "is_itf_usb_dsd_dac()".
So, remove the logic to distinguish UD-501 and UD-501V2 by the
"Product Name".
The integration is possible by changing the following two functions.
- snd_usb_select_mode_quirk():
Change the determination condition of the DSD mode switch command,
from the altset number being used, to the audio format being played.
Actually, this operation is same as playback using ASIO driver in
Windows environment.
- snd_usb_interface_dsd_format_quirk():
To which altset supports native DSD is determined by the number of altsets.
Previously, it's a constant "2" or "3".
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are two versions of TEAC UD-501, the normal version and
the vendor updated version(UD-501V2).
They have the same VID/PID, but the num of the altsetting is different,
UD-501 has 2 altsets for stream, and UD-501V2 has 3.
So, add the logic to distinguish them by the Product Name, not by the PID.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add native DSD support quirk for Luxman DA-06 DAC, by adding the
PID/VID 1852:5065.
Rename "is_marantz_denon_dac()" function to "is_itf_usb_dsd_2alts_dac()"
to cover broader device family sharing the same USB audio
implementation(*).
For the same reason, rename "is_teac_dsd_dac()" function to
"is_itf_usb_dsd_3alts_dac()".
(*)
These devices have the same USB controller "ITF-USB DSD", supplied by
INTERFACE Co., Ltd.
"ITF-USB DSD" USB controller has two patterns,
Pattern 1. (2 altsets version)
- Altset 0: for control
- Altset 1: for stream (S32)
- Altset 2: for stream (S32, DSD_U32)
Pattern 2. (3 altsets version)
- Altset 0: for control
- Altset 1: for stream (S16)
- Altset 2: for stream (S32)
- Altset 3: for stream (S32, DSD_U32)
"is_itf_usb_dsd_2alts_dac()" returns true, if the DAC has "Pattern 1"
USB controller, and "is_itf_usb_dsd_3alts_dac()" returns true, if
"Pattern2".
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge for applying more series of fixes for USB DSD support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add native DSD support quirk for TEAC UD-301 DAC,
by adding the PID/VID 0644:804a.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch fixes code readability and should have no functional change.
Correct uac control query functions to account for the 1-based indexing
of USB Audio Class control identifiers.
The function parameter, u8 control, should be the
constant defined in audio-v2.h to identify the control to be checked for
readability or writeability.
This patch fixes all callers that had adjusted, and makes explicit
the mapping between audio_feature_info[] array index and the associated
control identifier.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In loopback_open() and loopback_close(), we assign and release the
substream object to the corresponding cable in a racy way. It's
neither locked nor done in the right position. The open callback
assigns the substream before its preparation finishes, hence the other
side of the cable may pick it up, which may lead to the invalid memory
access.
This patch addresses these: move the assignment to the end of the open
callback, and wrap with cable->lock for avoiding concurrent accesses.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The aloop driver tries to stop the pending timer via timer_del() in
the trigger callback and in the close callback. The former is
correct, as it's an atomic operation, while the latter expects that
the timer gets really removed and proceeds the resource releases after
that. But timer_del() doesn't synchronize, hence the running timer
may still access the released resources.
A similar situation can be also seen in the prepare callback after
trigger(STOP) where the prepare tries to re-initialize the things
while a timer is still running.
The problems like the above are seen indirectly in some syzkaller
reports (although it's not 100% clear whether this is the only cause,
as the race condition is quite narrow and not always easy to
trigger).
For addressing these issues, this patch adds the explicit alls of
timer_del_sync() in some places, so that the pending timer is properly
killed / synced.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It will have a chance speaker no sound after system resume.
To toggle NID 0x53 index 0x2 bit 15 will solve this issue.
This usage will also suitable with ALC256.
Fixes: 4a219ef8f370 ("ALSA: hda/realtek - Add ALC256 HP depop function")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This platform was hardware fixed type for CTIA type for headset port.
Assigned 0x19 verb will fix can't record issue.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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User reports an issue in Ubuntu about the device switch upon playback.
We find the FLL will disalbe when switching headphone to speaker.
The pulseaudio will stop the headphone and close its power. Then,
it just opens the speaker and turn on its power. Therefore,
the supply of system clock does the OFF event and disables FLL.
But the FLL doesn't enable again when the speaker powers on.
The patch adds the recovery of system clock to enable FLL again
for this case. And it covers the case that system clock from MCLK.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Recently released USB Audio Class 3.0 specification
introduces many significant changes comparing to
previous versions, like
- new Power Domains, support for LPM/L1
- new Cluster descriptor
- changed layout of all class-specific descriptors
- new High Capability descriptors
- New class-specific String descriptors
- new and removed units
- additional sources for interrupts
- removed Type II Audio Data Formats
- ... and many other things (check spec)
It also provides backward compatibility through
multiple configurations, as well as requires
mandatory support for BADD (Basic Audio Device
Definition) on each ADC3.0 compliant device
This patch adds initial support of UAC3 specification
that is enough for Generic I/O Profile (BAOF, BAIF)
device support from BADD document.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We've observed too long probe time with Coffee Lake (CFL) machines,
and the likely cause is some communication problem between the
HD-audio controller and the codec chips. While the controller expects
an IRQ wakeup for each codec response, it seems sometimes missing, and
it takes one second for the controller driver to time out and read the
response in the polling mode.
Although we aren't sure about the real culprit yet, in this patch, we
put a workaround by forcing the polling mode as default for CFL
machines; the polling mode itself isn't too heavy, and much better
than other workarounds initially suggested (e.g. disabling
power-save), at least.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199007
Fixes: e79b0006c45c ("ALSA: hda - Add Coffelake PCI ID")
Reported-and-tested-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The driver does not select all the codec drivers that needs.
Fix it by selecting the analog and HDMI codecs.
Cc: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Ezequiel Garcia <ezequiel@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit 24069b589b02 ("ASoC: hdmi-codec: remove multi detection support")
changed the dai_name for the HDMI Codec, breaking the rk3288_hdmi_analog
driver, which fails to register with a:
rk3288-snd-hdmi-analog sound: ASoC: CODEC DAI hdmi-hifi.0 not registered
This commit fixes the dai_name, fixing the issue.
Fixes: 24069b589b02 ("ASoC: hdmi-codec: remove multi detection support")
Signed-off-by: Ezequiel Garcia <ezequiel@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-amd
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This patch fixes a spelling typo in printk
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Use SET_SYSTEM_SLEEP_PM_OPS() macro instead of direct assignment to
.suspend and .resume fields.
This makes driver working after restore from hibernation.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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DA7219 is clock master for other codecs. DA7219 has exposed clock
control by using common clock framework and same is used to enable
and disable clock for all codecs in the system.
TEST=aplay -D hw:0,0 -vv <file>
arecord -D hw:0,0 -f dat -d 5 -vv <file>
aplay -D hw:0,1 -vv <file>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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BT I2S is a bi-directional dai, we will use the same
cpu dai for playback and capture.
TEST=aplay -D hw:0,0 -vv <file>
arecord -D hw:0,0 -f dat -d 5 -vv <file>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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We don't need 2 .remove callback
Fixes: 7480389fb0d8 ("ASoC: twl6040: replace codec to component")
Reported-by: Julia Lawall <julia.lawall@lip6.fr>
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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"HV" and "VREF" will be powered up when jack type detection process is
started and will be powered off when jack type detection process is finished.
It will generate an unexpected interrupt signal when they are powered
up during the capture process. Codec driver will do the jack type detection
process and we can't capture properly before the jack detection process is
finished. This patch will not power off the "HV" and "VREF" widgets if
headset is detected and it will solve the unexpected interrupt issue. As a
result, it will also solve the silence data captured at the beginning in
headset mic recording issue.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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"HV" and "VREF" will be powered up when jack type detection process is
started and will be powered off when jack type detection process is finished.
It will generate an unexpected interrupt signal when they are powered
up during the capture process. Codec driver will do the jack type detection
process and we can't capture properly before the jack detection process is
finished. This patch will not power off the "HV" and "VREF" widgets if
headset is detected and it will solve the unexpected interrupt issue. As a
result, it will also solve the silence data captured at the beginning in
headset mic recording issue.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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SSM2602 driver is broken on recent kernels (at least
since 4.9). User space applications such as amixer or
alsamixer get EIO when attempting to access codec
controls via the relevant IOCTLs.
Root cause of these failures is the regcache_hw_init
function in drivers/base/regmap/regcache.c, which
prevents regmap cache initalization from the
reg_defaults_raw element of the regmap_config structure
when registers are write only. It also disables the
regmap cache entirely when all registers are write only
or volatile as is the case for the SSM2602 driver.
Using the reg_defaults element of the regmap_config
structure rather than the reg_defaults_raw element to
initalize the regmap cache avoids the logic in the
regcache_hw_init function entirely. It also makes this
driver consistent with other ASoC codec drivers, as
this driver was the ONLY codec driver that used the
reg_defaults_raw element to initalize the cache.
Tested on Digilent Zybo Z7 development board which has
a SSM2603 codec chip connected to a Xilinx Zynq SoC.
Signed-off-by: James Kelly <jamespeterkelly@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
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Offload USB audio interface parsing function by
moving quirks to a specially designed location (quirks.c)
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge of for-linus branch for applying the further UAC3 patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When driver register its component to ALSA SoC, almost all drivers are
using snd_soc_register_component(), but soc-generic-dmaengine-pcm is
using snd_soc_add_component().
Existing component function had been assumed that registered component
was allocated, and it calling kfree() for it.
But, the user who used snd_soc_add_component() doesn't.
This patch uses devm_kzalloc() instead of kzalloc() for component,
and doesn't call kree() anymore.
This patch fixes commit be7ee5f32a9a ("ASoC: soc-generic-dmaengine-pcm:
replace platform to component").
Allwinner H3 SoC will crash without this patch.
Thanks Jernej report.
Reported-by: Jernej Skrabec <jernej.skrabec@siol.net>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Jernej Skrabec <jernej.skrabec@siol.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Validate the topology input before we dereference the pointer.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Improve the DPCM BE search debug output to make it easier to debug
issues in topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now when a helper for unreferencing device nodes is available
we can get rid of the local implementation.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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hw_params()
The I2S controller can handle sample rates only up to 96000 and the CPU DAI
has already related constraint set so drop the impossible 176400, 192000
switch cases.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In case of sample rates lower than 44100 currently there is too low MCLK
frequency set for the CODEC. Playback fails with following errors:
$ speaker-test -c2 -t sine -f 1500 -l2 -r 32000
Sine wave rate is 1500.0000Hz
Rate set to 32000Hz (requested 32000Hz)
Buffer size range from 128 to 131072
Period size range from 64 to 65536
Using max buffer size 131072
Periods = 4
Unable to set hw params for playback: Invalid argument
Setting of hwparams failed: Invalid argument
[ 497.883700] max98090 1-0010: Invalid master clock frequency
To fix this the I2S root clock's frequency is increased, depending
on sampling rate.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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