| Commit message (Collapse) | Author | Age | Files | Lines |
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This patch fixes the pincfg assignment for the AE-5, which was
previously using the Recon3D pincfg's by mistake.
Fixes: d06feaf02fe6 ("ALSA: hda/ca0132 - Add pincfg for AE-5")
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds a new PCI subsys ID for the ZxR, as found and tested by
other users. Without a way to know if any Z's use it as well, it keeps
the quirk of QUIRK_SBZ and goes through the HDA subsys test function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We need to call pci_iounmap() instead of iounmap() for the regions
obtained via pci_iomap() call for some archs that need special
treatment.
Fixes: aa31704fd81c ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for
the mute LED control over mic3 pin. Added the corresponding quirk
entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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PCM OSS layer may allocate a few temporary buffers, one for the core
read/write and another for the conversions via plugins. Currently
both are allocated via vmalloc(). But as the allocation size is
equivalent with the PCM period size, the required size might be quite
small, depending on the application.
This patch replaces these vmalloc() calls with kvzalloc() for covering
small period sizes better. Also, we use "z"-alloc variant here for
addressing the possible uninitialized access reported by syzkaller.
Reported-by: syzbot+1cb36954e127c98dd037@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since the commit c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for
mic mute LED controls") we allow enabling the mic mute LED with
multiple ADCs. The commit changed the function return value to be
zero or a negative error, while this change was overlooked in the
thinkpad_acpi helper code where it still expects a positive return
value for success. This eventually leads to a NULL dereference on a
system that has only a mic mute LED.
This patch corrects the return value check in the corresponding code
as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621
Fixes: c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls")
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/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack"
* tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
ALSA: dice: fix to wait for releases of all ALSA character devices
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In a former commit, PCM constraint based on LCM of SYT_INTERVAL was
obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in
some cases that max/min values of size of buffer/period is not
multiples of one of values of SYT_INTERVAL. For example, pulseaudio
always fail to configure PCM substream.
This commit changes strategy for the PCM rule. Although the buggy rules
had a single dependency (rate from period, period from rate, rate from
buffer, buffer from rate), a revised rule has double dependencies
(period from period/rate, buffer from buffer/rate). A step of value is
calculated with table of SYT_INTERVAL and list of available rates. This
prevents interval template which brings -EINVAL to a call of
snd_interval_refine().
Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Creative Audigy SE (SB0570) card currently exhibits an audible pop
whenever playback is stopped or resumed, or during silent periods of an
audio stream. Initialise the IZD bit to the 0 to eliminate these pops.
The Infinite Zero Detection (IZD) feature on the DAC causes the output
to be shunted to Vcap after 2048 samples of silence. This discharges the
AC coupling capacitor through the output and causes the aforementioned
pop/click noise.
The behaviour of the IZD bit is described on page 15 of the WM8768GEDS
datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples
will cause all outputs to be connected directly to VCAP. This also
happens if 2048 consecutive zero input samples are applied to all 6
channels, and IZD=0. It will be removed as soon as any channel receives
a non-zero input". I believe the second sentence might be referring to
IZD=1 instead of IZD=0 given the observed behaviour of the card.
This change should make the DAC initialisation consistent with
Creative's Windows driver, as this popping persists when initialising
the card in Linux and soft rebooting into Windows, but is not present on
a cold boot to Windows.
Signed-off-by: Alex Stanoev <alex@astanoev.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In a development period for Linux kernel v4.20, drivers in ALSA firewire
stack were changed to wait for releases of all ALSA character devices at
.remove callback of bus driver. However, ALSA dice driver is partly out
of this change. This bug can bring fault to user process which holds
the last of character device in unplugging.
This commit fixes the driver to wait in the callback.
Fixes: 61ccc6f6b27c ('ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates. Here are
highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060"
* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
ASoC: stm32: sai: fix master clock naming
ASoC: stm32: add clock dependency for sai
ALSA: hda/ca0132 - Actually fix microphone issue
ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
ASoC: wm2000: Remove wm2000_read helper function
ASoC: cs42l51: fix mclk support
ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
ASoC: cs42l51: add mclk support
ASoC: stm32: sai: set sai as mclk clock provider
ASoC: dt-bindings: add mclk support to cs42l51
ASoC: dt-bindings: add mclk provider support to stm32 sai
ASoC: soc-core: fix trivial checkpatch issues
ASoC: dapm: Add support for hw_free on CODEC to CODEC links
ASoC: Intel: kbl_da7219_max98927: minor white space clean up
ALSA: i2c/cs8427: Fix int to char conversion
ALSA: doc: Brush up the old writing-an-alsa-driver
ASoC: rsnd: tidyup SSICR::SWSP for TDM
ASoC: rsnd: enable TDM settings for SSI parent
ASoC: pcm3168a: add hw constraint for capture channel
...
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional fixes for the next release
A couple of fixes for build bot issues in the STM32 SAI driver.
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Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider")
Fix warning issued by strncat when bound equals to source length.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider")
Add COMMON_CLK dependency for STM32 SAI,
as it is required by clock provider.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.0/v4.20
As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:
- More component transitions from Morimoto-san, I think we're about
finished with this. Thanks for all the hard work!
- Morimoto-san also added a bunch of for_each_foo macros
- A bunch of cleanups and fixes for DAPM from Charles.
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373.
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060.
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startup() and shutdown() hooks are called for both substreams,
so stopping either substream when another is running breaks the
latter.
E.g. playback breaks if capture is stopped when playback is running.
Move code from startup() and shutdown() to resume() and suspend()
hooks respectively to fix this issue
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The return type "unsigned int" was used by the wm2000_read()
function despite of the aspect that it will eventually return
a negative error code.
The resulting function doesn't add much to the code, so replace
wm2000_read with regmap_read.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The MCLK clock is made optional for cs42l51 codec.
However, ASoC DAPM clock supply widget, expects the clock to be defined
unconditionally.
Register MCLK DAPM conditionally in codec driver,
depending on clock presence in DT.
Fixes: 5e8d63a726f8 ("ASoC: cs42l51: add mclk support")
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Increase the address value width in the debug log from 4 digits to
8 digits to allow for DSP cores with larger memory address ranges.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The wm_adsp_buffer struct is the control header of a circular
buffer used to transfer data from the firmware over the
control interface to an ALSA compressed stream.
The original names of the fields pointing to the data buffer
were based on ADSP2V2 memory layout where they correspond to
{XM, XM, YM}. But this circular buffer could be used on other
types of DSP core that have different memory region types.
Also the names and description of the size fields were not
very clear. The field names and descriptions have been changed
to be generic and not imply any particular memory types.
This patch updates the wm_adsp driver to the new field names.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add MCLK dapm to allow configuration of cirrus CS42l51
codec as a master clock consumer.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add master clock generation support in STM32 SAI.
The master clock provided by SAI can be used to feed a codec.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fix a few trivial aka cosmetic only checkpatch issues like long lines,
wrong indentations, spurious blanks and newlines, missing newlines,
multi-line comments etc.
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently, on power down for a CODEC to CODEC DAI link we only call
digital_mute and shutdown. Provide a little more flexibility for drivers
by adding a call to hw_free as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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I just added a couple missing tabs.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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R-Car datasheet is indicating that WS output settings of SSICR::SWSP
is inverted on TDM mode from non TDM mode settings.
But, it is meaning that TDM should use 0 here.
Without this patch, sound input/output 1ch will be 2ch, 2ch will be 3ch
..., be jumbled on I2S + TDM settings. This patch fixup it.
This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some SSIs are sharing each pins (= WS/CLK pin for playback/capture).
Then, SSI parent needs control WS/CLK setting for SSI slave.
In such case, SSI parent needs TDM settings if SSI slave is working as
TDM mode. But it is not cared in current driver.
It can't capture TDM sound without this patch if SSIs were pin sharing.
This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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LEFT_J / I2S only can use TDM.
commit 594680ea4a394 ("ASoC: pcm3168a: add hw constraint for channel")
commit 3809688980205 ("ASoC: pcm3168a: add HW constraint for non
RIGHT_J") added channel constraint for it, but, it was only for playback.
This patch adds constraint for capture.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The STA32x chips feature an XTI clock input that needs to be stable before
the reset signal is released. Therefore, the chip driver needs to get a
handle to the clock. Instead of relying on other parts of the system to
enable the clock, let the codec driver grab a handle itself.
In order to keep existing boards working, clock support is made optional.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add driver for NAU88C22.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Similar to the following:
commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting")
commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting")
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Allwinner A64 uses the same digital codec part as in A33, so we need
to build this driver on ARM64 as well.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The internal codec on A64 is split into 2 parts. The analog path controls
are routed through an embedded custom register bus accessed through
the PRCM block.
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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It will be reused by sun50i-codec-analog later.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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BCLK / LRCK ratio should be sample size * channels, but it was
hardcoded to 32 (0x1 is 32 as per A33 and A64 datasheets).
Calculate it basing on sample size and number of channels.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The I2S block used for the audio codec in the A64 differs from other 3
I2S modules in A64 and isn't compatible with H3. But it is very similar
to what is found in A10(sun4i). However, its TX FIFO is
located at a different address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DMA API does its own zone decisions based on the coherent_dma_mask.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds da7219_max98927 machine driver entry into
machine table
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Kabylake I2S machine driver with:
DA7219 audio codec(SSP1) and MAXIM98927(SSP0) speaker amplifier.
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some 3 pole connectors report impedance greater than threshold of
1000Ohm. Thus, da7219 reports them as LINEOUT.
Adding the SND_JACK_LINEOUT type so that we don't fail to detect
any 3 pole jack type.
Also, changing
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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pcm3168a is supporting TDM on I2S/Left_J, but there is no
settings for it. This patch add it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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LEFT_J / I2S only can use TDM.
This patch adds channel constraint for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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RIGHT_J only can handle 16bit data bits.
Current driver just errored if user requests non RIGHT_J
+ 16bit combination. But it is not useful for user.
This patch adds HW constraint for it, and avoid
error on such situation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current audio-graph-card is supporting mclk-fs on CPU node
side only. But having Codec node also is good idea.
It will be just ignored if not defined.
"rcpu_ep" is same as "cpu_ep", This patch tidyup it, too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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