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azx_codec_configure() loops over the codecs found on the given
controller via a linked list. The code used to work in the past, but
in the current version, this may lead to an endless loop when a codec
binding returns an error.
The culprit is that the snd_hda_codec_configure() unregisters the
device upon error, and this eventually deletes the given codec object
from the bus. Since the list is initialized via list_del_init(), the
next object points to the same device itself. This behavior change
was introduced at splitting the HD-audio code code, and forgotten to
adapt it here.
For fixing this bug, just use a *_safe() version of list iteration.
Fixes: d068ebc25e6e ("ALSA: hda - Move some codes up to hdac_bus struct")
Reported-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we met a problem, the codec has valid adcs and input pins,
and they can form valid input paths, but the driver does not build
valid controls for them like "Mic boost", "Capture Volume" and
"Capture Switch".
Through debugging, I found the driver needs to shrink the invalid
adcs and input paths for this machine, so it will move the whole
column bitmap value to the previous column, after moving it, the
driver forgets to set the original column bitmap value to zero, as a
result, the driver will invalidate the path whose index value is the
original colume bitmap value. After executing this function, all
valid input paths are invalidated by a mistake, there are no any
valid input paths, so the driver won't build controls for them.
Fixes: 3a65bcdc577a ("ALSA: hda - Fix inconsistent input_paths after ADC reduction")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
audio-graph-scu-card can handle below connection which is mainly
for sound mixing purpose.
+----------+ +-------+
| CPU0--+--|-->| Codec |
| | | +-------+
| CPU1--+ |
+----------+
>From OF-graph point of view, it should have
CPU0 <-> Codec, and CPU1 <-> Codec on DT.
But current driver doesn't care about 2nd connection
of Codec, because it is dummy from DPCM point of view.
This patch can care 2nd Codec connection, and it should be
supported from OF-graph point of view.
It still have backward compatibility.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's
DAI links, not Card links.
This patch fixup it. Otherwise, audio-graph-card can't handle CPU
component correctly if CPU has mult-DAIs and Card uses only one of them
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's
DAI links, not Card links.
This patch fixup it. Otherwise, audio-graph-card can't handle CPU
component correctly if CPU has mult-DAIs and Card uses only one of them
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make crosstalk functoin optional.
The jack detection can speed up without crosstalk detection.
Let the decision of function usage to platform design.
The patch helps the issue concern as follows:
Google issue 35574278: Chell_headphone pop back from S3
There is a concern as follows:
cras getting blocked for 2 seconds (worst-case 3 seconds)
As I understand, ChromeOS expects resume finishes in 1 seconds.
Video/Audio playing after 3 seconds of resume seems against the spec.
If we really have to make the choice I would choose pop noise instead
of waiting for 3 seconds.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: John Hsu <supercraig0719@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the issue that mic type detection error after resume.
The microphone type detection procedure will recognize
testing signal on JKSLV pin, but before the procedure,
JKSLV already had supply voltage, that results in the failure.
Therefore, the patch turns off the power and reset the jack type
configuration before suspend. Then redo the jack detection
procedure after resume.
The patch help to fix the issue as follows:
Google issue 37973093: CTIA/OMTP jack type detection failure after resume
Reported Issue
Chrome OS Version : ChromeOS R59-9460.13.0
Type of hardware : DVT sample
What steps will reproduce the problem?
(1 Play a music
(2 Insert a headphones
(3 Close laptop lid 3 sec then open it
What is the expected output?
The music is normal in the headphones.
What do you see instead?
Singer voice in the music is not clear.
How frequently does this problem reproduce?
Always
What is the impact to the user, and is there a workaround?
If so, what is it?
Re-insert the headset or close the laptop lid and
then open it again can be repaired.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: John Hsu <supercraig0719@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Assign default value for codec private data when property not given.
If without those default value and property, the codec will work
abnormally.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: John Hsu <supercraig0719@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It adds ASoC driver for AUD96P22 stereo audio codec integrated on ZTE
ZX family SoCs. The driver includes the support for a number of volume
and mute controls, and power bits for various playback and recording
components.
Due to that the board for testing only supports playback, recording
support is untested.
Signed-off-by: Baoyou Xie <baoyou.xie@linaro.org>
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Without CONFIG_I2C, we get a build failure:
sound/soc/codecs/es8316.c:633:1: error: data definition has no type or storage class [-Werror]
sound/soc/codecs/es8316.c:633:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/es8316.c:633:1: error: parameter names (without types) in function declaration [-Werror]
sound/soc/codecs/es8316.c:623:26: error: 'es8316_i2c_driver' defined but not used [-Werror=unused-variable]
This adds the required Kconfig dependency.
Fixes: b8b88b70875a ("ASoC: add es8316 codec driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The array ni_div does not need to be in global scope and is not
modified, so make it static const.
Cleans up sparse warning:
"symbol 'ni_div' was not declared. Should it be static?"
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes this WARNING
sound/soc/sh/rcar/ssi.c:285:5-14: WARNING: Unsigned expression\
compared with zero: main_rate < 0
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It doesn't use asm header. We can add COMPILE_TEST
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Broxton-T was a forgotten child and we didn't apply the quirks for
Skylake+ properly. Meanwhile, a quirk for reducing the DMA latency
seems specific to the early Broxton model, so we leave as is.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SND_SOC_DAIFMT_MASTER bits are defined to specify the master/slave
mode for Codec, not I2S. So the I2S master/slave mode should be flipped
according to SND_SOC_DAIFMT_MASTER bits.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Currently compress driver hardcodes direction as playback to get
substream from the stream. This results in getting the incorrect
substream for compressed capture usecase.
To fix this, remove the hardcoding and derive substream based on
the stream direction.
Signed-off-by: Satish Babu Patakokila <sbpata@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
At Linux v3.5, packet processing can be done in process context of ALSA
PCM application as well as software IRQ context for OHCI 1394. Below is
an example of the callgraph (some calls are omitted).
ioctl(2) with e.g. HWSYNC
(sound/core/pcm_native.c)
->snd_pcm_common_ioctl1()
->snd_pcm_hwsync()
->snd_pcm_stream_lock_irq
(sound/core/pcm_lib.c)
->snd_pcm_update_hw_ptr()
->snd_pcm_udpate_hw_ptr0()
->struct snd_pcm_ops.pointer()
(sound/firewire/*)
= Each handler on drivers in ALSA firewire stack
(sound/firewire/amdtp-stream.c)
->amdtp_stream_pcm_pointer()
(drivers/firewire/core-iso.c)
->fw_iso_context_flush_completions()
->struct fw_card_driver.flush_iso_completion()
(drivers/firewire/ohci.c)
= flush_iso_completions()
->struct fw_iso_context.callback.sc
(sound/firewire/amdtp-stream.c)
= in_stream_callback() or out_stream_callback()
->...
->snd_pcm_stream_unlock_irq
When packet queueing error occurs or detecting invalid packets in
'in_stream_callback()' or 'out_stream_callback()', 'snd_pcm_stop_xrun()'
is called on local CPU with disabled IRQ.
(sound/firewire/amdtp-stream.c)
in_stream_callback() or out_stream_callback()
->amdtp_stream_pcm_abort()
->snd_pcm_stop_xrun()
->snd_pcm_stream_lock_irqsave()
->snd_pcm_stop()
->snd_pcm_stream_unlock_irqrestore()
The process is stalled on the CPU due to attempt to acquire recursive lock.
[ 562.630853] INFO: rcu_sched detected stalls on CPUs/tasks:
[ 562.630861] 2-...: (1 GPs behind) idle=37d/140000000000000/0 softirq=38323/38323 fqs=7140
[ 562.630862] (detected by 3, t=15002 jiffies, g=21036, c=21035, q=5933)
[ 562.630866] Task dump for CPU 2:
[ 562.630867] alsa-source-OXF R running task 0 6619 1 0x00000008
[ 562.630870] Call Trace:
[ 562.630876] ? vt_console_print+0x79/0x3e0
[ 562.630880] ? msg_print_text+0x9d/0x100
[ 562.630883] ? up+0x32/0x50
[ 562.630885] ? irq_work_queue+0x8d/0xa0
[ 562.630886] ? console_unlock+0x2b6/0x4b0
[ 562.630888] ? vprintk_emit+0x312/0x4a0
[ 562.630892] ? dev_vprintk_emit+0xbf/0x230
[ 562.630895] ? do_sys_poll+0x37a/0x550
[ 562.630897] ? dev_printk_emit+0x4e/0x70
[ 562.630900] ? __dev_printk+0x3c/0x80
[ 562.630903] ? _raw_spin_lock+0x20/0x30
[ 562.630909] ? snd_pcm_stream_lock+0x31/0x50 [snd_pcm]
[ 562.630914] ? _snd_pcm_stream_lock_irqsave+0x2e/0x40 [snd_pcm]
[ 562.630918] ? snd_pcm_stop_xrun+0x16/0x70 [snd_pcm]
[ 562.630922] ? in_stream_callback+0x3e6/0x450 [snd_firewire_lib]
[ 562.630925] ? handle_ir_packet_per_buffer+0x8e/0x1a0 [firewire_ohci]
[ 562.630928] ? ohci_flush_iso_completions+0xa3/0x130 [firewire_ohci]
[ 562.630932] ? fw_iso_context_flush_completions+0x15/0x20 [firewire_core]
[ 562.630935] ? amdtp_stream_pcm_pointer+0x2d/0x40 [snd_firewire_lib]
[ 562.630938] ? pcm_capture_pointer+0x19/0x20 [snd_oxfw]
[ 562.630943] ? snd_pcm_update_hw_ptr0+0x47/0x3d0 [snd_pcm]
[ 562.630945] ? poll_select_copy_remaining+0x150/0x150
[ 562.630947] ? poll_select_copy_remaining+0x150/0x150
[ 562.630952] ? snd_pcm_update_hw_ptr+0x10/0x20 [snd_pcm]
[ 562.630956] ? snd_pcm_hwsync+0x45/0xb0 [snd_pcm]
[ 562.630960] ? snd_pcm_common_ioctl1+0x1ff/0xc90 [snd_pcm]
[ 562.630962] ? futex_wake+0x90/0x170
[ 562.630966] ? snd_pcm_capture_ioctl1+0x136/0x260 [snd_pcm]
[ 562.630970] ? snd_pcm_capture_ioctl+0x27/0x40 [snd_pcm]
[ 562.630972] ? do_vfs_ioctl+0xa3/0x610
[ 562.630974] ? vfs_read+0x11b/0x130
[ 562.630976] ? SyS_ioctl+0x79/0x90
[ 562.630978] ? entry_SYSCALL_64_fastpath+0x1e/0xad
This commit fixes the above bug. This assumes two cases:
1. Any error is detected in software IRQ context of OHCI 1394 context.
In this case, PCM substream should be aborted in packet handler. On the
other hand, it should not be done in any process context. TO distinguish
these two context, use 'in_interrupt()' macro.
2. Any error is detect in process context of ALSA PCM application.
In this case, PCM substream should not be aborted in packet handler
because PCM substream lock is acquired. The task to abort PCM substream
should be done in ALSA PCM core. For this purpose, SNDRV_PCM_POS_XRUN is
returned at 'struct snd_pcm_ops.pointer()'.
Suggested-by: Clemens Ladisch <clemens@ladisch.de>
Fixes: e9148dddc3c7("ALSA: firewire-lib: flush completed packets when reading PCM position")
Cc: <stable@vger.kernel.org> # 4.9+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MCLK for DA7219 does not change in this platform, but is
currently being configured everytime as part of the platform_clock
event handler for DAPM. The upshot of this is that we have
unnecessary calls to this function, and it also means that if
a stream hasn't yet been started, DA7219 driver does not have the
correct MCLK rates programmed and so the HP detection feature does
not operate as expected.
This patch rectifies this issue by moving the sysclk call to
codec_init function so it's only called once at initialisation.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when HP detection procedure runs for certain MCLK
frequencies, when PLL is bypassed, the procedure will incorrectly
report Lineout instead of Headphones due to timing incosistencies.
To avoid this problem, the PLL is temporarily enabled (if currently
bypassed and MCLK present) to provide consistent timings for the
procedure, regardless of MCLK frequency.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds the deepbuffer device which can be opened with a bigger
buffer size. The application can disable interrupts and sleep for longer
duration.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA buffer size for gateway copier will be calculated based on:
For host DMA copier:
Input buffer size (ibs) for output direction (playback)
Output buffer size (obs) for input direction (capture)
For link DMA copier:
IBS for input direction (capture)
OBS for output direction (playback)
Update the driver to use the above.
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With this patch, the dma buffer size is fetched from topology binary. This
buffer size is applicable for gateway copier modules.
Now that we can configure DSP dma buffer size, the device can support deep
buffer playback. DSP fetches large buffer and can result fewer wakes,
which helps in power reduction.
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use a specific flag for SAI and I2S interfaces,
instead of common flag.
Signed-off-by: olivier moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable master clock by default, and activate
it only when requested.
Signed-off-by: olivier moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow peripheral clock enable/disable on regmap accesses.
Signed-off-by: olivier moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple card driver is supporting widgets on DT,
other simple/audio card drivers will support it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_of_parse_widgets
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple card drivers are parsing widgets on each own driver
(only simple-card at this point, but will be supported on all drivers)
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_of_parse_widgets for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ak4613 accepts all range of Sampling Rate, but it depends on
inputed master clock. This patch adds hw constraint rule for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch to use managed variant of acpi_dev_add_driver_gpios() to simplify
error path and fix potentially wrong assignment if ->probe() fails.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
GPIO ACPI mapping table is defined on platform basis. Codec driver
shouldn't have known what platform is using it.
Make codec driver more generic by moving platform code to where it
belongs.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2C devices are enumerated by IDs, and not by instances.
Make it clear by using proper module device table for ACPI case.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel SST driver allocates lots of pages at suspend for saving the
firmware states, and this may occasionally lead to the allocation
error due to the high order, ending up with the suspend failure.
Use kvzalloc() so that it can fall back to vmalloc() gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas sound driver is assuming that all Sampling rate and
channles are possible to use, but these are depends on inputed clock
and SSI connection situation.
For example, if it is using 1 SSI, enabled TDM mode and has 12288000
input clock, 2ch output can support until 192000Hz, but 6ch output can
support until 64000Hz, 8ch can support 48000Hz.
To control these situation correctly, it needs to support
hw_constraints / refine feature.
To support such feature, this patch adds new
rsnd_soc_hw_rule/constraint() which adds hw rule of Channel and
Sampling Rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas sound driver is assuming that all Sampling rate and
channles are possible to use, but these are depends on inputed clock
and SSI connection situation.
For example, if it is using 1 SSI, enabled TDM mode and has 12288000
input clock, 2ch output can support until 192000Hz, but 6ch output can
support until 64000Hz, 8ch can support 48000Hz.
To control these situation correctly, it needs to support
hw_constraints / refine feature.
To support such feature, this patch adds new rsnd_ssi_clk_query().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas sound driver is assuming that all Sampling rate and
channles are possible to use, but these are depends on inputed clock
and SSI connection situation.
For example, if it is using 1 SSI, enabled TDM mode and has 12288000
input clock, 2ch output can support until 192000Hz, but 6ch output can
support until 64000Hz, 8ch can support 48000Hz.
To control these situation correctly, it needs to support
hw_constraints / refine feature.
To support such feature, it needs SSI clock query feature, and it needs
ADG clock query feature. Current ADG has rsnd_adg_ssi_clk_try_start()
and it is doing similar things, but it try to setup ADG register in
same time. This is not needed.
This patch adds new rsnd_adg_clk_query() and separates query feature
and register setting feature in adg.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas sound driver has slots and slots_num in
struct rsnd_dai, but these are very un-understandable naming
(It had named from TDM slots).
In this driver, the "slots" means total usable channels, and
"stot_num" means SSI lane number if Multi SSI was used.
To more understandable code, this patch renames "slots" to
"max_channels", and "slots_num" to "ssi_lane", and replaces related
functions name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
routing on DT, but, doesn't use same function for it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_of_parse_routing
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
routing on DT, but, doesn't use same function for it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_of_parse_routing
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
routing on DT, but, doesn't use same function for it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_of_parse_routing
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple card drivers are parsing routing on each own driver.
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_of_parse_routing for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
convert-rate/convert-channels on DT, but, doesn't use same function
for it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
convert-rate/convert-channels on DT, but, doesn't use same function
for it.
Encapsulation is one of simple card util's purpose.
Let's use asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
