[ Upstream commit aa85822c61 ]
PC speaker works well on this platform in BIOS and in Linux until sound
card drivers are loaded. Then it stops working.
There seems to be a beep generator node at 0x1a in this CODEC
(ALC269_TYPE_ALC215) but it seems to be only connected to capture mixers
at nodes 0x22 and 0x23.
If I unmute the mixer input for 0x1a at node 0x23 and start recording
from its "ALC285 Analog" capture device I can clearly hear beeps in that
recording.
So the beep generator is indeed working properly, however I wasn't able to
figure out any way to connect it to speakers.
However, the bits in the "Passthrough Control" register (0x36) seems to
work at least partially: by zeroing "B" and "h" and setting "S" I can at
least make the PIT PC speaker output appear either in this laptop speakers
or headphones (depending on whether they are connected or not).
There are some caveats, however:
* If the CODEC gets runtime-suspended the beeps stop so it needs HDA beep
device for keeping it awake during beeping.
* If the beep generator node is generating any beep the PC beep passthrough
seems to be temporarily inhibited, so the HDA beep device has to be
prevented from using the actual beep generator node - but the beep device
is still necessary due to the previous point.
* In contrast with other platforms here beep amplification has to be
disabled otherwise the beeps output are WAY louder than they were on pure
BIOS setup.
Unless someone (from Realtek probably) knows how to make the beep generator
node output appear in speakers / headphones using PC beep passthrough seems
to be the only way to make PC speaker beeping actually work on this
platform.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Acked-by: kailang@realtek.com
Link: https://patch.msgid.link/7461f695b4daed80f2fc4b1463ead47f04f9ad05.1739741254.git.mail@maciej.szmigiero.name
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0eba2a7e85 ]
This reverts commit 9bdd10d57a ("ASoC: ops: Shift tested values in
snd_soc_put_volsw() by +min"), and makes some additional related
updates.
There are two ways the platform_max could be interpreted; the maximum
register value, or the maximum value the control can be set to. The
patch moved from treating the value as a control value to a register
one. When the patch was applied it was technically correct as
snd_soc_limit_volume() also used the register interpretation. However,
even then most of the other usages treated platform_max as a
control value, and snd_soc_limit_volume() has since been updated to
also do so in commit fb9ad24485 ("ASoC: ops: add correct range
check for limiting volume"). That patch however, missed updating
snd_soc_put_volsw() back to the control interpretation, and fixing
snd_soc_info_volsw_range(). The control interpretation makes more
sense as limiting is typically done from the machine driver, so it is
appropriate to use the customer facing representation rather than the
internal codec representation. Update all the code to consistently use
this interpretation of platform_max.
Finally, also add some comments to the soc_mixer_control struct to
hopefully avoid further patches switching between the two approaches.
Fixes: fb9ad24485 ("ASoC: ops: add correct range check for limiting volume")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e8343410dd ]
Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ba2de401d3 ]
Pass the PCI SSID of the audio interface through to the machine driver.
This allows the machine driver to use the SSID to uniquely identify the
specific hardware configuration and apply any platform-specific
configuration.
struct snd_sof_pdata is passed around inside the SOF code, but it then
passes configuration information to the machine driver through
struct snd_soc_acpi_mach and struct snd_soc_acpi_mach_params. So SSID
information has been added to both snd_sof_pdata and
snd_soc_acpi_mach_params.
PCI does not define 0x0000 as an invalid value so we can't use zero to
indicate that the struct member was not written. Instead a flag is
included to indicate that a value has been written to the
subsystem_vendor and subsystem_device members.
sof_pci_probe() creates the struct snd_sof_pdata. It is passed a struct
pci_dev so it can fill in the SSID value.
sof_machine_check() finds the appropriate struct snd_soc_acpi_mach. It
copies the SSID information across to the struct snd_soc_acpi_mach_params.
This done before calling any custom set_mach_params() so that it could be
used by the set_mach_params() callback to apply variant params.
The machine driver receives the struct snd_soc_acpi_mach as its
platform_data.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 47f56e38a1 ]
Add members to struct snd_soc_card to store the PCI subsystem ID (SSID)
of the soundcard.
The PCI specification provides two registers to store a vendor-specific
SSID that can be read by drivers to uniquely identify a particular
"soundcard". This is defined in the PCI specification to distinguish
products that use the same silicon (and therefore have the same silicon
ID) so that product-specific differences can be applied.
PCI only defines 0xFFFF as an invalid value. 0x0000 is not defined as
invalid. So the usual pattern of zero-filling the struct and then
assuming a zero value unset will not work. A flag is included to
indicate when the SSID information has been filled in.
Unlike DMI information, which has a free-format entirely up to the vendor,
the PCI SSID has a strictly defined format and a registry of vendor IDs.
It is usual in Windows drivers that the SSID is used as the sole identifier
of the specific end-product and the Windows driver contains tables mapping
that to information about the hardware setup, rather than using ACPI
properties.
This SSID is important information for ASoC components that need to apply
hardware-specific configuration on PCI-based systems.
As the SSID is a generic part of the PCI specification and is treated as
identifying the "soundcard", it is reasonable to include this information
in struct snd_soc_card, instead of components inventing their own custom
ways to pass this information around.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e123036be3 ]
In the BE hw_params configuration, the existing code checks if any of the
existing FEs are prepared, running, paused or suspended - and skips the
configuration in those cases. This allows multiple calls of hw_params
which the ALSA state machine supports.
This check is not handled for the prepare stage, which can lead to the
same BE being prepared multiple times. This patch adds a check similar to
that of the hw_params, with the main difference being that the suspended
state is allowed: the ALSA state machine allows a transition from
suspended to prepared with hw_params skipped.
This problem was detected on Intel IPC4/SoundWire devices, where the BE
dailink .prepare stage is used to configure the SoundWire stream with a
bank switch. Multiple .prepare calls lead to conflicts with the .trigger
operation with IPC4 configurations. This problem was not detected earlier
on Intel devices, HDaudio BE dailinks detect that the link is already
prepared and skip the configuration, and for IPC3 devices there is no BE
trigger.
Link: https://github.com/thesofproject/sof/issues/7596
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d045bceff5 ]
Some motherboards have multiple HDA codecs connected to the serial bus.
The current code may create multiple mixer controls with the almost
identical identification.
The current code use id.device field from the control element structure
to store the codec address to avoid such clashes for multiple codecs.
Unfortunately, the user space do not handle this correctly. For mixer
controls, only name and index are used for the identifiers.
This patch fixes this problem to compose the index using the codec
address as an offset in case, when the control already exists. It is
really unlikely that one codec will create 10 similar controls.
This patch adds new kernel module parameter 'ctl_dev_id' to allow
select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig
option sets the default value.
BugLink: https://github.com/alsa-project/alsa-lib/issues/294
BugLink: https://github.com/alsa-project/alsa-lib/issues/205
Fixes: 54d1740315 ("[ALSA] hda-codec - Fix connection list parsing")
Fixes: 1afe206ab6 ("ALSA: hda - Try to find an empty control index when it's occupied")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230202092013.4066998-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ee0b089d66 ]
When the new style KAE keep-alive implementation is used on compatible
Intel hardware, the clocks are maintained when codec is in D3. The
generic code in hda_cleanup_all_streams() can however interfere with
generation of audio samples in this mode, by setting the stream and
channel ids to zero.
To get full benefit of the keepalive, set the new
no_stream_clean_at_suspend quirk bit on affected Intel hardware. When
this bit is set, stream cleanup is skipped in hda_call_codec_suspend().
Special handling is needed for the case when system goes to suspend. The
stream id programming can be lost in this case. This will also cause
codec->cvt_setups to be out of sync. Handle this by implementing custom
suspend/resume handlers. If keep-alive is active for any converter, set
the quirk flags no_stream_clean_at_suspend and forced_resume. Upon
resume, keepalive programming is restored if needed.
Fixes: 15175a4f2b ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20221209101822.3893675-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b5172e6245 ]
Shifting signed 32-bit value by 31 bits is undefined, so changing
significant bit to unsigned. The UBSAN warning calltrace like below:
UBSAN: shift-out-of-bounds in sound/core/pcm_native.c:2676:21
left shift of 1 by 31 places cannot be represented in type 'int'
...
Call Trace:
<TASK>
dump_stack_lvl+0x8d/0xcf
ubsan_epilogue+0xa/0x44
__ubsan_handle_shift_out_of_bounds+0x1e7/0x208
snd_pcm_open_substream+0x9f0/0xa90
snd_pcm_oss_open.part.26+0x313/0x670
snd_pcm_oss_open+0x30/0x40
soundcore_open+0x18b/0x2e0
chrdev_open+0xe2/0x270
do_dentry_open+0x2f7/0x620
path_openat+0xd66/0xe70
do_filp_open+0xe3/0x170
do_sys_openat2+0x357/0x4a0
do_sys_open+0x87/0xd0
do_syscall_64+0x34/0x80
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Signed-off-by: Baisong Zhong <zhongbaisong@huawei.com>
Link: https://lore.kernel.org/r/20221121110044.3115686-1-zhongbaisong@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
ASoC: Fixes for v6.1
A clutch of small fixes that have come in in the past week, people seem
to have been unusually active for this late in the release cycle. The
most critical one here is the fix to renumber the SOF DAI types in order
to restore ABI compatibility which was broken by the addition of AMD
support.
ASoC: Fixes for v6.1
A relatively large collection of fixes and new platform quirks here,
they're all fairly minor though - the widest possible impact is the fix
to the use of prefixes on regulator names which would have broken any
device that integrates regulators with DAPM and was used in a system
where it had a name prefix assigning to it.
Originally in commit b2ebcf42a4 ("ASoC: SOF: free widgets in
sof_tear_down_pipelines() for static pipelines"), freeing of pipeline
components at suspend was only done with recent FW as there were known
limitations in older firmware versions.
Tests show that if static pipelines are used, i.e. all pipelines are
setup whenever firmware is powered up, the reverse action of freeing all
components at power down, leads to firmware failures with also SOF2.0
and SOF2.1 based firmware.
The problems can be specific to certain topologies with e.g. components
not prepared to be freed at suspend (as this did not happen with older
SOF kernels).
To avoid hitting these problems when kernel is upgraded and used with an
older firmware, bump the firmware requirement to SOF2.2 or newer. If an
older firmware is used, and pipeline is a static one, do not free the
components at suspend. This ensures the suspend flow remains backwards
compatible with older firmware versions. This limitation does not apply
if the product configuration is updated to dynamic pipelines.
The limitation is not linked to firmware ABI, as the interface to free
pipeline components has been available already before ABI3.19. The
problem is in the implementation, so firmware version should be used to
decide whether it is safe to use the newer flow or not. This patch adds
a new SOF_FW_VER() macro to compare SOF firmware release versions.
Link: https://github.com/thesofproject/sof/issues/6475
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221101114913.1292671-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fixes for v6.1
Quite a few fixes here, a lot driver specific, plus some new quirks.
There was a bit of a mess with the runtime PM handling due to some
confusion in the API there which resulted in a number of commits and
reverts but that should all be stable now.
Merge series from Siarhei Volkau <lis8215@gmail.com>:
The patchset fixes:
- Line In path stays powered off during capturing or
bypass to mixer.
- incorrectly represented dB values in alsamixer, et al.
- incorrect represented Capture input selector in alsamixer
in Playback tab.
- wrong control selected as Capture Master
The "convert-xxx" properties only have an effect for DPCM DAI links.
A DAI link is only created as DPCM if the device tree requires it;
part of this involves checking for the use of "convert-xxx" properties.
When the convert-sample-format property was added, the checks got out
of sync. A DAI link that specified only convert-sample-format but did
not pass any of the other DPCM checks would not go into DPCM mode and
the convert-sample-format property would be silently ignored.
Fix this by adding a function to do the "convert-xxx" property checks,
instead of open-coding it in simple-card and audio-graph-card. And add
"convert-sample-format" to the check function so that DAI links using
it will be initialized correctly.
Fixes: 047a05366f ("ASoC: simple-card-utils: Fixup DAI sample format")
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20221019012302.633830-1-aidanmacdonald.0x0@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Updates for v6.1
This has been a very quiet release for the core but quite a busy one for
drivers with a big crop of new drivers and lots of feature additions and
fixes to existing ones:
- A new string helper parse_int_array_user().
- Improvements to the SOF IPC4 code, especially around trace.
- Support for AMD Rembrant DSPs, AMD Pink Sardine ACP 6.2, Apple Silcon
systems, Everest ES8326, Intel Sky Lake and Kaby Lake, MediaTek
MT8186 support, NXP i.MX8ULP DSPs, Qualcomm SC8280XP, SM8250 and SM8450
and Texas Instruments SRC4392
There is a conflict with the conversion of I2C remove functions to void
in the cs42l42 driver which is fairly straightforward to resolve but
should be highlighted to Linus.
In the PCM core and driver code, there are lots place referring to the
current PCM state via runtime->status->state. This patch introduced a
local PCM state in runtime itself and replaces those references with
runtime->state. It has improvements in two aspects:
- The reduction of a indirect access leads to more code optimization
- It avoids a possible (unexpected) modification of the state via mmap
of the status record
The status->state is updated together with runtime->state, so that
user-space can still read the current state via mmap like before,
too.
This patch touches only the ALSA core code. The changes in each
driver will follow in later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220926135558.26580-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Per discussion on the alsa-devel mailing list [1], the legacy PIN to PCM
device mapping is obsolete nowadays. The maximum number of the simultaneously
usable PCM devices is equal to the HDMI codec converters.
Remove the extra PCM devices (beyond the detected converters) and force
the use of the dynamic PCM device allocation. The legacy code is removed.
I believe that all HDMI codecs have the jack sensing feature. Move the check
to the codec probe function and print a warning, if a codec without this
feature is detected.
[1] https://lore.kernel.org/alsa-devel/2f37e0b2-1e82-8c0b-2bbd-1e5038d6ecc6@perex.cz/
Cc: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220922084017.25925-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com>:
This series consists of
1.Make ACP core code generic for newer SOC transition
2.Add support for Rembrandt plaform
3.Adding amd HS functionality to the sof core
4.increase SRAM inbox and outbox size to 1024
Merge series from Martin Povišer <povik+lin@cutebit.org>:
there's a CS42L83 headphone jack codec found in Apple computers (in the
recent 'Apple Silicon' ones as well as in earlier models, one example
[1]). The part isn't publicly documented, but it appears almost
identical to CS42L42, for which we have a driver in kernel. This series
adapts the CS42L42 driver to the new part, and makes one change in
anticipation of a machine driver for the Apple computers.
Patch 1 adds new compatible to the cs42l42 schema.
Patches 2 to 7 are taken from Richard's recent series [2] adding
soundwire support to cs42l42. They are useful refactorings to build on
in the later patches, and also this way our work doesn't diverge.
(I fixed missing free_irq path in cs42l42_init, did
s/Soundwire/SoundWire/ in changelogs, rebased.)
Patch 8 exports some regmap-related symbols from cs42l42.c so they can
be used to create cs42l83 regmap in cs42l83-i2c.c later.
Patch 9 is the cs42l83 support proper.
Patch 10 implements 'set_bclk_ratio' on the cs42l42 core. This will be
called by the upcoming ASoC machine driver for 'Apple Silicon' Macs.
(We have touched on this change to be made in earlier discussion, see
[3] and replies.)
Patch 11 brings cs42l42-i2c.c in sync with cs42l83-i2c.c on
dev_err_probe() usage.
SOF topologies hard-code the MCLK used for SSP connections. That was a
bad idea in hindsight, this information should really come from BIOS
and/or machine driver.
This patch introduces a helper to scan all SSP endpoints connected to
a codec, and all formats to see what MCLK is used. When BIT(0) of the
mdivc offset if set in the SSP blob, MCLK0 is used, and likewise when
BIT(1) is set MCLK1 is used.
The case where both MCLKs are used is possible but has never been seen
in practice so should be treated as an error by the caller.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20220919115350.43104-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rtd has both dai_link pointer (A) and num_cpus/codecs (B).
(A) rtd->dai_link = dai_link;
(B) rtd->num_cpus = dai_link->num_cpus;
(B) rtd->num_codecs = dai_link->num_codecs;
But, we can get num_cpus/codecs (B) via dai_link (A).
This means we don't need to keep num_cpus/codecs on rtd.
This patch removes these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sfkmv9n3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS42L83 part is a headphone jack codec found in recent Apple
machines. It is a publicly undocumented part but as far as can be told
it is identical to CS42L42 except for two points:
* The chip ID is different.
* Of those registers for which we have a default value in the existing
CS42L42 kernel driver, one register (MCLK_CTL) differs in its reset
value on CS42L83.
To address those two points (and only those), add to the CS42L42 driver
a separate CS42L83 front.
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220915094444.11434-10-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Syed Saba Kareem <Syed.SabaKareem@amd.com>:
Pink Sardine platform is new APU series based on acp6.2 design.
This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block
on AMD Pink Sardine APU with DMIC endpoint support.
ALSA: Drop hackish GFP giveaway for CONTINUOUS pages
This is a series of cleanup patches for dropping the current hackish
way of passing the GFP_* flags for CONTINOUS and VMALLOC memory
allocations. There are only three users for this legacy feature, and
all of them seem superfluous. And, if any driver requires the memory
restriction in future, it can now pass the proper device pointer for
specifying the DMA mask.
Link: https://lore.kernel.org/r/20220823115740.14123-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that all users of snd_dma_continuous_data() is gone, let's drop
this ugly (and dangerous) way.
After this commit, SNDRV_DMA_TYPE_CONTINUOUS may take the standard
device pointer instead of the hacked pointer by the macro above, and
the memalloc core refers to the coherent_dma_mask of the given
device like other SNDRV_DMA_TYPE. It's still allowed to pass NULL
there, and in that case, the allocation is performed always in the
normal zone.
For SNDRV_DMA_TYPE_VMALLOC, the device pointer is simply ignored.
Link: https://lore.kernel.org/r/20220823115740.14123-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From current design in sof_machine_check and snd_sof_new_platform_drv,
the SOF can only support ACPI type machine.
1. In sof_machine_check if there is no ACPI machine exist, the function
will return -ENODEV directly, that's we don't expected if we do not
base on ACPI machine.
2. In snd_sof_new_platform_drv the component driver need a driver name
to do ignore_machine, currently the driver name is obtained from
machine->drv_name, and the type of machine is snd_soc_acpi_mach.
So we add a new function named sof_of_machine_select that we can pass
sof_machine_check and obtain info required by snd_sof_new_platform_drv.
Signed-off-by: Chunxu Li <chunxu.li@mediatek.com>
Link: https://lore.kernel.org/r/20220805070449.6611-2-chunxu.li@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai
Maciej S. Szmigiero
Charles Keepax
Jai Luthra
Richard Fitzgerald
Cezary Rojewski
Ranjani Sridharan
syed saba kareem
Lucas Tanure
Jaroslav Kysela
Kai Vehmanen
Baisong Zhong
Pierre-Louis Bossart
Maciej S. Szmigiero
Mark Brown
Aidan MacDonald
Amadeusz Sławiński
Gustavo A. R. Silva
V sujith kumar Reddy
Kuninori Morimoto
Martin Povišer
Peter Ujfalusi
Chunxu Li
Srinivas Kandagatla