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To make it clearer which legacy substream corresponds to which UMP
group, fill the subname field of each substream object with the group
number and the endpoint name, e.g. "Group 1 (My Device)".
Ideally speaking, we should have some better link information to the
derived UMP, but it's another feature extension.
Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support")
Link: https://lore.kernel.org/r/20230824075108.29958-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy rawmidi devices are the shadows of the main UMP devices,
hence it's better to initialize them after all UMP Endpoints are
parsed. Then, at the moment the legacy rawmidi is created, we already
know the static flag or the proper EP name string, and we can fill
those information at UMP core side instead of fiddling the attributes
at a later point.
Fixes: ec362b63c4b5 ("ALSA: usb-audio: Enable the legacy raw MIDI support")
Link: https://lore.kernel.org/r/20230824075108.29958-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before committing 79597c8bf64c, *rac97 always be NULL if there is
an error. When error happens, make sure *rac97 is NULL is safer.
For examble, in snd_vortex_mixer():
err = snd_ac97_mixer(pbus, &ac97, &vortex->codec);
vortex->isquad = ((vortex->codec == NULL) ?
0 : (vortex->codec->ext_id&0x80));
If error happened but vortex->codec isn't NULL, this may cause some
problems.
Move the judgement order to be clearer and better.
Fixes: 79597c8bf64c ("ALSA: ac97: Fix possible NULL dereference in snd_ac97_mixer")
Suggested-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Su Hui <suhui@nfschina.com>
Link: https://lore.kernel.org/r/20230823025212.1000961-1-suhui@nfschina.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add pause push/release support to the virtual PCM test driver. Add
'suspend' boolean field to the pcmtst_buf_iter structure, so we can
pause the timer without shutting it down. Update the trigger callback
handler correspondingly. Extract buffer initialization to the
'reset_buf_iterator' function since it is used in multiple places now.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20230822150541.8450-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CS35L41 HDA driver requires ACPI to contain correct _DSD properties to
correctly configure the device. Whilst the HP Zbook Fury 17 G9 contains
valid _DSD properties, the boost type has been configured incorrectly
in the _DSD for this laptop. We can override these properties to fix
the boost type.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230823143956.755758-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A relatively large but generally not super urgent set of fixes for ASoC,
including some quirks and a MAINTAINERS update. There's also an update
to cs35l56 to change the firmware ABI, there are no current shipping
systems which use the current interface and the sooner we get the new
interface in the less likely it is that something will start.
It'd be nice if these landed for v6.5 but not the end of the world if
they wait till v6.6.
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Merge tag 'asoc-fix-v6.5-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.5
A relatively large but generally not super urgent set of fixes for ASoC,
including some quirks and a MAINTAINERS update. There's also an update
to cs35l56 to change the firmware ABI, there are no current shipping
systems which use the current interface and the sooner we get the new
interface in the less likely it is that something will start.
It'd be nice if these landed for v6.5 but not the end of the world if
they wait till v6.6.
clang warns (or errors with CONFIG_WERROR=y):
sound/soc/codecs/cs42l43.c:1371:2: error: variable 'ret' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized]
1371 | default:
| ^~~~~~~
sound/soc/codecs/cs42l43.c:1377:9: note: uninitialized use occurs here
1377 | return ret;
| ^~~
sound/soc/codecs/cs42l43.c:1349:9: note: initialize the variable 'ret' to silence this warning
1349 | int ret;
| ^
| = 0
1 error generated.
Initialize ret to 0 in the default case, as there was nothing to do for
other event types.
Closes: https://github.com/ClangBuiltLinux/linux/issues/1922
Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20230823-cs42l43_pll_ev-init-ret-v1-1-5836f1ad5dad@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Xingyu Wu <xingyu.wu@starfivetech.com>:
This patch series adds I2S support for the StarFive JH7110 RISC-V
SoC based on Designware I2S controller. There has three I2S channels
(RX/TX0/TX1) on the JH7110 SoC, one of which is for record(RX) and
two for playback(TX).
The first patch adds support for the StarFive JH7110 SoC in the
Designware I2S bindings.
The second patch adds the ops to get data from platform bus in the
I2S driver.
The third patch adds support for the StarFive JH7110 SoC in
the Designware I2S driver.
The fourth patch fixes the name of I2STX1 pinmux.
The last patch adds device node of I2S RX/TX0/TX1 in JH7110 dts.
This patch series is based on Linux-next(20230818) which is merge
clock, syscon and dma nodes for the StarFive JH7110 SoC.
The series has been tested and works normally on the VisionFive 2
board by plugging an audio expansion board.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Renesas Sound has ADG for clock control. Basically it needs
accurately divisible external input clock. But sometimes
sometimes it doesn't have to be accurate for some reason.
We can use ADG clk_i for such case. It came from CPG as
very high rate clock, but is not accurately divisible for
48kHz/44.1kHz rate, but enough for approximate rate.
This patch set support such use case.
We don't need to have "format" property on DT any more if
CPU/Codec driver has .auto_selectable_formats settings
on snd_soc_dai_ops. The sample dtsi doesn't have it.
To avoid user confusion, this patch indicates it on comment.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87edjuzk2p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Like a few other drivers, YMFPCI driver needs to clean up with
snd_card_free() call at an error path of the probe; otherwise the
other devres resources are released before the card and it results in
the UAF.
This patch uses the helper for handling the probe error gracefully.
Fixes: f33fc1576757 ("ALSA: ymfpci: Create card with device-managed snd_devm_card_new()")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Takashi Yano <takashi.yano@nifty.ne.jp>
Closes: https://lore.kernel.org/r/20230823135846.1812-1-takashi.yano@nifty.ne.jp
Link: https://lore.kernel.org/r/20230823161625.5807-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add StarFive JH7110(TX0/TX1/RX channels) SoC support in the
designware I2S driver and a flag to check if it is on the JH7110 SoC.
These channels need to enable clocks, resets and syscon register on the
JH7110 SoC. So add init ops in platform data for the JH7110 SoC to do this.
Their resets should be deassert before changing the parent of clocks so
these are done in the init ops of platform data.
The I2S controllers use DMA controller by platform data on the JH7110
and their settings about snd_dmaengine_dai_dma_data() should be added
in the dw_configure_dai_by_pd(). And use dmaengine PCM registration if
these do not have IRQ on the JH7110 SoC.
Signed-off-by: Xingyu Wu <xingyu.wu@starfivetech.com>
Link: https://lore.kernel.org/r/20230821144151.207339-4-xingyu.wu@starfivetech.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c will configure BRGCKR/BRRA/BRRB to output clock
when it start sound. OTAH, rsnd_adg_clk_enable() will enables
clk_a/b/c when driver was probed.
But it is strange, these should be set in the same time.
This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h6oqzlei.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg has default division for BRRA/BRRB, but it was created at
very beginning of the driver implementation, and is now an unnecessary
settings.
Because it has this default division, unexpected clockout might
be selected. For example if it requests only 44.1kHz base clockout,
unrequested 48kHz base clockout also will be selected.
This patch remove default division of clock out
Reported-by: Vincenzo De Michele <vincenzo.michele@davinci.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87il96zlep.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c doesn't assume that requested clock out divide condition
doesn't match. In such case, it will indicate strange message, and will
register NULL clock, etc. It is just a DT setting miss, but is
confusable. This patch check all conditions for it.
Reported-by: Vincenzo De Michele <vincenzo.michele@davinci.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lee2zlf7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Basically Renesas sound ADG is assuming that it has accurately
divisible input clock. But sometimes / some board might not have it.
The clk_i from CPG is used for such case. It can't calculate accurate
division, but can be used as approximate rate.
This patch enable clk_i for such case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Adnan Ali <adnan.ali@bp.renesas.com>
Tested-by: Vincenzo De Michele <vincenzo.michele@davinci.de>
Tested-by: Patrick Keil <patrick.keil@conti-engineering.com>
Link: https://lore.kernel.org/r/87msyizlfd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the hardware may be designed as a single-ended input, the headset mic
record only supports single-ended input on the left side. This patch
will enhance microphone recording performance for single-end.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Link: https://lore.kernel.org/r/20230823071244.1861487-2-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This codec was used by the deleted S3C board
sound/soc/samsung/s3c24xx_uda134x.c.
Fixes: 503278c12701 ("ASoC: samsung: remove unused drivers")
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230822-delete-l3-v2-1-b3ffc07348af@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
DSP_SW_INTR_STAT_OFFSET is a common interrupt register which will be
accessed by both ACP firmware and driver. This register contains register
bits corresponds to host to dsp interrupts and vice versa.
when dsp to host interrupt is reported, only clear dsp to host
interrupt bit in DSP_SW_INTR_STAT_OFFSET.
Fixes: 2e7c6652f9b8 ("ASoC: SOF: amd: Fix for handling spurious interrupts from DSP")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-7-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to scratch memory persistence, Once the DSP panic is reported, need to
clear the panic mask after handling DSP panic. Otherwise, It results in DSP
panic on next reboot.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-6-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Few AMD platforms require ACP ACLK as clock source.
Add conditional check for clock mux selection register for
switching between internal clock and ACP ACLK.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-5-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add module parameter for firmware debug. If firmware debug
flag is enabled, clear the fusion stall bit which is required
for enabling firmware debugging through JTAG.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously ACP SOF firmware used to enable the ACP external
global interrupt register.
This will restrict to report ACP host interrupts only after
firmware loading is successful.
This register needs to be set from host driver to handle
other ACP interrupts(SoundWire Interrupts) before loading
the ACP firmware.
Add field for external interrupt enable register in acp descriptor
structure and enable the external interrupt enable register.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During initial development time for RN platform, when SHA
dma gets completed, SHA DMA engine used to raise the ACP interrupt.
In ACP interrupt handler, SHA DMA interrupt got handled.
Currently SHA DMA compleition is verified by checking
transfer count using read poll time out logic.
Remove unused SHA dma interrupt handling code.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230823073340.2829821-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current analog gain TLV seems to have completely incorrect values in
it. The gain starts at 0.5dB, proceeds in 1dB steps, and has no mute
value, correct the control to match.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230823085308.753572-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC merges DAI call backs into .ops.
This patch merge these into one.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202308152047.psX1QNDh-lkp@intel.com/
Cc: Randy Dunlap <rdunlap@infradead.org>
Fixes: 446b31e89493 ("ASoC: soc-dai.h: remove unused call back functions")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Randy Dunlap <rdunlap@infradead.org> # build-tested
Link: https://lore.kernel.org/r/87a5ujubj0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new module parameter ipc4_ignore_cpc which can be used to force the
kernel to ignore the queried CPC value for all firmware modules and use 0
instead.
The CPC lookup is still done to report missing configurations and the
debug print is going to be different to be explicit that the CPC is ignored
and what was the value we would have used.
The CPC value is sent to the firmware with the MOD_INIT_INSTANCE message
and it is used by the firmware as a parameter for clock scaling.
The flag is intended to be used only when there is a need to validate the
firmware behavior regarding to clock scaling since the 0 CPC value will
force the DSP to run in full speed, disabling the scaling and provides
additional counter point to rule out clock management related issues.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@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: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20230822065419.24374-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The SRC component in a pipeline provides flexibility on the sampling
rate which was not handled previously. This series will improve the
kernel side with the needed logic to be able to deal with the SRC type
of components in pipelines.
There have been reports of USB-audio driver spewing errors at the
probe time on a few devices like Jabra and Logitech. The suggested
fix there couldn't be applied as is, unfortunately, because it'll
likely break other devices.
But, the patch suggested an interesting point: looking at the current
init code in stream.c, one may notice that it does initialize
differently from the device setup in endpoint.c. Namely, for UAC1, we
should call snd_usb_init_pitch() and snd_usb_init_sample_rate() after
setting the interface, while the init sequence at parsing calls them
before setting the interface blindly.
This patch changes the init sequence at parsing for UAC1 (and other
devices that need a similar behavior) to be aligned with the rest of
the code, setting the interface at first. And, this fixes the
long-standing problems on a few UAC1 devices like Jabra / Logitech,
as reported, too.
Reported-and-tested-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Closes: https://lore.kernel.org/r/202bbbc0f51522e8545783c4c5577d12a8e2d56d.camel@infinera.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230821111857.28926-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the copier has only output valid_bits across all its output
formats, the reference for selecting the output format must be set that
instead of the valid_bits from the selected input format.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230821113629.5017-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When we walk the list of connected widgets from the source to the sink
to prepare all widgets, the pipeline_params must be modified to reflect
the output audio format at each widget. But, the copier only modifies
the sample format in the pipeline_params. So, fix it to also modify the
rate and channels.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230821113629.5017-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For playback, the SRC sink rate must be configured based on the requested
output format which is restricted to only handle DAI's that support a
single audio format for now. For capture, the SRC module should convert
the rate to match the rate requested by the PCM hw_params.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230821113629.5017-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify the pipeline_params based on the SRC output format and set the
sink_rate in the IPC data.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230821113629.5017-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a quirk entry. Fix it.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Fixes: 3babae915f4c ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Link: https://lore.kernel.org/r/20230821080003.16678-1-colin.i.king@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use "select" to ensure that the required kconfig symbols are set
as expected.
Drop HOSTAUDIO since it is now equivalent to UML_SOUND.
Set CONFIG_SOUND=m in ARCH=um defconfig files to maintain the
status quo of the default configs.
Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a
kconfig warning for unmet dependencies. (This was not an issue when
SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig
builds and didn't find any issues.)
This fixes build errors when CONFIG_SOUND is not set:
ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module':
hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer'
ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp'
ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module':
hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp'
ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer'
ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp'
and this kconfig warning:
WARNING: unmet direct dependencies detected for SOUND
Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2")
Fixes: d886e87cb82b ("sound: make OSS sound core optional")
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Reported-by: kernel test robot <lkp@intel.com>
Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com
Cc: Richard Weinberger <richard@nod.at>
Cc: Anton Ivanov <anton.ivanov@cambridgegreys.com>
Cc: Johannes Berg <johannes@sipsolutions.net>
Cc: linux-um@lists.infradead.org
Cc: Tejun Heo <tj@kernel.org>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Masahiro Yamada <masahiroy@kernel.org>
Cc: Nathan Chancellor <nathan@kernel.org>
Cc: Nick Desaulniers <ndesaulniers@google.com>
Cc: Nicolas Schier <nicolas@fjasle.eu>
Cc: linux-kbuild@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Reviewed-by: Masahiro Yamada <masahiroy@kernel.org>
Signed-off-by: Richard Weinberger <richard@nod.at>
The CS42L43 is an audio CODEC with integrated MIPI SoundWire interface
(Version 1.2.1 compliant), I2C, SPI, and I2S/TDM interfaces designed
for portable applications. It provides a high dynamic range, stereo
DAC for headphone output, two integrated Class D amplifiers for
loudspeakers, and two ADCs for wired headset microphone input or
stereo line input. PDM inputs are provided for digital microphones.
The ASoC component provides the majority of the functionality of the
device, all the audio functions.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230804104602.395892-7-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
