Merge series from Herve Codina <herve.codina@bootlin.com>:
The Infineon PEB2466 codec is a programmable DSP-based four channels
codec with filters capabilities.
It also provides signals as GPIOs.
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
The CS42L42 has a SoundWire interface for control and audio. This
chain of patches adds support for this.
Patches #1 .. #5 split out various changes to the existing code that
are needed for adding Soundwire. These are mostly around clocking and
supporting the separate probe and enumeration stages in SoundWire.
Patches #6 .. #8 actually adds the SoundWire handling.
Merge series from wangweidong.a@awinic.com:
The Awinic AW88395 is an I2S/TDM input, high efficiency
digital Smart K audio amplifier with an integrated 10.25V
smart boost converter.
Add a DT schema for describing Awinic AW88395 audio amplifiers. They are
controlled using I2C
This adds support for using CS42L42 as a SoundWire device.
SoundWire-specifics are kept separate from the I2S implementation as
much as possible, aiming to limit the risk of breaking the I2C+I2S
support.
There are some important differences in the silicon behaviour between
I2S and SoundWire mode that are reflected in the implementation:
- ASP (I2S) most not be used in SoundWire mode because the two interfaces
share pins.
- The SoundWire capture (record) port only supports 1 channel. It does
not have left-to-right duplication like the ASP.
- DP2 can only be prepared if the HP has powered-up. DP1 can only be
prepared if the ADC has powered-up. (This ordering restriction does
not exist for ASPs.) The SoundWire core port-prepare step is
triggered by the DAI-link prepare(). This happens before the
codec DAI prepare() or the DAPM sequence so these cannot be used
to enable HP/ADC. Instead the HP/ADC enable/disable are done during
the port_prep callback.
- The SRCs are an integral part of the audio chain but in silicon their
power control is linked to the ASP. There is no equivalent power link
to SoundWire DPs so the driver must take "manual" control of SRC power.
- The SoundWire control registers occupy the lower part of the SoundWire
address space so cs42l42 registers are offset by 0x8000 (non-paged) in
SoundWire mode.
- Register addresses are 8-bit paged in I2C mode but 16-bit unpaged in
SoundWire.
- Special procedures are needed on register read/writes to (a) ensure
that the previous internal bus transaction has completed, and
(b) handle delayed read results, when the read value could not be
returned within the SoundWire read command.
There are also some differences in driver implementation between I2S
and SoundWire operation:
- CS42L42 I2S does not runtime_suspend, but runtime_suspend/resume support
has been added into the driver in SoundWire mode as the most convenient
way to power-up the bus manager and to handle the unattach_request
condition, though the CS42L42 chip does not itself suspend or resume.
- Intel SoundWire host controllers have a low-power clock-stop mode that
requires resetting all peripherals when resuming. This means that the
interrupt registers will be reset in between the interrupt being
generated and the interrupt being handled, and since the interrupt
status is debounced, these values may not be accurate immediately,
and may cause spurious unplug events before settling.
- As in I2S mode, the PLL is only used while audio is active because
of clocking quirks in the silicon. For SoundWire the cs42l42_pll_config()
is deferred until the DAI prepare(), to allow the cs42l42_bus_config()
callback to set the SCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-7-sbinding@opensource.cirrus.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>
Add generic ASoC equivalent of ALSA HD-Audio codec. This codec is
designed to follow HDA_DEV_LEGACY convention. Driver wrapps existing
hda_codec.c handlers to prevent code duplication within the newly added
code. Number of DAIs created is dependent on capabilities exposed by the
codec itself. Because of this, single solution can be applied to support
every single HD-Audio codec type.
At the same time, through the ASoC topology, platform drivers may limit
the number of endpoints available to the userspace as codec driver
exposes BE DAIs only.
Both hda_codec_probe() and hda_codec_remove() declare their expectations
on device's usage_count and suspended-status. This is to catch any
unexpected behavior as PM-related code for HD-Audio has been changing
quite a bit throughout the years.
In order for codec DAI list to reflect its actual PCM capabilities, PCMs
need to be built and that can only happen once codec device is
constructed. To do that, a valid component->card->snd_card pointer is
needed. Said pointer will be provided by the framework once all card
components are accounted for and their probing can begin. Usage of
"binder" BE DAI solves the problem - codec can be listed as one of
HD-Audio card components without declaring any actual BE DAIs
statically.
Relation with hdac_hda:
Addition of parallel solution is motivated by behavioral differences
between hdac_hda.c and its legacy equivalent found in sound/pci/hda
e.g.: lack of dynamic, based on codec capabilities, resource allocation
and high cost of removing such differences on actively used targets.
Major goal of codec driver presented here is to follow HD-Audio legacy
behavior in 1:1 fashion by becoming a wrapper. Doing so increases code
coverage of the legacy code and reduces the maintenance cost for both
solutions.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220511162403.3987658-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series of patches adds support for Analog Devices MAX98396
mono amplifier with IV sense. The device provides a PCM interface
for audio data and a standard I2C interface for control data
communication. This driver also supports MAX98397 which is
a variant of MAX98396 with wide input supply range.
Signed-off-by: Ryan Lee <ryan.lee.analog@gmail.com>
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20220423021558.1773598-1-ryan.lee.analog@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Awinic AW8738 is a simple audio amplifier using a single GPIO.
The main difference to simple-amplifier is that there is a "one-wire
pulse control" that allows configuring the amplifier to one of a few
pre-defined modes. This can be used to configure the speaker-guard
function (primarily the power limit for the amplifier).
Add a simple driver that allows setting it up in the device tree
with a specified mode number.
Signed-off-by: Jonathan Albrieux <jonathan.albrieux@gmail.com>
Co-developed-by: Stephan Gerhold <stephan@gerhold.net>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20220304102452.26856-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
To support CS35L41 in HDA systems the HDA driver
for CS35L41 would have to duplicate some functions
that already exist on ASoC driver
So instead of duplicate the code, use the new lib
source as a shared resource for both ASoC and HDA
Also, change the way CONFIG_SND_SOC_CS35L41 is
selected, as reported by Intel Kernel test robot,
it is possible to build SND_SOC_CS35L41_SPI/I2C
without the main driver, which would lead to build
failures.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20211217115708.882525-2-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This drops the rt9210 support due to a race with a new version being
sent out for some incremental changes.
Signed-off-by: Mark Brown <broonie@kernel.org>
Vinod writes:
soundwire updates for 5.15-rc1
- Core has updates to support SoundWire mockup device (includes tag from
asoc), improved error handling and slave status.
- Drivers has update on Intel driver for new quriks and better handling of
errors and suspend routines
* tag 'soundwire-5.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire:
soundwire: cadence: do not extend reset delay
soundwire: intel: conditionally exit clock stop mode on system suspend
soundwire: intel: skip suspend/resume/wake when link was not started
soundwire: intel: fix potential race condition during power down
soundwire: cadence: override PDI configurations to create loopback
soundwire: cadence: add debugfs interface for PDI loopbacks
soundwire: stream: don't program mockup device ports
soundwire: bus: squelch error returned by mockup devices
soundwire: add flag to ignore all command/control for mockup devices
soundwire: stream: don't abort bank switch on Command_Ignored/-ENODATA
soundwire: cadence: add paranoid check on self-clearing bits
soundwire: dmi-quirks: add quirk for Intel 'Bishop County' NUC M15
soundwire: bus: update Slave status in sdw_clear_slave_status
soundwire: cadence: Remove ret variable from sdw_cdns_irq()
soundwire: bus: filter out more -EDATA errors on clock stop
soundwire: dmi-quirks: add ull suffix for SoundWire _ADR values
ASoC: Intel: boards: sof_sdw: add SoundWire mockup codecs for tests
ASoC: soc-acpi: tgl: add table for SoundWire mockup devices
ASoC: soc-acpi: cnl: add table for SoundWire mockup devices
ASoC: codecs: add SoundWire mockup device support
With SND_SOC_ALL_CODECS=y and SND_SOC_WCD938X_SDW=m, there is a link
error from a reverse dependency, since the built-in codec driver calls
into the modular soundwire back-end:
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_free':
wcd938x.c:(.text+0x2c0): undefined reference to `wcd938x_sdw_free'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_hw_params':
wcd938x.c:(.text+0x2f6): undefined reference to `wcd938x_sdw_hw_params'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_set_sdw_stream':
wcd938x.c:(.text+0x332): undefined reference to `wcd938x_sdw_set_sdw_stream'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_tx_swr_ctrl':
wcd938x.c:(.text+0x23de): undefined reference to `wcd938x_swr_get_current_bank'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_bind':
wcd938x.c:(.text+0x2579): undefined reference to `wcd938x_sdw_device_get'
x86_64-linux-ld: wcd938x.c:(.text+0x25a1): undefined reference to `wcd938x_sdw_device_get'
x86_64-linux-ld: wcd938x.c:(.text+0x262a): undefined reference to `__devm_regmap_init_sdw'
Work around this using two small hacks: An added Kconfig dependency
prevents the main driver from being built-in when soundwire support
itself is a loadable module to allow calling devm_regmap_init_sdw(),
and a Makefile trick links the wcd938x-sdw backend as built-in
if needed to solve the dependency between the two modules.
Fixes: 0454422288 ("ASoC: codecs: wcd938x: add audio routing and Kconfig")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210721150510.1837221-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
To test the host controller and bus management code, it is currently
required to have a physical SoundWire peripheral attached to the
bus. To help with pre-silicon or early hardware enablement, it would
be very useful to have a SoundWire 'mockup' device that is exposed in
platform firmware but does not drive any signal on the bus.
This is different to the existing ASoC 'dummy' codec uses for I2S/TDM,
the SoundWire spec makes it clear that a device that is not attached
to the bus is not permitted to interact with the bus, be it for
command/control or data.
This patch exposes a 'mockup' device, with a minimalist driver, with 4
partID values reserved by Intel for such test configurations. The
mockup device exposes one full-duplex DAI based on 2 ports (DP1 for
playback and DP8 for capture). The capture data port is just virtual,
such a mockup device is prevented by the SoundWire specification from
presenting any data generated by a Source port without being Attached.
All the callbacks exposed by the SoundWire Slave interface are
populated, even if they just return immediately. This is intentional
to describe what a minimal codec driver should do and implement and
help new codec vendors provide support for their devices.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20210714032209.11284-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the Rockchip rk817 audio codec integrated into the
rk817 PMIC. This is based on the sources provided by Rockchip from
their BSP kernel.
Signed-off-by: Chris Morgan <macromorgan@hotmail.com>
Tested-by: Maciej Matuszczyk <maccraft123mc@gmail.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
NXP's TFA98xx (now part of Goodix) are fairly popular speaker amplifiers
used in many smartphones and tablets. Most of them are sold as "smart
amplifiers" with built-in "CoolFlux DSP" that is used for volume control,
plus a "sophisticated speaker-boost and protection algorithm".
Unfortunately, they are also almost entirely undocumented. The short
datasheets (e.g. [1] for TFA9897) describe the available features,
but do not provide any information about the registers or how to use
the "CoolFlux DSP".
The amplifiers are most often configured through proprietary userspace
libraries. There are also some (rather complex) kernel drivers (e.g. [2])
but even those rely on obscure firmware blobs for configuration (so-called
"containers"). They seem to contain different "profiles" with tuned speaker
settings, sample rates and volume steps (which would be better exposed
as separate ALSA mixers).
The format of the firmware files seems to have changed a lot over the time,
so it's not even possible to simply re-use the firmware originally provided
by the vendor.
Overall, it seems close to impossible to develop a proper mainline driver
for these amplifiers that could make proper use of the built-in DSP.
This commit implements a compromise: At least the TFA1 family of the
TFA98xx amplifiers (usually called TFA989x) provide a way to *bypass*
the DSP using a special register sequence. The register sequence can be
found in similar variations in the kernel drivers from lots of vendors
e.g. in [3] and was probably mainly used for factory testing.
With the DSP bypassed, the amplifier acts mostly like a dumb standard
speaker amplifier, without (hardware) volume control. However, the setup
is much simpler and it works without any obscure firmware.
This driver implements the DSP bypass combined with chip-specific
initialization sequences adapted from [2]. Only TFA9895 is supported in
this initial commit. Except for the lack of volume control I can not hear
any difference with or without the DSP, it works just fine.
This driver allows the speaker to work on mainline Linux running on the
Samsung Galaxy A3/A5 (2015) [TFA9895] and Alcatel Idol 3 [TFA9897].
TFA9897 support will be added in separate patch set later.
[1]: https://product.goodix.com/en/docview/TFA9897%20SDS_Rev.3.1?objectId=47&objectType=document&version=78
[2]: https://source.codeaurora.org/external/mas/tfa98xx
[3]: 57b5050e34/sound/soc/codecs/tfa98xx.c (L1422-L1462)
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20210513104129.36583-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert the arizona extcon driver into a helper library for direct use
from the arizona codec-drivers, rather then being bound to a separate
MFD cell.
Note the probe (and remove) sequence is split into 2 parts:
1. The arizona_jack_codec_dev_probe() function inits a bunch of
jack-detect specific variables in struct arizona_priv and tries to get
a number of resources where getting them may fail with -EPROBE_DEFER.
2. Then once the machine driver has create a snd_sock_jack through
snd_soc_card_jack_new() it calls snd_soc_component_set_jack() on
the codec component, which will call the new arizona_jack_set_jack(),
which sets up jack-detection and requests the IRQs.
This split is necessary, because the IRQ handlers need access to the
arizona->dapm pointer and the snd_sock_jack which are not available
when the codec-driver's probe function runs.
Note this requires that machine-drivers for codecs which are converted
to use the new helper functions from arizona-jack.c are modified to
create a snd_soc_jack through snd_soc_card_jack_new() and register
this jack with the codec through snd_soc_component_set_jack().
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210307151807.35201-10-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
