Add a KUnit test for the cs-amp-lib library. This has test cases
for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs().
A KUNIT_STATIC_STUB_REDIRECT() has been added to
cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the
KUnit test can redirect these to test harness functions.
Much of the testing involves invoking the same function with different
parameters, i.e. the number of amps and the amp index within the array.
This uses parameterization rather than looping. The idea is to avoid
looping over configurations within one test case as that has a higher
chance of having a bug that doesn't actually test all the expected cases.
Having the test run exactly one configuration, and then tear-down, is less
prone to accidentally skipped configurations.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240304143705.26362-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
Factory calibration of the speakers stores the calibration information
into an EFI variable.
This set of patches adds support for applying speaker calibration
data from that EFI variable.
The HDA patch (#5) depends on the ASoC patches #2 and #3
Create a new library for code that is used by multiple Cirrus Logic
amps. This initially implements extracting amp calibration data
from EFI and writing it to firmware controls.
During factory calibration of built-in speakers the firmware
calibration constants are stored in an EFI file. The file contains
an array of calibration constants for each of the speakers.
cs_amp_get_calibration_data() searches for an entry matching the
requested UID stamp, otherwise by array index. If the data is found in
EFI the constants for that speaker are copied back to the caller.
If EFI is not enabled, the cs_amp_get_calibration_data() implementation
will compile to simply return -ENOENT and the linker can drop the code.
The code to write calibration controls uses cs_dsp. Building of cs_dsp
is not forced. Instead, the code will compile away the calls to
cs_dsp if cs_dsp is not reachable.
This strategy of conditional code allows cs-amp-lib to be shared by
multiple drivers without forcing inclusion of other modules that might
be unnecessary.
The calls to efi.get_variable() and cs_dsp are in small wrapper
functions. This is so that a KUNIT_STATIC_STUB_REDIRECT can be added in
a future patch to redirect these calls to replacement functions for
KUnit testing.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the main WCD9390/WCD9395 Audio Codec driver to support:
- 4 ADC inputs for up to 5 Analog Microphones
- 4 DMIC inputs for up to 8 Digital Microphones
- 4 Microphone BIAS
- Stereo Headphone output
- Mono EAR output
- MBHC engine for Headset Detection
It makes usage of the generic MBHC and CLSH generic code and
the USB Type-C mux and switch helpers to gather USB-C Events
in order to properly setup Headset Detection mechanism
when connected behind the separate USB-C Mux subsystem.
WCD9390/WCD9395 supports a PCM path for Playback instead
of the actually implemented PDM playback, it will be
implemented later.
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://msgid.link/r/20231219-topic-sm8650-upstream-wcd939x-codec-v4-5-1c3bbff2d7ab@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The CS35L56 combines a high-performance mono audio amplifier, Class-H
tracking inductive boost converter, Halo Core(TM) DSP and a DC-DC boost
converter supporting Class-H tracking.
Supported control interfaces are I2C, SPI or SoundWire.
Supported audio interfaces are I2S/TDM or SoundWire.
Most chip functionality is controlled by on-board ROM firmware that is
always running. The driver must apply patch/tune to the firmware
before using the CS35L56.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230320112245.115720-9-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
