ASoC: Additional v6.16 updates
A couple more updates on top of the last set I sent you, a new driver
for the ES8375 and a fix for the Cirrus KUnit tests from Jaroslav.
ASoC: Updates for v6.16
The changes in this release are quite large, mainly in drivers rather
than the core. This is partly due to cleanups that touch a lot of
drivers and partly due to several relatively large new drivers.
- Support for automatically enumerating DAIs from standards conforming
SoundWire SDCA devices, further work is required for these to be
useful in an actual card.
- Conversion of quite a few drivers to newer GPIO APIs.
- More helpers and cleanups from Mormimoto-san.
- Support for a wider range of AVS platforms.
- Support for AMD ACP 7.x platforms, Cirrus Logic CS35L63 and CS48L32,
Everest Semiconductor ES8389, Longsoon-1 AC'97 controllers, nVidia
Tegra264, Richtek ALC203 and RT9123 and Rockchip SAI controllers.
Most codes in tas2781_spi_fwlib.c are same as tas2781-fmwlib.c, mainly for
firmware parsing, only differece is the register reading, bit update and
book switching in i2c and spi. The main purpose of this patch is for code
cleaup and arrange the shared part for i2c and spi.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250429111055.567-1-shenghao-ding@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
formance low-power audio DSP with analog and
PDM digital inputs and support for low-power always-on voice-trigger
functionality.
This series adds the devicetree bindings and the ASoC codec driver.
Add a codec driver for the Cirrus Logic CS48L32 audio DSP.
The CS48L32 is a low-power audio DSP with microphone inputs for
"Always on Voice" (i.e. voice trigger) and voice command processing.
It has a programmable Halo Core DSP and a variety of power-efficient
fixed-function audio processors, with configurable digital mixing
and routing.
There are two I2S/TDM audio serial ports.
Four analogue inputs are available through IN1. These feed into a
2-channel ADC through an analogue mux. There is an ALSA control for
each IN1 ADC channel to select which analogue input to use.
A dedicated digital mic (DMIC) PDM input is available on IN2.
Two PDM outputs can feed DMIC inputs on another codec or a host DMIC/PDM
input.
An on-board regulator provides a power supply or bias voltage to
attached microphones. Three switchable MICBIAS outputs are fed from this
allowing only the microphone in use to be powered-up. There are DAPM
widgets for these outputs: MICBIAS1A, MICBIAS1B and MICBIAS1C. The machine
driver must create a DAPM route from the required MICBIAS1x widget to the
INn input widgets to make the MICBIAS switch on when the audio input is
powered-up. For example if the microphone feeding CS48L32 pin IN1LN_1 is
powered from MICBIAS1A, the machine driver must create the path:
(sink) IN1LN_1 <----- (source) MICBIAS1A
Co-developed-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Co-developed-by: Qi Zhou <qi.zhou@cirrus.com>
Signed-off-by: Qi Zhou <qi.zhou@cirrus.com>
Co-developed-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250415115016.505777-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound updates from Takashi Iwai:
"Lots of changes in this cycle, but mostly for cleanups and
refactoring.
Significant amount of changes are about DT schema conversions for ASoC
at this time while we see other usual suspects, too.
Some highlights below:
Core:
- Re-introduction of PCM sync ID support API
- MIDI2 time-base extension in ALSA sequencer API
ASoC:
- Syncing of features between simple-audio-card and the two
audio-graph cards
- Support for specifying the order of operations for components
within cards to allow quirking for unusual systems
- Lots of DT schema conversions
- Continued SOF/Intel updates for topology, SoundWire, IPC3/4
- New support for Asahi Kasei AK4619, Cirrus Logic CS530x, Everest
Semiconductors ES8311, NXP i.MX95 and LPC32xx, Qualcomm LPASS v2.5
and WCD937x, Realtek RT1318 and RT1320 and Texas Instruments
PCM5242
HD-audio:
- More quirks, Intel PantherLake support, senarytech codec support
- Refactoring of Cirrus codec component-binding
Others:
- ALSA control kselftest improvements, and fixes for input value
checks in various drivers"
* tag 'sound-6.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (349 commits)
kselftest/alsa: Log the PCM ID in pcm-test
kselftest/alsa: Use card name rather than number in test names
ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book Pro 360
ALSA: hda/tas2781: Add new quirk for Lenovo Hera2 Laptop
ALSA: seq: ump: Skip useless ports for static blocks
ALSA: pcm_dmaengine: Don't synchronize DMA channel when DMA is paused
ALSA: usb: Use BIT() for bit values
ALSA: usb: Fix UBSAN warning in parse_audio_unit()
ALSA: hda/realtek: Enable headset mic on Positivo SU C1400
ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume
ASoC: codecs: wcd937x: Remove separate handling for vdd-buck supply
ASoC: codecs: wcd937x: Remove the string compare in MIC BIAS widget settings
ASoC: codecs: wcd937x-sdw: Fix Unbalanced pm_runtime_enable
ASoC: dt-bindings: cirrus,cs42xx8: Convert to dtschema
ASoC: cs530x: Remove bclk from private structure
ASoC: cs530x: Calculate proper bclk rate using TDM
ASoC: dt-bindings: cirrus,cs4270: Convert to dtschema
firmware: cs_dsp: Rename fw_ver to wmfw_ver
firmware: cs_dsp: Clarify wmfw format version log message
firmware: cs_dsp: Make wmfw and bin filename arguments const char *
...
Merge series from Matteo Martelli <matteomartelli3@gmail.com>:
This patch set adds support for the Everest-semi ES8311 codec.
Everest-semi ES8311 codec is a low-power mono audio codec with I2S audio
interface and I2C control.
Implemented and tested most of the codec features, with few limitations
listed in the driver commit message. The test setup was composed of a
ESP32-LyraT-Mini board, which embeds the codec, connected via I2C and
I2S to a Raspberry Pi Zero W host board. Some tests were also performed
on a Pine64 A64 host board (e.g. to test the suspend/resume not
supported by the rpi). The codec driver was bound with the simple-card
driver running on kernel v6.9-rc7.
Add support for the Everest-semi ES8311 codec.
Everest-semi ES8311 codec is a low-power mono audio codec with I2S audio
interface and I2C control.
Supported features:
* Both master and slave mode. Master clock is optional in slave mode.
* Sample rates from 8KHz to 96KHz.
* Sample formats: S16_LE, S18_3LE, S20_3LE, S24_3LE, S24_LE and S32_LE.
* I2S formats: I2S, LEFT_J, DSP_A, DSP_B.
* BCLK and FSYNC clocks inversion.
* Component suspend/resume.
* ADC, PGA, DAC controls.
* ADC DSP controls: volume, fade (ramp rate), ALC, automute, HPF, EQ.
* DAC DSP controls: volume, fade (ramp rate), DRC, EQ.
* DAPM routes: capture path with input source selection (differential
MIC/DMIC) and AIF channel source selection; playback path with DAC
channel source selection.
Limitations:
* Support only for master clocks with a ratio of ADC (or DAC) clock to
LRCLK equal to 256. This to keep the default ADC and DAC oversampling
and ADC scale settings. Anyway all 8-96KHz sample rates are supported
when the ratio of MCLK to sample rate is 32, 64, 128, 256, 384 or 512
(upper limit due to max MCLK freq of 49.2MHz).
* Coefficients for ADC HPF and ADC/DAC EQ not supported.
* Digital mic supported but not tested.
* S18_3LE, S20_3LE and S24_3LE formats supported but not tested.
Signed-off-by: Matteo Martelli <matteomartelli3@gmail.com>
Link: https://msgid.link/r/20240522164722.954656-3-matteomartelli3@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Takashi Iwai
Zhang Yi
Mark Brown
Shenghao Ding
Richard Fitzgerald
ChiYuan Huang
Weidong Wang
Kiseok Jo
Martin Povišer
Binbin Zhou
Jack Yu
Igor Prusov
Nicolas Belin
Linus Torvalds
James Ogletree
Paul Handrigan
Khanh Le
Prasad Kumpatla
Shuming Fan
Matteo Martelli
Luca Ceresoli
Herve Codina
Neil Armstrong
Pierre-Louis Bossart
Krzysztof Kozlowski
Linus Walleij