ASoC: Updates for v6.14
This was quite a quiet release for what I imagine are holiday related
reasons, the diffstat is dominated by some Cirrus Logic Kunit tests.
There's the usual mix of small improvements and fixes, plus a few new
drivers and features. The diffstat includes some DRM changes due to
work on HDMI audio.
- Allow clocking on each DAI in an audio graph card to be configured
separately.
- Improved power management for Renesas RZ-SSI.
- KUnit testing for the Cirrus DSP framework.
- Memory to meory operation support for Freescale/NXP platforms.
- Support for pause operations in SOF.
- Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek
ALC5682I-VE
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>
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
Adds some helper functions and data for applying amp calibration.
1. cs35l56_read_silicon_uid() to get the silicon ID that is used to
search for the correct calibration data entry.
2. Add the registers for the silicon ID to the readable registers.
3. cs35l56_get_calibration() wrapper around
cs_amp_get_efi_calibration_data()
4. cs35l56_calibration_controls() table of the firmware controls
for calibration data.
5. Added members to struct cs35l56_base to store the calibration
data.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
SND_SOC_WCD939X has an optional dependency on TYPEC, so the newly added
SND_SOC_WCD939X_SDW option that selects it needs the same dependency, otherwise
it can fail randconfig builds like:
WARNING: unmet direct dependencies detected for SND_SOC_WCD939X
Depends on [m]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_WCD939X_SDW [=y] && (SOUNDWIRE [=y] || !SOUNDWIRE [=y]) && (TYPEC [=m]
|| !TYPEC [=m])
Selected by [y]:
- SND_SOC_WCD939X_SDW [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SOUNDWIRE [=y]
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_soc_codec_remove':
wcd939x.c:(.text+0x1950): undefined reference to `wcd_clsh_ctrl_free'
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_ear_dac_event':
wcd939x.c:(.text+0x35d8): undefined reference to `wcd_clsh_ctrl_set_state'
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_enable_hphr_pa':
wcd939x.c:(.text+0x39b0): undefined reference to `wcd_clsh_ctrl_set_state'
arm-linux-gnueabi-ld: wcd939x.c:(.text+0x39dc): undefined reference to `wcd_clsh_set_hph_mode'
arm-linux-gnueabi-ld: wcd939x.c:(.text+0x3bc0): undefined reference to `wcd_clsh_ctrl_set_state'
Fixes: be2af391ce ("ASoC: codecs: Add WCD939x Soundwire devices driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://lore.kernel.org/r/20240204212207.3158914-2-arnd@kernel.org
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 aw88395_lib module is shared by all the aw883* drivers that
need to select the corresponding Kconfig symbol. The newly added
aw88399 incorrectly selects SND_SOC_AW88399_LIB instead, which
is not defined anywhere in the kernel, causing a link failure when
the actual one is missing:
arm-linux-gnueabi-ld: sound/soc/codecs/aw88399.o: in function `aw88399_codec_probe':
aw88399.c:(.text+0xbc6): undefined reference to `aw88395_dev_load_acf_check'
arm-linux-gnueabi-ld: aw88399.c:(.text+0xbea): undefined reference to `aw88395_dev_cfg_load'
Fixes: 8ade6cc7e2 ("ASoC: codecs: Add aw88399 amplifier driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20231027152403.386257-2-arnd@kernel.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>
