Pull soundwire updates from Vinod Koul:
"This is a small update which features a bit of core changes and driver
updates in Intel and cadence driver.
Core:
- sdw_transfer_defer() API change to drop an argument
- Reset page address rework
- Export sdw_nwrite_no_pm and sdw_nread_no_pm APIs
Drivers:
- Cadence and related intel driver updates for FIFO handling and low
level msg transfers"
* tag 'soundwire-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire:
soundwire: cadence: further simplify low-level xfer_msg_defer() callback
soundwire: cadence: use directly bus sdw_defer structure
soundwire: bus: remove sdw_defer argument in sdw_transfer_defer()
soundwire: stream: use consistent pattern for freeing buffers
soundwire: bus: Remove unused reset_page_addr() callback
soundwire: bus: Don't zero page registers after every transaction
soundwire: bus_type: Avoid lockdep assert in sdw_drv_probe()
soundwire: stream: Move remaining register accesses over to no_pm
soundwire: debugfs: Switch to sdw_read_no_pm
soundwire: Provide build stubs for common functions
soundwire: bus: export sdw_nwrite_no_pm and sdw_nread_no_pm functions
soundwire: cadence: remove unused sdw_cdns_master_ops declaration
soundwire: enable optional clock registers for SoundWire 1.2 devices
ASoC/soundwire: remove is_sdca boolean property
soundwire: cadence: Drain the RX FIFO after an IO timeout
soundwire: cadence: Remove wasted space in response_buf
soundwire: cadence: Don't overflow the command FIFOs
soundwire: intel: remove DAI startup/shutdown
ASoC: Updates for v6.3
There's been quite a lot of activity this release, but not really
one big feature - lots of new devices, plus a lot of cleanup and
modernisation work spread throughout the subsystem:
- More factoring out of common operations into helper functions
by Morimoto-san.
- DT schema conversons and stylistic nits.
- Continued work on building out the new SOF IPC4 scheme.
- Support for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1.
The rt5682s driver switches its regmap to cache-only when the
device suspends and back to regular mode on resume. When the
jack detect interrupt fires rt5682s_irq() schedules the jack
detect work. This can result in invalid reads from the regmap
in cache-only mode if the work runs before the device has
resumed:
[ 19.672162] rt5682s 2-001a: ASoC: error at soc_component_read_no_lock on rt5682s.2-001a for register: [0x000000f0] -16
Disable the jack detection interrupt during suspend and
re-enable it on resume. The driver already schedules the
jack detection work on resume, so any state change during
suspend is still handled.
Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Link: https://lore.kernel.org/r/20230209012002.1.Ib4d6481f1d38a6e7b8c9e04913c02ca88c216cf6@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
When the 'ti,gpio-config' property is not defined, the
device_property_count_u32() will return an error, rather than zero.
The current check, only handles a return value of zero, which assumes that
the property is defined and has nothing defined.
This change extends the check to also check for an error case (most likely
to be hit by the case that the 'ti,gpio-config' is not defined).
In case that the 'ti,gpio-config' and the returned 'gpio_count' is not
correct, there is a 'if (gpio_count != ADCX140_NUM_GPIO_CFGS)' check, a few
lines lower that will return -EINVAL.
This means that someone tried to define 'ti,gpio-config', but with the
wrong number of GPIOs.
Fixes: d521432149 ("ASoC: tlv320adcx140: Add support for configuring GPIO pin")
Signed-off-by: Steffen Aschbacher <steffen.aschbacher@stihl.de>
Signed-off-by: Alexandru Ardelean <alex@shruggie.ro>
Link: https://lore.kernel.org/r/20230213073805.14640-1-alex@shruggie.ro
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound fixes from Takashi Iwai:
"Hopefully the last one for 6.2, a collection of the fixes that have
been gathered since the last pull.
All changes are small and trivial device-specific fixes"
* tag 'sound-6.2-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Add Positivo N14KP6-TG
ASoC: topology: Return -ENOMEM on memory allocation failure
ALSA: emux: Avoid potential array out-of-bound in snd_emux_xg_control()
ASoC: fsl_sai: fix getting version from VERID
ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform.
ALSA: hda/realtek: Add quirk for ASUS UM3402 using CS35L41
ASoC: codecs: es8326: Fix DTS properties reading
ASoC: tas5805m: add missing page switch.
ASoC: tas5805m: rework to avoid scheduling while atomic.
ALSA: hda/realtek: Enable mute/micmute LEDs on HP Elitebook, 645 G9
ASoC: SOF: amd: Fix for handling spurious interrupts from DSP
ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book2 Pro 360
ALSA: pci: lx6464es: fix a debug loop
ASoC: rt715-sdca: fix clock stop prepare timeout issue
Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
From q6dsp side issues are around locking of position pointer and handle
From LPASS codec side most of the staiblity issues were around runtime pm,:
While testing X13s audio, we found multiple stablity issues this patchset
fixes these issues.
From q6dsp side issues are around locking of position pointer and handle
multiple prepare cases along with pulse audio timerbased scheduling workaround.
From LPASS codec side most of the staiblity issues were around runtime pm,
hitting various issues as the codec was firstly resetting the soundwire block
for every clk disable/enable which is taking the slaves out of sync and
resulting in re-enumerating. Second issue was around fsgen clk is not
brining up the codec out of suspend as it was not added after
runtime pm enabled. Final issue was with codec mclk rate which should
have been 192KHz same as npl instead of 96KHz. We were getting lucky as
wsa drivers are setting the same clk to 192KHz.
With this patches, x13s audio is pretty stable.
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.
For some reason we ended up with incorrect mclk rate which should be
1920000 instead of 96000, So far we were getting lucky as the same clk
is set to 192000 by wsa and va macro. This issue is discovered when there
is no wsa macro active and only rx or tx path is tested.
Fix this by setting correct rate.
Fixes: c39667ddcf ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro")
Fixes: af3d54b997 ("ASoC: codecs: lpass-rx-macro: add support for lpass rx macro")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-7-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Daniel Beer <daniel.beer@igorinstitute.com>:
This pair of patches fixes two issues which crept in while revising the
original submission, at a time when I no longer had access to test
hardware.
The fixes here have been tested and verified on hardware.
cppcheck reports
sound/soc/codecs/aw88395/aw88395_lib.c:789:6: error: Uninitialized variable: cur_scene_id [uninitvar]
if (cur_scene_id == 0) {
^
Passing a garbage value to aw_dev_parse_data_by_sec_type_v1() will cause a crash
when the value is used as an array index. This check assumes cur_scene_id is
initialized to 0, so initialize it to 0.
Fixes: 4345865b00 ("ASoC: codecs: ACF bin parsing and check library file for aw88395")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20230205015733.1721009-1-trix@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Seems like properties parsing and reading was copy-pasted,
so "everest,interrupt-src" and "everest,interrupt-clk" are saved into
the es8326->jack_pol variable. This might lead to wrong settings
being saved into the reg 57 (ES8326_HP_DET).
Fix this by using proper variables while reading properties.
Signed-off-by: Alexey Firago <a.firago@yadro.com>
Reviewed-by: Yang Yingliang <yangyingliang@huawei.com
Link: https://lore.kernel.org/r/20230204195106.46539-1-a.firago@yadro.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There's some setup we need to do in order to get the DSP initialized,
and this can't be done until a bit-clock is ready. In an earlier version
of this driver, this work was done in a DAPM callback.
The DAPM callback doesn't guarantee that the bit-clock is running, so
the work was moved instead to the trigger callback. Unfortunately this
callback runs in atomic context, and the setup code needs to do I2C
transactions.
Here we use a work_struct to kick off the setup in a thread instead.
Fixes: ec45268467 ("ASoC: add support for TAS5805M digital amplifier")
Signed-off-by: Daniel Beer <daniel.beer@igorinstitute.com>
Link: https://lore.kernel.org/r/85d8ba405cb009a7a3249b556dc8f3bdb1754fdf.1675497326.git.daniel.beer@igorinstitute.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound fixes from Takashi Iwai:
"A bit higher volume of changes than wished, but each change is
relatively small and the fix targets are mostly device-specific, so
those should be safe as a late stage merge.
The most significant LoC is about the memalloc helper fix, which is
applied only to Xen PV. The other major parts are ASoC Intel SOF and
AVS fixes that are scattered as various small code changes. The rest
are device-specific fixes and quirks for HD- and USB-audio, FireWire
and ASoC AMD / HDMI"
* tag 'sound-6.2-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: firewire-motu: fix unreleased lock warning in hwdep device
ALSA: memalloc: Workaround for Xen PV
ASoC: cs42l56: fix DT probe
ASoC: codecs: wsa883x: correct playback min/max rates
ALSA: hda/realtek: Add Acer Predator PH315-54
ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table
ALSA: hda: Do not unset preset when cleaning up codec
ASoC: SOF: sof-audio: prepare_widgets: Check swidget for NULL on sink failure
ASoC: hdmi-codec: zero clear HDMI pdata
ASoC: SOF: ipc4-mtrace: prevent underflow in sof_ipc4_priority_mask_dfs_write()
ASoC: Intel: sof_ssp_amp: always set dpcm_capture for amplifiers
ASoC: Intel: sof_nau8825: always set dpcm_capture for amplifiers
ASoC: Intel: sof_cs42l42: always set dpcm_capture for amplifiers
ASoC: Intel: sof_rt5682: always set dpcm_capture for amplifiers
ALSA: hda/via: Avoid potential array out-of-bound in add_secret_dac_path()
ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless
ALSA: hda/realtek: fix mute/micmute LEDs, speaker don't work for a HP platform
ASoC: SOF: keep prepare/unprepare widgets in sink path
ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL
ASoC: SOF: sof-audio: unprepare when swidget->use_count > 0
...
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 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
struct snd_soc_dai need to have info for playback/capture,
but it is using "playback/capture_xxx" or "tx/tx_xxx" or array.
This kind of random definition is very difficult to read.
This patch-set add helper functions and each driver use it.
And cleanup the definition.
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
Merge series from Herve Codina <herve.codina@bootlin.com>:
The Renesas IDT821034 codec is four channel PCM codec with on-chip
filters and programmable gain setting. It also provides SLIC
(Subscriber Line Interface Circuit) signals as GPIOs.
idle_bias_on was set because cs42l42 has a "VMID" type pseudo-midrail
supply (named FILT+), and these typically take a long time to charge.
But the driver never enabled pm_runtime so it would never have powered-
down the cs42l42 anyway.
In fact, FILT+ can charge to operating voltage within 12.5 milliseconds
of enabling HP or ADC. This time is already covered by the startup
delay of the HP/ADC.
The datasheet warning about FILT+ taking up to 1 second to charge only
applies in the special cases that either the PLL is started or
DETECT_MODE set to non-zero while both HP and ADC are off. The driver
never does either of these.
Removing idle_bias_on allows the Soundwire host controller to suspend
if there isn't a snd_soc_jack handler registered.
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/20230127165111.3010960-8-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 chosen clocking configuration must give an internal MCLK (MCLKint)
that is an integer multiple of the sample rate.
On I2S each of the supported bit clock frequencies can only be generated
from one sample rate group (either the 44100 or the 48000) so the code
could use only the bitclock to look up a PLL config.
The relationship between sample rate and bitclock frequency is more
complex on Soundwire and so it is possible to set a frame shape to
generate a bitclock from the "wrong" group. For example 2*147 with a
48000 sample rate would give a bitclock of 14112000 which on I2S
could only be derived from a 44100 sample rate.
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/20230127165111.3010960-4-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
