Mark Brown [Sat, 26 Apr 2025 12:28:57 +0000 (13:28 +0100)]
ASoC: add Renesas MSIOF sound driver
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Renesas MSIOF can work as both SPI and I2S.
Current Linux supports MSIOF-SPI. This patch-set adds new MSIOF-I2S.
Because it is using same HW-IP, we want to share same compatible for both
MSIOF-SPI/I2S case. MSIOF-I2S (Sound) will use Audio-Graph-Card/Card2 which
uses Of-Graph, but MSIOF-SPI is not use Of-Graph.
So, this patch-set assumes it was used as MSIOF-I2S if DT is using Of-Graph,
otherwise, it is MSIOF-SPI (This assumption will works if SPI *never*
use Of-Graph in the future).
One note so far is that it is using "spi@xxx" node name for both
MSIOF-SPI/I2S. DTC will automatically checks "spi@xxx" node as SPI device
which requests #address-cells/#size-cells. But is not needed for I2S.
So we will get warning about it on Sparrow Hawk which uses MSIOF-I2S.
We have no solution about it, so far.
ADG needs its MSTP to use it, and it was handled as "clk_i" before.
R-Car Gen2/Gen3 are using it, but Gen4 doesn't have it.
"clk_i" is not intuitive for ADG MSTP.
Let's enable to use "adg" clock. It can keep compatible with R-Car
Gen2/Gen3 and Gen4.
ASoC: renesas: rsnd: care BRGA/BRGB select in rsnd_adg_clk_enable()
Renesas rsnd related clocks are enabled by rsnd_adg_clk_enable(),
but it doesn't care about BRGA/BRGB selection (It is handled when
SSI was started) (BRGA is used for 44.1kHz lineage, BRGB is used for
48kHz lineage in this driver).
But it should be handled since probe time.
Includes BRGCKR_31 in adg->ckr to handle it since boot time.
ASoC: renesas: rsnd: allow to use ADG as standalone
Audio clock generator (= ADG) can be used standalone (without DAI),
but current driver indicates error if it doesn't have DAI on DT.
It is not error, allow to use ADG as standalone.
dt-bindings: renesas,sh-msiof: Add MSIOF I2S Sound support
Renesas MSIOF (Clock-Synchronized Serial Interface with FIFO) can work as
both SPI and I2S. MSIOF-I2S will use Audio Graph Card/Card2 driver which
uses Of-Graph in DT.
MSIOF-SPI/I2S are using same DT compatible properties.
MSIOF-I2S uses Of-Graph for Audio-Graph-Card/Card2,
MSIOF-SPI doesn't use Of-Graph.
Mark Brown [Fri, 25 Apr 2025 17:40:40 +0000 (18:40 +0100)]
ASoC: codecs: wcd938x: fix mux error handling
Merge series from Johan Hovold <johan+linaro@kernel.org>:
A recent change added support for looking up an optional mux control
before falling back to gpio control for us-euro plug selection.
The mux framework does however not yet support optional muxes and an
error message is now incorrectly logged on machines like the Lenovo
ThinkPad X13s which do not have one:
wcd938x_codec audio-codec: /audio-codec: failed to get mux-control (0)
Suppress the bogus error and add the missing mux error handling by
making sure that the 'mux-controls' DT property is present before
looking up the mux control.
spi: sh-msiof: ignore driver probing if it was MSIOF Sound
Renesas MSIOF (Clock-Synchronized Serial Interface with FIFO) can work as
both SPI and I2S. MSIOF-I2S will use Audio Graph Card/Card2 driver which
Of-Graph in DT.
MSIOF-SPI/I2S are using same DT compatible properties.
MSIOF-I2S uses Of-Graph for Audio-Graph-Card/Card2,
MSIOF-SPI doesn't use Of-Graph.
Richard Fitzgerald [Wed, 23 Apr 2025 19:27:13 +0000 (20:27 +0100)]
MAINTAINERS: ASoC: Simplify references to Cirrus Logic include files
Change the references to Cirrus Logic files under include/sound to be a
wildcard of all cs* and use X: to exclude the three cs* files that are not
related to ASoC.
This means that new Cirrus Logic files added to include/sound will be
picked up automatically as an ASoC file without any changes to MAINTAINERS.
New files are most likely to be for ASoC drivers now.
It's also easier to manage than ranges of wildcarding for different sets of
files.
Due to unfortunate interaction between two recent patchsets, avs driver
ends up dereferencing NULL pointer when registering i2s boards.
Commit [1] does a call to avs_mach_ssp_tdm() which requires mach->pdata,
however due to another change in [2], mach->pdata is initialized after
the call. Reshuffle code to initialise mach->pdata before calling
avs_mach_ssp_tdm().
Martin Blumenstingl [Sat, 19 Apr 2025 21:34:48 +0000 (23:34 +0200)]
ASoC: meson: meson-card-utils: use of_property_present() for DT parsing
Commit c141ecc3cecd ("of: Warn when of_property_read_bool() is used on
non-boolean properties") added a warning when trying to parse a property
with a value (boolean properties are defined as: absent = false, present
without any value = true). This causes a warning from meson-card-utils.
meson-card-utils needs to know about the existence of the
"audio-routing" and/or "audio-widgets" properties in order to properly
parse them. Switch to of_property_present() in order to silence the
following warning messages during boot:
OF: /sound: Read of boolean property 'audio-routing' with a value.
OF: /sound: Read of boolean property 'audio-widgets' with a value.
Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Tested-by: Christian Hewitt <christianshewitt@gmail.com> Cc: stable@vger.kernel.org Signed-off-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com> Link: https://patch.msgid.link/20250419213448.59647-1-martin.blumenstingl@googlemail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Johan Hovold [Tue, 15 Apr 2025 07:41:44 +0000 (09:41 +0200)]
ASoC: codecs: wcd938x: fix mux error handling
A recent change added support for looking up an optional mux control
before falling back to gpio control for us-euro plug selection.
The mux framework does however not yet support optional muxes and an
error message is now incorrectly logged on machines like the Lenovo
ThinkPad X13s which do not have one:
wcd938x_codec audio-codec: /audio-codec: failed to get mux-control (0)
Suppress the bogus error and add the missing mux error handling by
making sure that the 'mux-controls' DT property is present before
looking up the mux control.
Peng Fan [Mon, 14 Apr 2025 08:34:25 +0000 (16:34 +0800)]
ASoC: codec: tpa6130a2: Convert to GPIO descriptors
of_gpio.h is deprecated, update the driver to use GPIO descriptors.
- Use devm_gpiod_get_optional to get GPIO descriptor with default
polarity GPIOD_OUT_LOW, set consumer name.
- Use gpiod_set_value to configure output value.
Checking the DTS polarity, all users are using GPIOD_ACTIVE_HIGH.
so all should work as expected with this patch.
Mark Brown [Wed, 23 Apr 2025 11:43:22 +0000 (12:43 +0100)]
Add RK3576 SAI Audio Controller Support
Merge series from Nicolas Frattaroli <nicolas.frattaroli@collabora.com>:
This series adds support for Rockchip's Serial Audio Interface (SAI)
controller, found on SoCs such as the RK3576. The SAI is a flexible
controller IP that allows both transmitting and receiving digital audio
in the I2S, TDM and PCM formats. Instances of this controller are used
both for externally exposed audio interfaces, as well as for audio on
video interfaces such as HDMI.
Charles Keepax [Wed, 23 Apr 2025 09:09:44 +0000 (10:09 +0100)]
ASoC: cs42l43: Disable headphone clamps during type detection
The headphone clamps cause fairly loud pops during type detect
because they sink current from the detection process itself. Disable
the clamps whilst the type detect runs, to improve the detection
pop performance.
Nicolas Frattaroli [Thu, 10 Apr 2025 19:39:58 +0000 (21:39 +0200)]
ASoC: rockchip: add Serial Audio Interface (SAI) driver
The Rockchip RK3576 SoC features a new audio controller, the Serial
Audio Interface, or SAI for short. It is capable of both sending and
receiving audio over up to 4 lanes for each direction using the I2S,
PCM or TDM formats.
This driver is derived from the downstream vendor driver. That is why
its original author, Sugar Zhang, is listed as a Co-developer, with
their signoff. Since adjustments to make the driver suitable for
upstream were quite extensive, I've added myself to the authors and put
myself as the commit author; all added bugs are my fault alone, and not
that of the original author at Rockchip.
The hardware is somewhat similar to the Rockchip I2S-TDM hardware when
judged based on their register map, except it uses the same mclk for
tx and rx. It appears to be much more flexible with regards to TDM.
The loopback stuff and mono mode are new as well.
In line with the changes that were made to the Rockchip I2S-TDM driver
after upstreaming, the mclk-calibrate functionality was dropped, and
setting the mclk rate properly is now left up to the Common Clock
Framework, similar to how it is in the upstream I2S-TDM driver now.
A spinlock has been introduced to protect register write accesses that
depend on the bclk/fs to be stopped, i.e. XFER[1:0] being 0. I couldn't
find whether the asoc core held a per-instance lock so only one callback
can run at a time, and so it seemed prudent to add this.
I couldn't successfully test whether TDM was working, though I've tried
with a TAS6424 codec board. I'm not sure yet whether to blame the codec
driver, this version of the SAI driver, or the vendor implementation of
the SAI driver. The TDM mask registers remain untouched in both this
version and the downstream vendor version, which is suspicious, though
the Linux ASoC core wouldn't be able to support the 128 (!!!) slots of
TDM the hardware supports anyway.
Regular old 2-channel stereo I2S thrown at an I2S stereo codec works
well though. I tested with the CPU-side SAI controller in provider mode
and an Everest ES8388 codec as the consumer.
Some vendor driver features (no-dmaengine, fifo rockchip performance
monitoring, many kcontrols) were dropped for this initial upstream
version. They can always be added later if they make sense for upstream.
Co-developed-by: Sugar Zhang <sugar.zhang@rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com> Signed-off-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com> Link: https://patch.msgid.link/20250410-rk3576-sai-v2-6-c64608346be3@collabora.com Signed-off-by: Mark Brown <broonie@kernel.org>
Nicolas Frattaroli [Thu, 10 Apr 2025 19:39:57 +0000 (21:39 +0200)]
ASoC: dt-bindings: add schema for rockchip SAI controllers
Rockchip introduced a new audio controller called the "Serial Audio
Interface", or "SAI" for short, on some of their newer SoCs. In
particular, this controller is used several times on the RK3576 SoC.
Add a schema for it, with only an RK3576 compatible for now. Other SoCs
may follow as mainline support for them lands.
When building for a platform that does not support CONFIG_PM, such as
s390, cs48l32_runtime_{suspend,resume}() are unused because
SET_RUNTIME_PM_OPS does not reference its argument when CONFIG_PM is not
set:
sound/soc/codecs/cs48l32.c:3822:12: error: 'cs48l32_runtime_suspend' defined but not used [-Werror=unused-function]
3822 | static int cs48l32_runtime_suspend(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/cs48l32.c:3779:12: error: 'cs48l32_runtime_resume' defined but not used [-Werror=unused-function]
3779 | static int cs48l32_runtime_resume(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~
cc1: all warnings being treated as errors
Use RUNTIME_PM_OPS and pm_ptr() to ensure these functions are seen as
used by the compiler but be dropped in the final object file when
CONFIG_PM is not set, matching the current behavior while clearing up
the warnings.
Mark Brown [Thu, 17 Apr 2025 15:01:05 +0000 (16:01 +0100)]
ASoC: mt8195: Add support for MT8395 Radxa NIO 12L
Merge series from Julien Massot <julien.massot@collabora.com>:
This patch series adds support for audio playback on the MT8395-based Radxa NIO 12L platform, which uses the integrated MT6359 codec via internal DAI links.
Key additions:
- Support for a new `mediatek,mt8195_mt6359` card configuration that does not rely on external codecs like rt5682.
- Proper memory region declarations and pinctrl setup for the audio front-end (AFE) and audio DSP (ADSP).
- A device tree sound node for headphone audio routing using `DL_SRC_BE` and `AIF1`.
- Enhancements to the DT bindings to document the new compatible string, missing link-name, and additional audio routes (Headphone L/R).
ASoC: mediatek: mt8195: Move rt5682 specific dapm routes
'HPOL', 'HPOR', and 'Headset Mic' are rt5682 specific,
move these routes to mt8195_rt5682_init so we keep
'mt8195_rt5682_routes' for connection between mt8195 and
mt6359.
Richard Fitzgerald [Wed, 16 Apr 2025 12:24:22 +0000 (13:24 +0100)]
firmware: cs_dsp: Add some sanity-checking to test harness
Add sanity checking to some test harness functions to help catch bugs
in the test code. This consists of checking the range of some arguments
and checking that reads from the dummy regmap succeed.
Most of the harness code already had sanity-checking but there were a
few places where it was missing or was assumed that the test could be
trusted to pass valid values.
Shuming Fan [Wed, 16 Apr 2025 09:25:28 +0000 (17:25 +0800)]
ASoC: rt722: fix pop noise at the beginning of headphone playback
This patch added the function_status check to avoid the calibration again.
The codec driver reinitializes when the 'FUNCTION_NEEDS_INITIALIZATION' flag raises.
Richard Fitzgerald [Wed, 16 Apr 2025 12:16:04 +0000 (13:16 +0100)]
ASoC: cs-amp-lib-test: Use faux bus instead of creating a dummy platform device
Change the code to use faux bus for the dummy codec driver device instead
of creating a platform device. Also use KUnit automatic resource cleanup to
destroy the device instead of doing it "manually" in a test case exit()
function.
Mark Brown [Tue, 15 Apr 2025 20:23:56 +0000 (21:23 +0100)]
ASoC: skip the endpoint that doesn't present and
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
A codec endpoint may not be used. We could check the present SDCA
functions to know if the endpoint is used or not. Skip the endpoint
which is not used. And load the topology dynamically for each endpoint.
With this feature, we don't need to use the quirk to determine the
existence of the optional codec DAIs.
Richard Fitzgerald [Tue, 15 Apr 2025 13:18:37 +0000 (14:18 +0100)]
ASoC: cs-amp-lib-test: Use flex_array_size()
Use flex_array_size() when calculating the number of bytes argument to
get_random_bytes(). This replaces a calculation based on passing sizeof()
the expected type of the destination.
Richard Fitzgerald [Tue, 15 Apr 2025 11:50:16 +0000 (12:50 +0100)]
ASoC: cs48l32: Add driver for Cirrus Logic CS48L32 audio DSP
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>
The CS48L32 is an Audio DSP with microphone inputs and SPI
control interface. It has a programmable DSP and a variety of
power-efficient fixed-function audio processors, with configurable
digital mixing and routing.
Most properties are core properties: supply regulators, gpios, clocks,
interrupt parent and SPI interface. The custom properties define
the configuration of the microphone inputs to match what is physically
attached to them.
Richard Fitzgerald [Tue, 15 Apr 2025 12:29:27 +0000 (13:29 +0100)]
ASoC: Add Cirrus and Wolfson headers to ASoC section of MAINTAINERS
Specifically list various Cirrus Logic and Wolfson Micro codec header files
under include/sound/ within the ASoC section of MAINTAINERS. Note that not
all the include/sound/cs* files are part of ASoC, so more-specific patterns
are needed.
These files are all part of ASoC codec drivers, and are owned by specific
Cirrus Logic and Wolfson Micro sections of MAINTAINERS. But the overall
include/sound/* maintainership is only Takashi Iwai and Jaroslav Kysela.
So by default get_maintainer.pl would only show Takashi and Jaroslav as
maintainers for any patch that changes these files without changing any
code under sound/soc.
There is a separate MAINTAINERS section for ASoC, so the headers must be
added there to make the ASoC maintainers show up in get_maintainer.pl.
ASoC: cs-amp-lib: Annotate struct cirrus_amp_efi_data with __counted_by()
Add the __counted_by() compiler attribute to the flexible array member
'data' to improve access bounds-checking via CONFIG_UBSAN_BOUNDS and
CONFIG_FORTIFY_SOURCE.
ASoC: cs-amp-lib: Replace offsetof() with struct_size()
Use struct_size() to calculate the number of bytes to allocate and used
by 'cirrus_amp_efi_data'. Compared to offsetof(), struct_size() provides
additional compile-time checks (e.g., __must_be_array()).
ASoC: simple-card-utils: fixup dlc->xxx handling for error case
Current graph_util_parse_dai() has 2 issue for dlc->xxx handling.
1) dlc->xxx might be filled if snd_soc_get_dai_via_args() (A) works.
In such case it will fill dlc->xxx first (B), and detect error
after that (C). We need to fill dlc->xxx in success case only.
(A) dai = snd_soc_get_dai_via_args(&args);
if (dai) {
ret = -ENOMEM;
^ dlc->of_node = ...
(B) dlc->dai_name = ...
v dlc->dai_args = ...
(C) if (!dlc->dai_args)
goto end;
...
}
2) graph_util_parse_dai() itself has 2 patterns (X)(Y) to fill dlc->xxx.
Both case, we need to call of_node_put(node) (Z) in error case, but we
are calling it only in (Y) case.
int graph_util_parse_dai(...)
{
...
dai = snd_soc_get_dai_via_args(&args);
if (dai) {
...
^ dlc->of_node = ...
(X) dlc->dai_name = ...
v dlc->dai_args = ...
...
}
...
(Y) ret = snd_soc_get_dlc(&args, dlc);
if (ret < 0) {
(Z) of_node_put(node);
...
}
...
}
This patch fixup both case. Make it easy to understand, update
lavel "end" to "err", too.
Mark Brown [Mon, 14 Apr 2025 10:24:01 +0000 (11:24 +0100)]
Add support for Loongson-1 AC97
Merge series from Keguang Zhang <keguang.zhang@gmail.com>:
Add the driver and dt-binding document for Loongson-1 AC97.
Add the dt-binding document for Realtek ALC203 Codec.
Add DT support for the AC97 generic codec driver.
Mark Brown [Mon, 14 Apr 2025 10:23:57 +0000 (11:23 +0100)]
ASoC: convert GPIO chips to using new value setters
Merge series from Bartosz Golaszewski <brgl@bgdev.pl>:
struct gpio_chip now has callbacks for setting line values that return
an integer, allowing to indicate failures. We're in the process of
converting all GPIO drivers to using the new API. This series converts
all ASoC GPIO controllers.
Mark Brown [Mon, 14 Apr 2025 10:23:48 +0000 (11:23 +0100)]
ASoC: tas27{64,70}: improve support for Apple codec
Merge series from James Calligeros <jcalligeros99@gmail.com>:
This series introduces a number of changes to the drivers for
the Texas Instruments TAS2764 and TAS2770 amplifiers in order to
introduce (and improve in the case of TAS2770) support for the
variants of these amps found in Apple Silicon Macs.
Apple's variant of TAS2764 is known as SN012776, and as always with
Apple is a subtly incompatible variant with a number of quirks. It
is not publicly available. The TAS2770 variant is known as TAS5770L,
and does not require incompatible handling.
Much as with the Cirrus codec patches, I do not
expect that we will get any official acknowledgement that these parts
exist from TI, however I would be delighted to be proven wrong.
This series has been living in the downstream Asahi kernel tree[1]
for over two years, and has been tested by many thousands of users
by this point[2].
v4 drops the TDM idle TX slot behaviour patches. I experimented with
the API discussed in v3, however this did not work on any of the machines
I tested it with. More tweaking is probably needed.
ASoC: soc_sdw_utils: skip the endpoint that doesn't present
A codec endpoint may not be used. We could check the present SDCA
functions to know if the endpoint is used or not. Skip the endpoint
which is not used.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://patch.msgid.link/20250414063239.85200-12-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Currently asoc_sdw_get_codec_name will return codec_info->codec_name if
it is set. However, in some case we need the sdw codec name no matter if
codec_info->codec_name is set or not.
_asoc_sdw_get_codec_name() will be used in the follow up commit.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://patch.msgid.link/20250414063239.85200-11-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we always use single topology file to describe the widgets.
However, with SDCA, we want to be able to load sub-topologies based on
the supported device functions.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://patch.msgid.link/20250414063239.85200-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
We always use a single topology that contains all PCM devices belonging
to a machine configuration.
However, with SDCA, we want to be able to load function topologies based
on the supported device functions. This change is in preparation for
loading those function topologies.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://patch.msgid.link/20250414063239.85200-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
James Calligeros [Sat, 5 Apr 2025 23:15:12 +0000 (09:15 +1000)]
ASoC: tas2764: expose die temp to hwmon
TAS2764 contains an ADC that reports the chip's die temperature.
The temperature in degrees Celsius is yielded by subtracting 93
from the raw value reported by the ADC.
Expose the codec die temperature to the hwmon interface.
The chip will initialise the temperature register to 2.6 *C
to avoid triggering over temp protection. As the ADC is powered
down during software shutdown, this value will persist until the
chip is fully powered up (e.g. when the PCM it's attached to is
opened). When the chip is powered back down, the last value sampled
will persist in the register.
James Calligeros [Sat, 5 Apr 2025 23:15:11 +0000 (09:15 +1000)]
ASoC: tas2770: expose die temp to hwmon
TAS2770 includes an ADC which reports the chip's die temperature.
As per the datasheet, the temperature in degrees Celsius is derived
by taking the raw value stored in the ADC's registers, dividing by 16,
then subtracting 93.
Create and register a hwmon device to expose the chip's die temperature
to the hwmon interface.
The ADC is shut down during software shutdown mode, and its registers
are initialised to 0 on reset. This means that the die temperature will
read -93 *C until the chip has been fully powered up at least once (e.g.
the PCM its attached to is opened). Once the chip is put into software
shutdown again, the ADC will also shut down. The last value sampled
before this happens will persist in the ADC's registers.
Martin Povišer [Sat, 5 Apr 2025 23:15:10 +0000 (09:15 +1000)]
ASoC: tas2764: Apply Apple quirks
Apple's SN012776 driver has some peculiar aspects to its behavior that
are suspected to work around issues in the codec part. Add a module
parameter for enabling individual quirks that should be imitated after
the Apple driver.
Setting some of these by default seems to be required. For example,
setting 0xf fixes an issue with transient overcurrent errors which
can crash the chip until the next system reboot. To be safe, let's
enable all of them by default.
Reviewed-by: Neal Gompa <neal@gompa.dev> Co-developed-by: Hector Martin <marcan@marcan.st> Signed-off-by: Hector Martin <marcan@marcan.st> Signed-off-by: Martin Povišer <povik+lin@cutebit.org> Co-developed-by: James Calligeros <jcalligeros99@gmail.com> Signed-off-by: James Calligeros <jcalligeros99@gmail.com> Link: https://patch.msgid.link/20250406-apple-codec-changes-v5-6-50a00ec850a3@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Hector Martin [Sat, 5 Apr 2025 23:15:08 +0000 (09:15 +1000)]
ASoC: tas2764: Enable main IRQs
IRQ handling was added in commit dae191fb957f ("ASoC: tas2764: Add IRQ
handling") however that same commit masks all interrupts coming from
the chip. Unmask the "main" interrupts so that we can see and
deal with a number of errors including clock, voltage, and current.
Fixes: dae191fb957f ("ASoC: tas2764: Add IRQ handling") Reviewed-by: Neal Gompa <neal@gompa.dev> Signed-off-by: Hector Martin <marcan@marcan.st> Signed-off-by: James Calligeros <jcalligeros99@gmail.com> Link: https://patch.msgid.link/20250406-apple-codec-changes-v5-4-50a00ec850a3@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Martin Povišer [Sat, 5 Apr 2025 23:15:07 +0000 (09:15 +1000)]
ASoC: tas2764: Reinit cache on part reset
When the part is reset in component_probe, do not forget to reinit the
regcache, otherwise the cache can get out of sync with the part's
actual state. This fix is similar to commit 0a0342ede303
("ASoC: tas2770: Reinit regcache on reset") which concerned the
tas2770 driver.
Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764") Reviewed-by: Neal Gompa <neal@gompa.dev> Signed-off-by: Martin Povišer <povik+lin@cutebit.org> Signed-off-by: James Calligeros <jcalligeros99@gmail.com> Link: https://patch.msgid.link/20250406-apple-codec-changes-v5-3-50a00ec850a3@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Hector Martin [Sat, 5 Apr 2025 23:15:06 +0000 (09:15 +1000)]
ASoC: tas2770: Support setting the PDM TX slot
We don't actually support configuring the PDM input right now. Rather,
this is useful as a hack.
On Apple Silicon machines, amps are split between two I2S buses which
are logically ANDed internally at the SoC. Odd and even slot groups are
driven by amps on either bus respectively. Since the signals are ANDed,
unused slot groups must be driven as zero to avoid corrupting the data
from the other side.
On most recent machines (TAS2764-based), this is accomplished using the
"SDOUT zero mask" feature of that chip. Unfortunately, TAS2770 does not
support this. It does support zeroing out *all* unused slots, which
works well for machines with a single amp per I2S bus. That is all,
except one.
The 13" M1 MacBook Pro is the only machine using TAS2770 and two amps
per I2S bus:
L Bus: SPK0I SPK0V Hi-Z Hi-Z SPK2I SPK2V Hi-Z Hi-Z
R Bus: Hi-Z Hi-Z SPK1I SPK2V Hi-Z Hi-Z SPK3I SPK3V
To ensure uncorrupted data, we need to force all the Hi-Z periods to
zero. We cannot use the "force all zero" feature, as that would cause a
bus conflict between both amps. We can use the pull-down feature, but
that leaves a few bits of garbage on the trailing edge of the speaker
data, since the pull-down is weak.
This is where the PDM transmit feature comes in. With PDM grounded and
disabled (the default state), the PDM slot is transmitted as all zeroes.
We can use that to force a zero 16-bit slot after the voltage data for
each speaker, cleaning it up. Then the pull-down ensures the line stays
low for the subsequent slot:
L Bus: SPK0I SPK0V PDM0 PulDn SPK2I SPK2V PDM0 PulDn
R Bus: PDM0 PulDn SPK1I SPK2V PDM0 PulDn SPK3I SPK3V
Yes, this is a horrible hack, but it beats adding dummy slots that would
be visible to the userspace capture side. There may be some other way to
fix the logical AND behavior on the MCA side... that would make this
unnecessary.
("How does Apple deal with this"? - they don't, macOS does not use
IVSENSE on TAS2770 machines even though it's physically wired up,
but we want to do so on Linux.)
Hector Martin [Sat, 5 Apr 2025 23:15:05 +0000 (09:15 +1000)]
ASoC: tas2770: Power cycle amp on ISENSE/VSENSE change
The ISENSE/VSENSE blocks are only powered up when the amplifier
transitions from shutdown to active. This means that if those controls
are flipped on while the amplifier is already playing back audio, they
will have no effect.
Fix this by forcing a power cycle around transitions in those controls.
ASoC: soc-pcm: Fix hw_params() and DAPM widget sequence
Issue:
When multiple audio streams share a common BE DAI, the BE DAI
widget can be powered up before its hardware parameters are configured.
This incorrect sequence leads to intermittent pcm_write errors.
For example, the below Tegra use-case throws an error:
aplay(2 streams) -> AMX(mux) -> ADX(demux) -> arecord(2 streams),
here, 'AMX TX' and 'ADX RX' are common BE DAIs.
For above usecase when failure happens below sequence is observed:
aplay(1) FE open()
- BE DAI callbacks added to the list
- BE DAI state = SND_SOC_DPCM_STATE_OPEN
aplay(2) FE open()
- BE DAI callbacks are not added to the list as the state is
already SND_SOC_DPCM_STATE_OPEN during aplay(1) FE open().
aplay(2) FE hw_params()
- BE DAI hw_params() callback ignored
aplay(2) FE prepare()
- Widget is powered ON without BE DAI hw_params() call
aplay(1) FE hw_params()
- BE DAI hw_params() is now called
Fix:
Add BE DAIs in the list if its state is either SND_SOC_DPCM_STATE_OPEN
or SND_SOC_DPCM_STATE_HW_PARAMS as well.
It ensures the widget is powered ON after BE DAI hw_params() callback.
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