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.
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>
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.
Fixes: e2bcbf99d0 ("ASoC: cs48l32: Add driver for Cirrus Logic CS48L32 audio DSP")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250418-cs48l32-modern-pm_ops-v1-1-640559407619@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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).
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.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250416122422.783215-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
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.
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.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: "Rob Herring (Arm)" <robh@kernel.org>
Link: https://patch.msgid.link/20250415115016.505777-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87fribr2ns.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
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.
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.
[1] https://github.com/AsahiLinux/linux/tree/asahi-wip
[2] https://stats.asahilinux.org/
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.
Co-developed-by: Martin Povišer <povik+lin@cutebit.org>
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-v5-8-50a00ec850a3@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
