Merge series from Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>:
This patchset adds support for sound card on Qualcomm QCS9100 and
QCS9075 boards.
Merge series from Simon Trimmer <simont@opensource.cirrus.com>:
These two patches introduce a log message when provisioning the cs35l56
family of devices that uniquely identifies the firmware tuning.
We have below 2 functions, but these are very similar
(A) snd_soc_unregister_component_by_driver()
(B) snd_soc_unregister_component()
(A) void snd_soc_unregister_component_by_driver(...)
{
...
(a) mutex_lock(&client_mutex);
^ (X) component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
| if (!component) ^^^^^^^^^^^^^^^^^^^^^^
| goto out;
(b)
| snd_soc_del_component_unlocked(component);
v
out:
(c) mutex_unlock(&client_mutex);
}
(B) void snd_soc_unregister_component_by_driver(...)
{
(a) mutex_lock(&client_mutex);
^ while (1) {
| (X) struct snd_soc_component *component = snd_soc_lookup_component_nolocked(dev, NULL);
| ^^^^
(b) if (!component)
| break;
|
| snd_soc_del_component_unlocked(component);
v }
(c) mutex_unlock(&client_mutex);
}
Both are calling lock (a), find component and remove it (b), and
unlock (c). The big diff is whether use driver name for lookup() or
not (X).
Merge these into snd_soc_unregister_component_by_driver() (B), and
snd_soc_unregister_component_by_driver() (A) can be macro.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87h61qy2vn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_gus_use_dec(), snd_gus_use_inc() and snd_gf1_print_voice_registers()
last uses were removed in 2007 by
commit e5723b41ab ("[ALSA] Remove sequencer instrument layer")
Remove them.
While there, remove big #if 0 blocks next to the code being deleted.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Link: https://patch.msgid.link/20250508000225.195766-1-linux@treblig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hdac_stream_get_spbmaxfifo() was originally added in 2015
in commit ee8bc4df1b ("ALSA: hdac: Add support to enable SPIB for hdac
ext stream")
when it was originally called snd_hdac_ext_stream_set_spbmaxfifo,
it was renamed snd_hdac_ext_stream_get_spbmaxfifo shortly after
and was finally renamed to snd_hdac_stream_get_spbmaxfifo in 2022.
But it was never used.
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Link: https://patch.msgid.link/20250505011037.340592-1-linux@treblig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
CS35L63 is a Mono Class-D PC Smart Amplifier, with Speaker Protection
and Audio Enhancement Algorithms.
CS35L63 uses a similar control interface to CS35L56 so support for
it can be added into the CS35L56 driver.
CS35L63 only has SoundWire and I2C control interfaces.
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
SOF will load the function topologies by default. However, user may want
to use the monolithic topology. Add a flag amd a module parameter to
allow user specify the topology or not using function topologies.
Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>:
Fix a small bug that can cause the sof_sdw machine driver to fail probe
after the first time it has probed. Also do some minor tidy up on the
handling of the platform_component of the dai links.
Merge series from "Peng Fan (OSS)" <peng.fan@oss.nxp.com>:
This patchset is separate from [1], and not merging changes in one
patch. So separate changes into three patches for each chip.
- sort headers
- Drop legacy platform support
- 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_cansleep to configure output value.
I not have platforms to test, just do the patches with my best efforts,
and make build pass.
[1] https://lore.kernel.org/all/20250408-asoc-gpio-v1-0-c0db9d3fd6e9@nxp.com/
There is no in-tree user of "include/sound/cs42l52.h", so move
'struct cs42l52_platform_data ' to cs42l52.c and remove the header file.
And platform data is mostly for legacy platforms that create devices
non using device tree. So drop cs42l52.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-8-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no in-tree user of "include/sound/cs42l56.h", so move
'struct cs42l73_platform_data ' to cs42l73.c and remove the header file.
And platform data is mostly for legacy platforms that create devices
non using device tree. So drop cs42l73.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-5-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no in-tree user of "include/sound/cs42l56.h", so move
'struct cs42l56_platform_data' to cs42l56.c and remove the header file.
And platform data is mostly for platforms that create
devices non using device tree. CS42L56 is a discontinued product,
there is less possibility that new users will use legacy method
to create devices. So drop cs42l56.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-2-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Philipp Stanner <phasta@kernel.org>:
A year ago we spent quite some work trying to get PCI into better shape.
Some pci_ functions can be sometimes managed with devres, which is
obviously bad. We want to provide an obvious API, where pci_ functions
are never, and pcim_ functions are always managed.
Thus, everyone enabling his device with pcim_enable_device() must be
ported to pcim_ functions. Porting all users will later enable us to
significantly simplify parts of the PCI subsystem. See here [1] for
details.
This patch series does that for sound.
Feel free to squash the commits as you see fit.
P.
[1] https://elixir.bootlin.com/linux/v6.14-rc4/source/drivers/pci/devres.c#L18
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
We are using dummy component/dlc, but didn't have its check funciton.
This patch adds it.
ASoC: Fixes for v6.15
A moderately large batch of fixes for v6.15, many driver specific
including cleanups for the enabling of the Cirrus KUnit tests and a fix
for a nasty crash on resume on AMD systems. We also have one core fix,
for an ordering issue between DAPM and DPCM which could leave things
incorrectly unpowered.
The volume control for cs35l56 speakers has a maximum gain of +12 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the case of systems which use the soc_sdw_* driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 400, which is 0 dB maximum volume inside this driver.
The volume limit should be applied to both soundwire and soundwire
bridge configurations.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The conversion function from MIDI 1.0 to UMP packet contains an
internal buffer to keep the incoming MIDI bytes, and its size is 4, as
it was supposed to be the max size for a MIDI1 UMP packet data.
However, the implementation overlooked that SysEx is handled in a
different format, and it can be up to 6 bytes, as found in
do_convert_to_ump(). It leads eventually to a buffer overflow, and
may corrupt the memory when a longer SysEx message is received.
The fix is simply to extend the buffer size to 6 to fit with the SysEx
UMP message.
Fixes: 0b5288f5fe ("ALSA: ump: Add legacy raw MIDI support")
Reported-by: Argusee <vr@darknavy.com>
Link: https://patch.msgid.link/20250429124845.25128-1-tiwai@suse.de
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.
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>
Create a USB BE component that will register a new USB port to the ASoC USB
framework. This will handle determination on if the requested audio
profile is supported by the USB device currently selected.
Check for if the PCM format is supported during the hw_params callback. If
the profile is not supported then the userspace ALSA entity will receive an
error, and can take further action.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-25-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
