Commit Graph

52679 Commits

Author SHA1 Message Date
Florian Meier
9e9d0643e8 ASoC: Add support for all the downstream rpi sound card drivers
ASoC: Add support for Rpi-DAC

ASoC: Add prompt for ICS43432 codec

Without a prompt string, a config setting can't be included in a
defconfig. Give CONFIG_SND_SOC_ICS43432 a prompt so that Pi soundcards
can use the driver.

Signed-off-by: Phil Elwell <phil@raspberrypi.org>

Add IQaudIO Sound Card support for Raspberry Pi

Set a limit of 0dB on Digital Volume Control

The main volume control in the PCM512x DAC has a range up to
+24dB. This is dangerously loud and can potentially cause massive
clipping in the output stages. Therefore this sets a sensible
limit of 0dB for this control.

Allow up to 24dB digital gain to be applied when using IQAudIO DAC+

24db_digital_gain DT param can be used to specify that PCM512x
codec "Digital" volume control should not be limited to 0dB gain,
and if specified will allow the full 24dB gain.

Modify IQAudIO DAC+ ASoC driver to set card/dai config from dt

Add the ability to set the card name, dai name and dai stream name, from
dt config.

Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk>

IQaudIO: auto-mute for AMP+ and DigiAMP+

IQAudIO amplifier mute via GPIO22. Add dt params for "one-shot" unmute
and auto mute.

Revision 2, auto mute implementing HiassofT suggestion to mute/unmute
using set_bias_level, rather than startup/shutdown....
"By default DAPM waits 5 seconds (pmdown_time) before shutting down
playback streams so a close/stop immediately followed by open/start
doesn't trigger an amp mute+unmute."

Tested on both AMP+ (via DAC+) and DigiAMP+, with both options...

dtoverlay=iqaudio-dacplus,unmute_amp
 "one-shot" unmute when kernel module loads.

dtoverlay=iqaudio-dacplus,auto_mute_amp
 Unmute amp when ALSA device opened by a client. Mute, with 5 second delay
 when ALSA device closed. (Re-opening the device within the 5 second close
 window, will cancel mute.)

Revision 4, using gpiod.

Revision 5, clean-up formatting before adding mute code.
 - Convert tab plus 4 space formatting to 2x tab
 - Remove '// NOT USED' commented code

Revision 6, don't attempt to "one-shot" unmute amp, unless card is
successfully registered.

Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk>

ASoC: iqaudio-dac: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: iqaudio-dac: use modern dai_link style

Signed-off-by: Matthias Reichl <hias@horus.com>

Added support for HiFiBerry DAC+

The driver is based on the HiFiBerry DAC driver. However HiFiBerry DAC+ uses
a different codec chip (PCM5122), therefore a new driver is necessary.

Add support for the HiFiBerry DAC+ Pro.

The HiFiBerry DAC+ and DAC+ Pro products both use the existing bcm sound driver with the DAC+ Pro having a special clock device driver representing the two high precision oscillators.

An addition bug fix is included for the PCM512x codec where by the physical size of the sample frame is used in the calculation of the LRCK divisor as it was found to be wrong when using 24-bit depth sample contained in a little endian 4-byte sample frame.

Limit PCM512x "Digital" gain to 0dB by default with HiFiBerry DAC+

24db_digital_gain DT param can be used to specify that PCM512x
codec "Digital" volume control should not be limited to 0dB gain,
and if specified will allow the full 24dB gain.

Add dt param to force HiFiBerry DAC+ Pro into slave mode

"dtoverlay=hifiberry-dacplus,slave"

Add 'slave' param to use HiFiBerry DAC+ Pro in slave mode,
with Pi as master for bit and frame clock.

Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk>

Fixed a bug when using 352.8kHz sample rate

Signed-off-by: Daniel Matuschek <daniel@hifiberry.com>

ASoC: pcm512x: revert downstream changes

This partially reverts commit 185ea05465
which was added by https://github.com/raspberrypi/linux/pull/1152

The downstream pcm512x changes caused a regression, it broke normal
use of the 24bit format with the codec, eg when using simple-audio-card.

The actual bug with 24bit playback is the incorrect usage
of physical_width in various drivers in the downstream tree
which causes 24bit data to be transmitted with 32 clock
cycles. So it's not the pcm512x that needs fixing, it's the
soundcard drivers.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: hifiberry_dacplus: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: hifiberry_dacplus: transmit S24_LE with 64 BCLK cycles

Signed-off-by: Matthias Reichl <hias@horus.com>

hifiberry_dacplus: switch to snd_soc_dai_set_bclk_ratio

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: hifiberry_dacplus: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add driver for rpi-proto

Forward port of 3.10.x driver from https://github.com/koalo
We are using a custom board and would like to use rpi 3.18.x
kernel. Patch works fine for our embedded system.

URL to the audio chip:
http://www.mikroe.com/add-on-boards/audio-voice/audio-codec-proto/

Playback tested with devicetree enabled.

Signed-off-by: Waldemar Brodkorb <wbrodkorb@conet.de>

ASoC: rpi-proto: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add Support for JustBoom Audio boards

justboom-dac: Adjust for ALSA API change

As of 4.4, snd_soc_limit_volume now takes a struct snd_soc_card *
rather than a struct snd_soc_codec *.

Signed-off-by: Phil Elwell <phil@raspberrypi.org>

ASoC: justboom-dac: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Also remove hw_params as it's no longer needed.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: justboom-dac: use modern dai_link style

Signed-off-by: Matthias Reichl <hias@horus.com>

New AudioInjector.net Pi soundcard with low jitter audio in and out.

Contains the sound/soc/bcm ALSA machine driver and necessary alterations to the Kconfig and Makefile.
Adds the dts overlay and updates the Makefile and README.
Updates the relevant defconfig files to enable building for the Raspberry Pi.
Thanks to Phil Elwell (pelwell) for the review, simple-card concepts and discussion. Thanks to Clive Messer for overlay naming suggestions.

Added support for headphones, microphone and bclk_ratio settings.

This patch adds headphone and microphone capability to the Audio Injector sound card. The patch also sets the bit clock ratio for use in the bcm2835-i2s driver. The bcm2835-i2s can't handle an 8 kHz sample rate when the bit clock is at 12 MHz because its register is only 10 bits wide which can't represent the ch2 offset of 1508. For that reason, the rate constraint is added.

ASoC: audioinjector-pi-soundcard: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

New driver for RRA DigiDAC1 soundcard using WM8741 + WM8804

ASoC: digidac1-soundcard: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add support for Dion Audio LOCO DAC-AMP HAT

Using dedicated machine driver and pcm5102a codec driver.

Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk>

ASoC: dionaudio_loco: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Allo Piano DAC boards: Initial 2 channel (stereo) support (#1645)

Add initial 2 channel (stereo) support for Allo Piano DAC (2.0/2.1) boards,
using allo-piano-dac-pcm512x-audio overlay and allo-piano-dac ALSA ASoC
machine driver.

NB. The initial support is 2 channel (stereo) ONLY!
(The Piano DAC 2.1 will only support 2 channel (stereo) left/right output,
 pending an update to the upstream pcm512x codec driver, which will have
 to be submitted via upstream. With the initial downstream support,
 provided by this patch, the Piano DAC 2.1 subwoofer outputs will
 not function.)

Signed-off-by: Baswaraj K <jaikumar@cem-solutions.net>
Signed-off-by: Clive Messer <clive.messer@digitaldreamtime.co.uk>
Tested-by: Clive Messer <clive.messer@digitaldreamtime.co.uk>

ASoC: allo-piano-dac: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Also remove hw_params and ops as they are no longer needed.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: allo-piano-dac: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add support for Allo Piano DAC 2.1 plus add-on board for Raspberry Pi.

The Piano DAC 2.1 has support for 4 channels with subwoofer.

Signed-off-by: Baswaraj K <jaikumar@cem-solutions.net>
Reviewed-by: Vijay Kumar B. <vijaykumar@zilogic.com>
Reviewed-by: Raashid Muhammed <raashidmuhammed@zilogic.com>

Add clock changes and mute gpios (#1938)

Also improve code style and adhere to ALSA coding conventions.

Signed-off-by: Baswaraj K <jaikumar@cem-solutions.net>
Reviewed-by: Vijay Kumar B. <vijaykumar@zilogic.com>
Reviewed-by: Raashid Muhammed <raashidmuhammed@zilogic.com>

PianoPlus: Dual Mono & Dual Stereo features added (#2069)

allo-piano-dac-plus: Master volume added + fixes

Master volume added, which controls both DACs volumes.

See: https://github.com/raspberrypi/linux/pull/2149

Also fix initial max volume, default mode value, and unmute.

Signed-off-by: allocom <sparky-dev@allo.com>

ASoC: allo-piano-dac-plus: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Signed-off-by: Matthias Reichl <hias@horus.com>

sound: bcm: Fix memset dereference warning

This warning appears with GCC 6.4.0 from toolchains.bootlin.com:

../sound/soc/bcm/allo-piano-dac-plus.c: In function ‘snd_allo_piano_dac_init’:
../sound/soc/bcm/allo-piano-dac-plus.c:711:30: warning: argument to ‘sizeof’ in ‘memset’ call is the same expression as the destination; did you mean to dereference it? [-Wsizeof-pointer-memaccess]
  memset(glb_ptr, 0x00, sizeof(glb_ptr));
                              ^

Suggested-by: Phil Elwell <phil@raspberrypi.org>
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>

ASoC: allo-piano-dac-plus: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add support for Allo Boss DAC add-on board for Raspberry Pi. (#1924)

Signed-off-by: Baswaraj K <jaikumar@cem-solutions.net>
Reviewed-by: Deepak <deepak@zilogic.com>
Reviewed-by: BabuSubashChandar <babusubashchandar@zilogic.com>

Add support for new clock rate and mute gpios.

Signed-off-by: Baswaraj K <jaikumar@cem-solutions.net>
Reviewed-by: Deepak <deepak@zilogic.com>
Reviewed-by: BabuSubashChandar <babusubashchandar@zilogic.com>

ASoC: allo-boss-dac: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: allo-boss-dac: transmit S24_LE with 64 BCLK cycles

Signed-off-by: Matthias Reichl <hias@horus.com>

allo-boss-dac: switch to snd_soc_dai_set_bclk_ratio

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: allo-boss-dac: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Support for Blokas Labs pisound board

Pisound dynamic overlay (#1760)

Restructuring pisound-overlay.dts, so it can be loaded and unloaded dynamically using dtoverlay.

Print a logline when the kernel module is removed.

pisound improvements:

* Added a writable sysfs object to enable scripts / user space software
to blink MIDI activity LEDs for variable duration.
* Improved hw_param constraints setting.
* Added compatibility with S16_LE sample format.
* Exposed some simple placeholder volume controls, so the card appears
in volumealsa widget.

Add missing SND_PISOUND selects dependency to SND_RAWMIDI

Without it the Pisound module fails to compile.
See https://github.com/raspberrypi/linux/issues/2366

Updates for Pisound module code:

	* Merged 'Fix a warning in DEBUG builds' (1c8b82b).
	* Updating some strings and copyright information.
	* Fix for handling high load of MIDI input and output.
	* Use dual rate oversampling ratio for 96kHz instead of single
	  rate one.

Signed-off-by: Giedrius Trainavicius <giedrius@blokas.io>

Fixing memset call in pisound.c

Signed-off-by: Giedrius Trainavicius <giedrius@blokas.io>

Fix for Pisound's MIDI Input getting blocked for a while in rare cases.

There was a possible race condition which could lead to Input's FIFO queue
to be underflown, causing high amount of processing in the worker thread for
some period of time.

Signed-off-by: Giedrius Trainavicius <giedrius@blokas.io>

Fix for Pisound kernel module in Real Time kernel configuration.

When handler of data_available interrupt is fired, queue_work ends up
getting called and it can block on a spin lock which is not allowed in
interrupt context. The fix was to run the handler from a thread context
instead.

Pisound: Remove spinlock usage around spi_sync

ASoC: pisound: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

ASoC: pisound: fix the parameter for spi_device_match

Signed-off-by: Hui Wang <hui.wang@canonical.com>

ASoC: Add driver for Cirrus Logic Audio Card

Note: due to problems with deferred probing of regulators
the following softdep should be added to a modprobe.d file

softdep arizona-spi pre: arizona-ldo1

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: rpi-cirrus: use modern dai_link style

Signed-off-by: Matthias Reichl <hias@horus.com>

sound: Support for Dion Audio LOCO-V2 DAC-AMP HAT

Signed-off-by: Miquel Blauw <info@dionaudio.nl>

ASoC: dionaudio_loco-v2: fix S24_LE format

Remove set_bclk_ratio call so 24-bit data is transmitted in
24 bclk cycles.

Also remove hw_params and ops as they are no longer needed.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: dionaudio_loco-v2: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add support for Fe-Pi audio sound card. (#1867)

Fe-Pi Audio Sound Card is based on NXP SGTL5000 codec.
Mechanical specification of the board is the same the Raspberry Pi Zero.
3.5mm jacks for Headphone/Mic, Line In, and Line Out.

Signed-off-by: Henry Kupis <fe-pi@cox.net>

ASoC: fe-pi-audio: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Add support for the AudioInjector.net Octo sound card

AudioInjector Octo: sample rates, regulators, reset

This patch adds new sample rates to the Audioinjector Octo sound card. The
new supported rates are (in kHz) :
96, 48, 32, 24, 16, 8, 88.2, 44.1, 29.4, 22.05, 14.7

Reference the bcm270x DT regulators in the overlay.

This patch adds a reset GPIO for the AudioInjector.net octo sound card.

Audioinjector octo : Make the playback and capture symmetric

This patch ensures that the sample rate and channel count of the audioinjector
octo sound card are symmetric.

audioinjector-octo: Add continuous clock feature

By user request, add a switch to prevent the clocks being stopped when
the stream is paused, stopped or shutdown. Provide access to the switch
by adding a 'non-stop-clocks' parameter to the audioinjector-addons
overlay.

See: https://github.com/raspberrypi/linux/issues/2409

Signed-off-by: Phil Elwell <phil@raspberrypi.org>

sound: Fixes for audioinjector-octo under 4.19

1. Move the DT alias declaration to the I2C shim in the cases
where the shim is enabled. This works around a problem caused by a
4.19 commit [1] that generates DT/OF uevents for I2C drivers.

2. Fix the diagnostics in an error path of the soundcard driver to
correctly identify the reason for the failure to load.

3. Move the declaration of the clock node in the overlay outside
the I2C node to avoid warnings.

4. Sort the overlay nodes so that dependencies are only to earlier
fragments, in an attempt to get runtime dtoverlay application to
work (it still doesn't...)

See: https://github.com/Audio-Injector/Octo/issues/14
Signed-off-by: Phil Elwell <phil@raspberrypi.org>

[1] af503716ac ("i2c: core: report OF style module alias for devices registered via OF")

ASoC: audioinjector-octo-soundcard: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Driver support for Google voiceHAT soundcard.

ASoC: googlevoicehat-codec: Use correct device when grabbing GPIO

The fixup for the VoiceHAT in 4.18 incorrectly tried to find the
sdmode GPIO pin under the card device, not the codec device.
This failed, and therefore caused the device probe to fail.

Signed-off-by: Dave Stevenson <dave.stevenson@raspberrypi.org>

ASoC: googlevoicehat-codec: Reformat for kernel coding standards

Fix all whitespace, indentation, and bracing errors.

Signed-off-by: Dave Stevenson <dave.stevenson@raspberrypi.org>

ASoC: googlevoicehat-codec: Make driver function structure const

Make voicehat_component_driver a const structure.

Signed-off-by: Dave Stevenson <dave.stevenson@raspberrypi.org>

ASoC: googlevoicehat-codec: Only convert from ms to jiffies once

Minor optimisation and allows to become checkpatch clean.
A msec value is read out of DT or from a define, and convert once to
jiffies, rather than every time that it is used.

Signed-off-by: Dave Stevenson <dave.stevenson@raspberrypi.org>

Driver and overlay for Allo Katana DAC

Allo Katana DAC: Updated default values

Signed-off-by: Jaikumar <jaikumar@cem-solutions.com>

Added mute stream func

Signed-off-by: Jaikumar <jaikumar@cem-solutions.net>

codecs: Correct Katana minimum volume

Update Katana minimum volume to get the exact 0.5 dB value in each step.

Signed-off-by: Sudeep Kumar <sudeepkumar@cem-solutions.net>

ASoC: Add generic RPI driver for simple soundcards.

The RPI simple sound card driver provides a generic ALSA SOC card driver
supporting a variety of Pi HAT soundcards. The intention is to avoid
the duplication of code for cards that can't be fully supported by
the soc simple/graph cards but are otherwise almost identical.

This initial commit adds support for the ADAU1977 ADC, Google VoiceHat,
HifiBerry AMP, HifiBerry DAC and RPI DAC.

Signed-off-by: Tim Gover <tim.gover@raspberrypi.org>

ASoC: Use correct card name in rpi-simple driver

Use the specific card name from drvdata instead of the snd_rpi_simple

rpi-simple-soundcard: Use nicer driver name "RPi-simple"

Rename the driver from "RPI simple soundcard" to "RPi-simple" so that
the driver name won't be mangled allowing to be used unaltered as the
card conf filename.

ASoC: rpi-simple-soundcard: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

ASoC: Add Kconfig and Makefile for sound/soc/bcm

Signed-off-by: popcornmix <popcornmix@gmail.com>

ASoC: Create a generic Pi Hat WM8804 driver

Reduce the amount of duplicated code by creating a generic driver for
Pi Hat digi cards using the WM8804 codec.

This replaces the
Allo DigiOne, Hifiberry Digi/Pro, JustBoom Digi and IQAudIO Digi
dedicate soundcard drivers with a generic driver.

There are no significant changes to the runtime behavior of the drivers
and end users should not have to change any configuration settings
after upgrading.

Minor changes
* Check the return value of snd_soc_component_update_bits
* Added some pr_debug tracing
* Various checkpatch tidyups
* Updated allodigi-one to use use 128FS at > 96 Khz. This appears to
  be an omission in the original driver code so followed the Hifiberry
  DAC driver approach.

ASoC: rpi-wm8804-soundcard: use modern dai_link style

Signed-off-by: Matthias Reichl <hias@horus.com>

rpi-wm8804-soundcard: drop PWRDN register writes

Since kernel 4.0 the PWRDN register bits are under DAPM
control from the wm8804 driver.

Drop code that modifies that register to avoid interfering
with DAPM.

Signed-off-by: Matthias Reichl <hias@horus.com>

rpi-wm8804-soundcard: configure wm8804 clocks only on rate change

This should avoid clicks when stopping and immediately afterwards
starting a stream with the same samplerate as before.

Signed-off-by: Matthias Reichl <hias@horus.com>

rpi-wm8804-soundcard: Fixed MCLKDIV for Allo Digione

The Allo Digione board wants a fixed MCLKDIV of 256.

See: https://github.com/raspberrypi/linux/issues/3296

Signed-off-by: Phil Elwell <phil@raspberrypi.org>

ASoC: Add support for AudioSense-Pi add-on soundcard

AudioSense-Pi is a RPi HAT based on a TI's TLV320AIC32x4 stereo codec

This hardware provides multiple audio I/O capabilities to the RPi.
The codec connects to the RPi's SoC through the I2S Bus.

The following devices can be connected through a 3.5mm jack
	1. Line-In: Plain old audio in from mobile phones, PCs, etc.,
	2. Mic-In: Connect a microphone
	3. Line-Out: Connect the output to a speaker
	4. Headphones: Connect a Headphone w or w/o microphones

Multiple Inputs:
	It supports the following combinations
	1. Two stereo Line-Inputs and a microphone
	2. One stereo Line-Input and two microphones
	3. Two stereo Line-Inputs, a microphone and
		one mono line-input (with h/w hack)
	4. One stereo Line-Input, two microphones and
		one mono line-input (with h/w hack)

Multiple Outputs:
	Audio output can be routed to the headphones or
		speakers (with additional hardware)

Signed-off-by: b-ak <anur.bhargav@gmail.com>

ASoC: audiosense-pi: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Added driver for the HiFiBerry DAC+ ADC (#2694)

Signed-off-by: Daniel Matuschek <daniel@hifiberry.com>

hifiberry_dacplusadc: switch to snd_soc_dai_set_bclk_ratio

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: hifiberry_dacplusadc: fix DAI link setup

The driver only defines a single DAI link and the code that tries
to setup the second (non-existent) DAI link looks wrong - using dmic
as a CPU/platform driver doesn't make any sense.

The DT overlay doesn't define a dmic property, so the code was never
executed (otherwise it would have resulted in a memory corruption).

So drop the offending code to prevent issues if a dmic property
should be added to the DT overlay.

Signed-off-by: Matthias Reichl <hias@horus.com>

ASoC: hifiberry_dacplusadc: use modern dai_link style

Signed-off-by: Matthias Reichl <hias@horus.com>

Audiophonics I-Sabre 9038Q2M DAC driver

Signed-off-by: Audiophonics <contact@audiophonics.fr>

ASoC: i-sabre-q2m: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

Added IQaudIO Pi-Codec board support (#2969)

Add support for the IQaudIO Pi-Codec board.

Signed-off-by: Gordon <gordon@iqaudio.com>

Fixed 48k timing issue

ASoC: iqaudio-codec: use modern dai_link style

Signed-off-by: Hui Wang <hui.wang@canonical.com>

adds the Hifiberry DAC+ADC PRO version

This adds the driver for the DAC+ADC PRO version of the Hifiberry soundcard with software controlled PCM1863 ADC
Signed-off-by: Joerg Schambacher joerg@i2audio.com

Add Hifiberry DAC+DSP soundcard driver (#3224)

Adds the driver for the Hifiberry DAC+DSP. It supports capture and
playback depending on the DSP firmware.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

Allow simultaneous use of JustBoom DAC and Digi

Signed-off-by: Johannes Krude <johannes@krude.de>

Pisound: MIDI communication fixes for scaled down CPU.

* Increased maximum SPI communication speed to avoid running too slow
  when the CPU is scaled down and losing MIDI data.

* Keep track of buffer usage in millibytes for higher precision.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

sound: Add the HiFiBerry DAC+HD version

This adds the driver for the DAC+HD version supporting HiFiBerry's
PCM179x based DACs. It also adds PLL control for clock generation.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

Fix master mode settings of HiFiBerry DAC+ADC PRO card (#3424)

This patch fixes the board DAI setting when in master-mode.
Wrong setting could have caused random pop noise.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

adds LED OFF feature to HiFiBerry DAC+ADC PRO sound card

This adds a DT overlay parameter 'leds_off' which allows
to switch off the onboard activity LEDs at all times
which has been requested by some users.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

adds LED OFF feature to HiFiBerry DAC+ADC sound card

This adds a DT overlay parameter 'leds_off' which allows
to switch off the onboard activity LEDs at all times
which has been requested by some users.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

adds LED OFF feature to HiFiBerry DAC+/DAC+PRO sound cards

This adds a DT overlay parameter 'leds_off' which allows
to switch off the onboard activity LEDs at all times
which has been requested by some users.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

pisound: Added reading Pisound board hardware revision and exposing it (#3425)

pisound: Added reading Pisound board hardware revision and exposing it in kernel log and sysfs file:

/sys/kernel/pisound/hw_version

Signed-off-by: Giedrius <giedrius@blokas.io>

Added driver for HiFiBerry Amp amplifier add-on board

The driver contains a low-level hardware driver for the TAS5713 and the
drivers for the Raspberry Pi I2S subsystem.

TAS5713: return error if initialisation fails

Existing TAS5713 driver logs errors during initialisation, but does not return
an error code. Therefore even if initialisation fails, the driver will still be
loaded, but won't work. This patch fixes this. I2C communication error will now
reported correctly by a non-zero return code.

HiFiBerry Amp: fix device-tree problems

Some code to load the driver based on device-tree-overlays was missing. This is added by this patch.

According to 5713 pdf doc CLOCK_CTRL is a readonly status register, and it behaves so. Remove useless setting

sound: pcm512x-codec: Adding 352.8kHz samplerate support

sound/soc: only first codec is master in multicodec setup

When using multiple codecs, at most one codec should generate the master
clock. All codecs except the first are therefore configured for slave
mode.

Signed-off-by: Johannes Krude <johannes@krude.de>

ASoC: Fix snd_soc_get_pcm_runtime usage

Commit [1] changed the snd_soc_get_pcm_runtime to take a dai_link
pointer instead of a string. Patch up the downstream drivers to use
the modified API.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

[1] 4468189ff3 ("ASoC: soc-core: find rtd via dai_link pointer at snd_soc_get_pcm_runtime()")

Add support for the AudioInjector.net Isolated sound card

This patch adds support for the Audio Injector Isolated sound card.

Signed-off-by: Matt Flax <flatmax@flatmax.org>

Add support for merus-amp soundcard and ma120x0p codec

Add 96KHz rate support to MA120X0P codec and make enable and mute gpio
pins optional.

Signed-off-by: AMuszkat <ariel.muszkat@gmail.com>

Fixes a problem with clock settings of HiFiBerry DAC+ADC PRO (#3545)

This patch fixes a problem of the re-calculation of
i2s-clock and -parameter settings when only the ADC is activated.

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

configs: Enable the AD193x codecs

See: https://github.com/raspberrypi/linux/issues/2850

Signed-off-by: Phil Elwell <phil@raspberrypi.org>

Switch to snd_soc_dai_set_bclk_ratio
Replaces obsolete function snd_soc_dai_set_tdm_slot

Signed-off-by: Joerg Schambacher <joerg@i2audio.com>

Enhances the DAC+ driver to control the optional headphone amplifier

Probes on the I2C bus for TPA6130A2, if successful, it sets DT-parameter
'status' from 'disabled' to 'okay' using change_sets to enable
the headphone control.

Signed-off-by: Joerg Schambacher joerg@i2audio.com

Update Allo Piano Dac Driver

Add unique names to the individual dac coded drivers
Remove some of the codec controls that are not used.

Signed-off-by: Paul Hermann <paul@picoreplayer.org>

Fixes an onboard clock detection problem of the PRO versions

Increasing the sleep time after clock selection to 3-4ms
allows the correct detection of all combinations of DAC+ Pro
and DAC+ADC Pro sound cards and the various PI revisions.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

ASoC:ma120x0p: Increase maximum sample rate to 192KHz

Change the maximum sample rate for the amplifier to
192KHz as given in the Infineon specification.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

ASoC: ma120x0p: Remove unnecessary const specifier

Clang warns:

  sound/soc/codecs/ma120x0p.c:891:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier]
  static const SOC_VALUE_ENUM_SINGLE_DECL(pwr_mode_ctrl,
               ^
  ./include/sound/soc.h:362:2: note: expanded from macro 'SOC_VALUE_ENUM_SINGLE_DECL'
          SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
          ^
  ./include/sound/soc.h:359:2: note: expanded from macro 'SOC_VALUE_ENUM_DOUBLE_DECL'
          const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
          ^
  1 warning generated.

SOC_VALUE_ENUM_DOUBLE_DECL already has a const specifier. Remove the duplicate
const to clean up the warning.

Fixes: 42444979e7 ("Add support for all the downstream rpi sound card drivers")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>

ASoC: bcm: allo-piano-dac-plus: Remove unnecessary const specifiers

Clang warns:

  sound/soc/bcm/allo-piano-dac-plus.c:66:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier]
  static const SOC_ENUM_SINGLE_DECL(allo_piano_mode_enum,
               ^
  ./include/sound/soc.h:355:2: note: expanded from macro 'SOC_ENUM_SINGLE_DECL'
          SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
          ^
  ./include/sound/soc.h:352:2: note: expanded from macro 'SOC_ENUM_DOUBLE_DECL'
          const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
          ^
  sound/soc/bcm/allo-piano-dac-plus.c:75:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier]
  static const SOC_ENUM_SINGLE_DECL(allo_piano_dual_mode_enum,
               ^
  ./include/sound/soc.h:355:2: note: expanded from macro 'SOC_ENUM_SINGLE_DECL'
          SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
          ^
  ./include/sound/soc.h:352:2: note: expanded from macro 'SOC_ENUM_DOUBLE_DECL'
          const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
          ^
  sound/soc/bcm/allo-piano-dac-plus.c:96:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier]
  static const SOC_ENUM_SINGLE_DECL(allo_piano_enum,
               ^
  ./include/sound/soc.h:355:2: note: expanded from macro 'SOC_ENUM_SINGLE_DECL'
          SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
          ^
  ./include/sound/soc.h:352:2: note: expanded from macro 'SOC_ENUM_DOUBLE_DECL'
          const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
          ^
  3 warnings generated.

SOC_VALUE_ENUM_DOUBLE_DECL already has a const specifier. Remove the duplicate
const specifiers to clean up the warnings.

Fixes: 42444979e7 ("Add support for all the downstream rpi sound card drivers")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>

rpi-simple-soundcard: Add Dion Audio KIWI streamer

Signed-off-by: Miquel Blauw <miquelblauw@hotmail.com>

rpi-simple-soundcard: adds definitions for the HiFiBerry AMP3 card

Uses Infineon MA120x0 amplifier and supports full sample rate of 192ksps.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

sound: soc: bcm: Added Sound card driver for Dacberry400 Audio card for Raspberry Pi 400

Added Sound card driver for DACberry400 Audio card.

Signed-off-by: Ashish Vara <ashishhvara@gmail.com>

ASoC:ma120x0p: Corrects the volume level display

Fixes the wrongly changed 'limiter volume' display back to -50dB minimum
and sets the correct minimum volume level to -144dB to be aligned with
the controls and display in alsamixer etc.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

ASoC: bcm: Fix Rpi-PROTO and audioinjector.net Pi

As of kernel 5.19 the WM8731 driver has separate I2C and SPI support
modules. Change the Kconfig definitions for the audioinjector.net Pi
and Rpi-PROTO soundcards to select SND_SOC_WM8731_I2C.

See: https://github.com/raspberrypi/linux/issues/5364

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: adau1977: Add correct compatible strings

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: bcm2835-i2s: Use phys addresses for DAI DMA

Contrary to what struct snd_dmaengine_dai_dma_data suggests, the
configuration of addresses of DMA slave interfaces should be done in
CPU physical addresses.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

rpi sound cards: Fix Codec Zero rate switching

The Raspberry Pi Codec Zero (and IQaudIO Codec) don't notify the DA7213
codec when it needs to change PLL frequencies. As a result, audio can
be played at the wrong rate - play a 48kHz sound immediately after a
44.1kHz sound to see the effect, but in some configurations the codec
can lock into the wrong state and always get some rates wrong.

Add the necessary notification to fix the issue.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: dwc: Support set_bclk_ratio

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: dwc: Add DMACR handling

Add control of the DMACR register, which is required for paced DMA
(i.e. DREQ) support.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASOC: dwc: Improve DMA shutdown

Disabling the I2S interface with outstanding transfers prevents the
DMAC from shutting down, so keep it partially active after a stop.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASOC: dwc: Fix 16-bit audio handling

IMO the Synopsys datasheet could be clearer in this area, but it seems
that the DMA data ports (DMATX and DMARX) expect left and right samples
in alternate writes; if a stereo pair is pushed in a single 32-bit
write, the upper half is ignored, leading to double speed audio with a
confused stereo image. Make sure the necessary changes happen by
updating the DMA configuration data in the hw_params method.

The set_bclk_ratio change was made at a time when it looked like it
could be causing an error, but I think the division of responsibilities
is clearer this way (and the kernel log clearer without the info-level
message).

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: bcm: Remove dependency on BCM2835 I2S

These soundcard drivers don't rely on a specific I2S interface, so
remove the dependency declarations.

See: https://github.com/raspberrypi/linux-2712/issues/111

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: bcm: audioinjector_octo: Add soundcard "owner"

See: https://github.com/raspberrypi/linux/issues/5697

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

Pisound: Don't export the button GPIO via sysfs GPIO class.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

Pisound: Read out the SPI speed to use from the Device Tree.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

ASoC: DACplus - fix 16bit sample support in clock consumer mode

The former code did not adjust the physical sample width when
in clock consumer mode and has taken the fixed 32 bit default.
This has caused the audio to be played at half its frequency due to
the fixed bclk_ratio of 64.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

ASoC: adds support for AMP4 Pro to the DAC Plus driver

The AMP4 Pro is a I2S master mode capable amplifier with
clean onboard clock generators.
We can share the card driver between TAS575x amplifiers
and the PCM512x DACs as they are SW compatible.
From a HW perspective though we need to limit the sample
rates to the standard audio rates to avoid running the
onboard clocks through the PLL. Using the PLL would require
even a different HW.
DAI/stream name are also set accordingly to allow the user
a convenient identification of the soundcard

Needs the pcm512x driver with TAS575x support (already in
upstream kernel).

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

ASoC: DACplusADCPro - fix 16bit sample support in clock consumer mode

The former code did not adjust the physical sample width when in
clock consumer mode and has taken the fixed 32 bit default. This
has caused the audio to be played at half its frequency due to
the fixed bclk_ratio of 64.

Problem appears only on PI5 as on the former PIs the I2S module
did simply run at fixed 64x rate.

Signed-off-by: Joerg Schambacher <joerg@hifiberry.com>

Impliment driver support for Interlude Audio Digital Hat

Implementing driver support for
Interlude audio's WM8805 based digital hat
by leveraging existing drivers

ASOc: Add HiFiBerry DAC8X to the simple card driver

Defines the settings for the 8 channel version of the standard
DAC by overwriting the number of channels in the DAI defs.
It can run in 8ch mode only on PI5 using the 4 lane data output
of the designware I2S0 module.

Signed-off-by: j-schambacher <joerg@hifiberry.com>

ASoC: bcm: Use the correct sample width value

ALSA's concept of the physical width of a sample is how much memory it
occupies, including any padding. This not the same as the count of bits
of actual sample content. In particular, S24_LE has a width of 24 bits
but a physical width of 32 bits because there is a byte of padding with
each sample.

When calculating bclk_ratio, etc., it is width that matters, not
physical width. Correct the error that has been replicated across the
drivers for many Raspberry Pi-compatible soundcards.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: dwc: Correct channel count reporting

The DWC I2S driver treats the channel count register values as if they
encode a power of two (2, 4, 8, 16), but they actually encode a
multiple of 2 (2, 4, 6, 8).

Also improve the error message when asked for an unsupported number
of channels.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: Fix 16bit sample support for Hifiberry DACplusADC

Same issue as #5919.
'width' needs to be set independent of clocking mode.

Signed-off-by: j-schambacher <joerg@hifiberry.com>

allo-boss-dac mute output when changing parameters

Since I noticed that sometimes changing sample rates causes some digital
quirks and noises, I've changed the function to mute the output before
performing the changes and then unmute it when an error occurs or the
parameters got set.

Signed-off-by: Alessandro Marcon <marconalessandro04@gmail.com>

ASoC: bcm: Use power-of-2 bclk_ratios

The soundcard drivers originally used snd_pcm_format_physical_width,
but a later commit changed that to snd_pcm_format_width because the
in-memory sample storage width should not be a factor in determining
the bclk_ratio. However, the physical width rounds the sample bits up
to the nearest power of 2, which makes it easier to find integer clock
divisors.

Restore the old behaviour, but with an implementation that makes it
clear what is going on.

See: https://github.com/raspberrypi/linux/issues/6104

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: bcm: Add "owner" info for more soundcards

See: https://github.com/raspberrypi/linux/issues/5697

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: da7213: Add a set_bclk_ratio method

Following [1], it becomes harder for the CPU DAI to know the correct
BCLK ratio. We can either bake the same knowledge into the sound card
driver, or implement and use set_bclk_ratio on the codec. This commit
does the latter.

[1] commit c89e652e84 ("ASoC: da7213: Add support for mono, set
frame width to 32 when possible")

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

iqaudio-codec: Use the codec's new set_bclk_ratio

To ensure that the CPU DAI and codec agree over the BCLK ratio, impose
a fixed value of 64 on both of them.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: add driver for new HiFiBerry ADC only board(s)

Adds the driver for the soon to be released first ADC only board.
It includes the same ADC controls as used by the DAC+ADC Pro driver.

Signed-off-by: j-schambacher <joerg@hifiberry.com>

ASoC: add HiFiBerry ADC8x 8-channel ADC to simple-card-driver

Definitions for the 8 channel ADC card. The card uses only
HW-controlled devices which allows the uses of the 'dummy-dai'.
It will run only on a PI5 as it requires the designware I2S0 module.

The necessary output lanes I2S0_DI[0..3] are claimed from within the
DT overlay.

Signed-off-by: j-schambacher <joerg@hifiberry.com>

sound/soc: dwc-i2s: choose FIFO thresholds based on DMA burst constraints

Valid ranges for the I2S peripheral's FIFO configuration include a depth
of 16 - unconditionally setting the burst length to 16 with a fifo
threshold of size/2 will cause under/overflows.

For DMA engines with restricted capabilities the requested burst length
and FIFO thresholds need to be adjusted downward accordingly.

Both the RX and TX FIFOs operate on "less-than" thresholds. Setting the
TX threshold to fifo_size minus burst means the FIFO is kept nearly-full.

Signed-off-by: Jonathan Bell <jonathan@raspberrypi.com>

ASoC: allo-piano-dac-plus: Fix volume limit locking

Calling snd_soc_limit_volume from within a kcontrol put handler seems
to cause a deadlock as it attempts to claim a write lock that is already
held. Call snd_soc_limit_volume from the main initialisation code
instead, to avoid the recursive locking.

See: https://github.com/raspberrypi/linux/issues/6527

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: allo-piano-dac-plus: Suppress -517 errors

Use dev_err_probe to simplify the code and suppress EPROBE_DEFER errors.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

soc: pcm3168a: Add DT binding to force clock consumer mode

ASoC cannot configure the codec correctly when the ADC and DAC share clock
lines and one of them is the clock producer. Add a DT binding that
overrides ASoC and forces the component into clock consumer mode.

Signed-off-by: Stephen Gordon <gordoste@iinet.net.au>

ASoC: pcm512x: Demote "No SCLK" to debug level

Designing a PCM512X-based soundcard with no external SCLK is a valid
choice supported by the driver. Don't alarm users with messages that
say "No SCLK, using BCLK: -2" - reclassify them as debug information.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: allo-piano-dac-plus: Fix volume limiting

Controls which only exist when snd_soc_register_card returns can't be
modified before then. Move the setting of volume limits to just before
the end of the probe function.

Link: https://github.com/raspberrypi/linux/issues/6527

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: allo-piano-dac-plus: Remove pointless code

The codec control Digital Playback Volume is one of the controls deleted
by the allo-piano-dac-plus driver. It is effectively replaced by the
soundcard controls Master Playback Volume and Subwoofer Playback Volume.

Delete the code that sets the volume limit on those codec controls - the
limits on the soundcard volume controls are sufficient.

Signed-off-by: Phil Elwell <phil@raspberrypi.com>

ASoC: adds ADC8x support to the Hifiberry DAC8x

The driver probes for the ADC8x which can be stacked on top
of the DAC8x. It enables a symmetric 8 channel capture using
the dummy-dai.

Signed-off-by: j-schambacher <joerg@hifiberry.com>

sound: soc: raspberrypi: RP1 Audio Out driver as an ASOC DAI

Only 48000Hz stereo 16-bit output is currently supported.

It requires some additional OF plumbing to connect it to a
"dummy" codec and generic sound card.

Signed-off-by: Nick Hollinghurst <nick.hollinghurst@raspberrypi.com>

Adding Pimidi kernel module.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

Adding Pisound Micro kernel module

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

Pisound Micro: Fix for MIDI output under full load.

This fixes MIDI output of Pisound Micro after running for a while under
full load and increases timing stability.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>

ALSA: korg1212: replace del_timer with timer_delete

pisound-micro: Added pin_pull and pin_b_pull sysfs attributes for Pisound Micro.

These attributes are available only for GPIO input and Encoder elements.

Signed-off-by: Giedrius <giedrius@blokas.io>

Pisound Micro: Workaround for snd_soc_dai_set_tdm_slot with slots=0

Even though it's documented that specifying slots=0 can be used to disable
the TDM mode, error checking introduced in 6.12.31 version broke this,
therefore, for the time being, a workaround is to provide a xlate_tdm_slot_mask
operation implementation to return 0 instead of -EINVAL as it does in case
slots argument is 0.

Signed-off-by: Giedrius Trainavičius <giedrius@blokas.io>
2025-10-14 13:35:40 +01:00
Jack Yu
f61d7926df ASoC: rt5682s: Adjust SAR ADC button mode to fix noise issue
[ Upstream commit 1dd28fd86c ]

Adjust register settings for SAR adc button detection mode
to fix noise issue in headset.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/766cd1d2dd7a403ba65bb4cc44845f71@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-12 13:01:02 +02:00
Venkata Prasad Potturu
08fcde8b56 ASoC: amd: acp: Adjust pdm gain value
[ Upstream commit f1d0260362 ]

Set pdm gain value by setting PDM_MISC_CTRL_MASK value.
To avoid low pdm gain value.

Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20250821054606.1279178-1-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-12 13:01:01 +02:00
Shuming Fan
4840047864 ASoC: rt712: avoid skipping the blind write
[ Upstream commit f54d87dad7 ]

Some devices might not use the DMIC function of the RT712VB.
Therefore, this patch avoids skipping the blind write with RT712VB.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20250901085757.1287945-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-12 13:01:01 +02:00
Shenghao Ding
3d46a1e39c ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
commit 71d2893a23 upstream.

A bug reported by one of my customers that the order of TAS2781
calibrated-data is incorrect, the correct way is to move R0_Low
and insert it between R0 and InvR0.

Fixes: 4fe2385134 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20250907222728.988-1-shenghao-ding@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-10-12 13:00:59 +02:00
Srinivas Kandagatla
8f9c9fafc0 ASoC: qcom: audioreach: fix potential null pointer dereference
commit 8318e04ab2 upstream.

It is possible that the topology parsing function
audioreach_widget_load_module_common() could return NULL or an error
pointer. Add missing NULL check so that we do not dereference it.

Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Cc: Stable@vger.kernel.org
Fixes: 36ad9bf1d9 ("ASoC: qdsp6: audioreach: add topology support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20250825101247.152619-2-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-10-06 11:18:48 +02:00
Jeongjun Park
af600e7f55 ALSA: usb-audio: fix race condition to UAF in snd_usbmidi_free
commit 9f2c0ac142 upstream.

The previous commit 0718a78f6a ("ALSA: usb-audio: Kill timer properly at
removal") patched a UAF issue caused by the error timer.

However, because the error timer kill added in this patch occurs after the
endpoint delete, a race condition to UAF still occurs, albeit rarely.

Additionally, since kill-cleanup for urb is also missing, freed memory can
be accessed in interrupt context related to urb, which can cause UAF.

Therefore, to prevent this, error timer and urb must be killed before
freeing the heap memory.

Cc: <stable@vger.kernel.org>
Reported-by: syzbot+f02665daa2abeef4a947@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=f02665daa2abeef4a947
Fixes: 0718a78f6a ("ALSA: usb-audio: Kill timer properly at removal")
Signed-off-by: Jeongjun Park <aha310510@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-10-06 11:18:47 +02:00
qaqland
4ceb739a32 ALSA: usb-audio: Add mute TLV for playback volumes on more devices
[ Upstream commit 2cbe4ac193 ]

Applying the quirk of that, the lowest Playback mixer volume setting
mutes the audio output, on more devices.

Suggested-by: Cryolitia PukNgae <cryolitia@uniontech.com>
Signed-off-by: qaqland <anguoli@uniontech.com>
Link: https://patch.msgid.link/20250829-sound_quirk-v1-1-745529b44440@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:26 +02:00
Cryolitia PukNgae
f20938fb3b ALSA: usb-audio: move mixer_quirks' min_mute into common quirk
[ Upstream commit 2c3ca8cc55 ]

We have found more and more devices that have the same problem, that
the mixer's minimum value is muted. Accroding to pipewire's MR[1]
and Arch Linux wiki[2], this should be a very common problem in USB
audio devices. Move the quirk into common quirk,as a preparation of
more devices' quirk's patch coming on the road[3].

1. https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/2514
2. https://wiki.archlinux.org/index.php?title=PipeWire&oldid=804138#No_sound_from_USB_DAC_until_30%_volume
3. On the road, in the physical sense. We have been buying ton of
   these devices for testing the problem.

Tested-by: Guoli An <anguoli@uniontech.com>
Signed-off-by: Cryolitia PukNgae <cryolitia@uniontech.com>
Link: https://patch.msgid.link/20250827-sound-quirk-min-mute-v1-1-4717aa8a4f6a@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:26 +02:00
noble.yang
001470af94 ALSA: usb-audio: Add DSD support for Comtrue USB Audio device
[ Upstream commit e9df175548 ]

The vendor Comtrue Inc. (0x2fc6) produces USB audio chipsets like
the CT7601 which are capable of Native DSD playback.

This patch adds QUIRK_FLAG_DSD_RAW for Comtrue (VID 0x2fc6), which enables
native DSD playback (DSD_U32_LE) on their USB Audio device. This has been
verified under Ubuntu 25.04 with JRiver.

Signed-off-by: noble.yang <noble.yang@comtrue-inc.com>
Link: https://patch.msgid.link/20250731110614.4070-1-noble228@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:26 +02:00
Balamurugan C
1e1873264e ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL.
[ Upstream commit 03aa2ed9e1 ]

Added match table entry on ptl machines to support HDMI-In capture
with rt5682 I2S audio codec. also added the respective quirk
configuration in rt5682 machine driver.

Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250716082300.1810352-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:25 +02:00
Balamurugan C
eae9d5c299 ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in PTL match table
[ Upstream commit fb00ab1f39 ]

Adding HDMI-In capture via I2S feature support in PTL platform.

Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250708080030.1257790-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 03aa2ed9e1 ("ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL.")
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:25 +02:00
Balamurugan C
71f64a3244 ASoC: Intel: soc-acpi: Add entry for sof_es8336 in PTL match table.
[ Upstream commit 2813f535b5 ]

Adding ES83x6 I2S codec support for PTL platforms and entry in match table.

Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250708080030.1257790-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 03aa2ed9e1 ("ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL.")
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:25 +02:00
Takashi Iwai
330e7cc51c ALSA: usb-audio: Fix build with CONFIG_INPUT=n
[ Upstream commit d0630a0b80 ]

The recent addition of DualSense mixer quirk relies on the input
device handle, and the build can fail if CONFIG_INPUT isn't set.
Put (rather ugly) workarounds to wrap with IS_REACHABLE() for avoiding
the build error.

Fixes: 79d561c4ec ("ALSA: usb-audio: Add mixer quirk for Sony DualSense PS5")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202506130733.gnPKw2l3-lkp@intel.com/
Reviewed-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20250613081543.7404-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:24 +02:00
Stefan Binding
645c7aa98d ALSA: hda/realtek: Add support for ASUS NUC using CS35L41 HDA
[ Upstream commit 84fc8896f0 ]

Add support for ASUS NUC14LNS.

This NUC uses a single CS35L41 Amp in using Internal Boost with SPI.
To support the Single Amp, a new quirk is required.

Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250612160029.848104-3-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:24 +02:00
Chen Ni
9a183aeb23 ALSA: usb-audio: Convert comma to semicolon
[ Upstream commit 9ca30a1b00 ]

Replace comma between expressions with semicolons.

Using a ',' in place of a ';' can have unintended side effects.
Although that is not the case here, it is seems best to use ';'
unless ',' is intended.

Found by inspection.
No functional change intended.
Compile tested only.

Fixes: 79d561c4ec ("ALSA: usb-audio: Add mixer quirk for Sony DualSense PS5")
Signed-off-by: Chen Ni <nichen@iscas.ac.cn>
Reviewed-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20250612060228.1518028-1-nichen@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:24 +02:00
Cristian Ciocaltea
0105cfc41a ALSA: usb-audio: Add mixer quirk for Sony DualSense PS5
[ Upstream commit 79d561c4ec ]

The Sony DualSense wireless controller (PS5) features an internal mono
speaker, but it also provides a 3.5mm jack socket for headphone output
and headset microphone input.

Since this is a UAC1 device, it doesn't advertise any jack detection
capability.  However, the controller is able to report HP & MIC insert
events via HID, i.e. through a dedicated input device managed by the
hid-playstation driver.

Add a quirk to create the jack controls for headphone and headset mic,
respectively, and setup an input handler for each of them in order to
intercept the related hotplug events.

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-9-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
042ce4cb97 ALSA: usb-audio: Remove unneeded wmb() in mixer_quirks
[ Upstream commit 9cea742559 ]

Adding a memory barrier before wake_up() in
snd_usb_soundblaster_remote_complete() is supposed to ensure the write
to mixer->rc_code is visible in wait_event_interruptible() from
snd_usb_sbrc_hwdep_read().

However, this is not really necessary, since wake_up() is just a wrapper
over __wake_up() which already executes a full memory barrier before
accessing the state of the task to be waken up.

Drop the redundant call to wmb() and implicitly fix the checkpatch
complaint:

  WARNING: memory barrier without comment

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-8-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
9f76d2c9e8 ALSA: usb-audio: Simplify NULL comparison in mixer_quirks
[ Upstream commit f2d6d660e8 ]

Handle report from checkpatch.pl:

  CHECK: Comparison to NULL could be written "t->name"

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-7-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
8af6015e38 ALSA: usb-audio: Avoid multiple assignments in mixer_quirks
[ Upstream commit 03ddd3bdb9 ]

Handle report from checkpatch.pl:

  CHECK: multiple assignments should be avoided

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-6-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
d3934ea7fb ALSA: usb-audio: Drop unnecessary parentheses in mixer_quirks
[ Upstream commit c0495cef8b ]

Fix multiple 'CHECK: Unnecessary parentheses around ...' reports from
checkpatch.pl.

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-5-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
0afc2246dd ALSA: usb-audio: Fix block comments in mixer_quirks
[ Upstream commit 231225d8a2 ]

Address a couple of comment formatting issues indicated by
checkpatch.pl:

  WARNING: Block comments use a trailing */ on a separate line

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-4-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
c11341fb8f ALSA: usb-audio: Fix whitespace & blank line issues in mixer_quirks
[ Upstream commit df6b4dcf2e ]

Address all whitespace & blank line(s) related issues reported by
checkpatch.pl:

  ERROR: trailing whitespace
  ERROR: space required after that ',' (ctx:VxV)
  WARNING: Missing a blank line after declarations
  CHECK: Please use a blank line after function/struct/union/enum declarations
  CHECK: Please don't use multiple blank lines

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-2-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Cristian Ciocaltea
2ea8b2ce48 ALSA: usb-audio: Fix code alignment in mixer_quirks
[ Upstream commit bca638aa73 ]

Format code to fix all alignment issues reported by checkpatch.pl:

  CHECK: Alignment should match open parenthesis

Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-1-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-10-02 13:48:23 +02:00
Takashi Iwai
09b473a80c ALSA: usb: qcom: Fix false-positive address space check
[ Upstream commit 44499ecb4f ]

The sanity check previously added to uaudio_transfer_buffer_setup()
assumed the allocated buffer being linear-mapped.  But the buffer
allocated via usb_alloc_coherent() isn't always so, rather to be used
with (SG-)DMA API.  This leaded to a false-positive warning and the
driver failed to work.

Actually uaudio_transfer_buffer_setup() deals only with the DMA-API
addresses for MEM_XFER_BUF type, while other callers of
uaudio_iommu_map() are with pages with physical addresses for
MEM_EVENT_RING and MEM_XFER_RING types.  So this patch splits the
mapping helper function to two different ones, uaudio_iommu_map() for
the DMA pages and uaudio_iommu_map_pa() for the latter, in order to
handle mapping differently for each type.  Along with it, the
unnecessary address check that caused probe error is dropped, too.

Fixes: 3335a1bbd6 ("ALSA: qc_audio_offload: try to reduce address space confusion")
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Reported-and-tested-by: Luca Weiss <luca.weiss@fairphone.com>
Closes: https://lore.kernel.org/DBR2363A95M1.L9XBNC003490@fairphone.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:52 +02:00
Venkata Prasad Potturu
65c5cfbd6d ASoC: amd: acp: Fix incorrect retrival of acp_chip_info
[ Upstream commit d7871f400c ]

Use dev_get_drvdata(dev->parent) instead of dev_get_platdata(dev)
to correctly obtain acp_chip_info members in the acp I2S driver.
Previously, some members were not updated properly due to incorrect
data access, which could potentially lead to null pointer
dereferences.

This issue was missed in the earlier commit
("ASoC: amd: acp: Fix NULL pointer deref in acp_i2s_set_tdm_slot"),
which only addressed set_tdm_slot(). This change ensures that all
relevant functions correctly retrieve acp_chip_info, preventing
further null pointer dereference issues.

Fixes: e3933683b2 ("ASoC: amd: acp: Remove redundant acp_dev_data structure")

Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250910171419.3682468-1-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Amadeusz Sławiński
34f3a9e04e ASoC: Intel: catpt: Expose correct bit depth to userspace
[ Upstream commit 690aa09b18 ]

Currently wrong bit depth is exposed in hw params, causing clipped
volume during playback. Expose correct parameters.

Fixes: a126750fc8 ("ASoC: Intel: catpt: PCM operations")
Reported-by: Andy Shevchenko <andriy.shevchenko@intel.com>
Tested-by: Andy Shevchenko <andriy.shevchenko@intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Message-ID: <20250909092829.375953-1-amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Charles Keepax
f643373332 ASoC: SDCA: Fix return value in sdca_regmap_mbq_size()
[ Upstream commit f81e630476 ]

The MBQ size function returns an integer representing the size of a
Control. Currently if the Control is not found the function will return
false which makes little sense. Correct this typo to return -EINVAL.

Fixes: e3f7caf74b ("ASoC: SDCA: Add generic regmap SDCA helpers")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Message-ID: <20250820163717.1095846-2-ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Colin Ian King
9ff967d7e8 ASoC: SOF: Intel: hda-stream: Fix incorrect variable used in error message
[ Upstream commit 35fc531a59 ]

The dev_err message is reporting an error about capture streams however
it is using the incorrect variable num_playback instead of num_capture.
Fix this by using the correct variable num_capture.

Fixes: a1d1e266b4 ("ASoC: SOF: Intel: Add Intel specific HDA stream operations")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250902120639.2626861-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Dan Carpenter
cd59ca8f75 ASoC: codec: sma1307: Fix memory corruption in sma1307_setting_loaded()
[ Upstream commit 78338108b5 ]

The sma1307->set.header_size is how many integers are in the header
(there are 8 of them) but instead of allocating space of 8 integers
we allocate 8 bytes.  This leads to memory corruption when we copy data
it on the next line:

        memcpy(sma1307->set.header, data,
               sma1307->set.header_size * sizeof(int));

Also since we're immediately copying over the memory in ->set.header,
there is no need to zero it in the allocator.  Use devm_kmalloc_array()
to allocate the memory instead.

Fixes: 576c57e6b4 ("ASoC: sma1307: Add driver for Iron Device SMA1307")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://patch.msgid.link/aLGjvjpueVstekXP@stanley.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Charles Keepax
7c28b31b22 ASoC: wm8974: Correct PLL rate rounding
[ Upstream commit 9b17d3724d ]

Using a single value of 22500000 for both 48000Hz and 44100Hz audio
will sometimes result in returning wrong dividers due to rounding.
Update the code to use the actual value for both.

Fixes: 51b2bb3f25 ("ASoC: wm8974: configure pll and mclk divider automatically")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250821082639.1301453-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Charles Keepax
badf614319 ASoC: wm8940: Correct typo in control name
[ Upstream commit b4799520dc ]

Fixes: 0b5e92c5e0 ("ASoC WM8940 Driver")
Reported-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Link: https://patch.msgid.link/20250821082639.1301453-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:51 +02:00
Charles Keepax
7a372ac1e8 ASoC: wm8940: Correct PLL rate rounding
[ Upstream commit d05afb53c6 ]

Using a single value of 22500000 for both 48000Hz and 44100Hz audio
will sometimes result in returning wrong dividers due to rounding.
Update the code to use the actual value for both.

Fixes: 294833fc9e ("ASoC: wm8940: Rewrite code to set proper clocks")
Reported-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Link: https://patch.msgid.link/20250821082639.1301453-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:50 +02:00
Praful Adiga
519b95c74f ALSA: hda/realtek: Fix mute led for HP Laptop 15-dw4xx
commit d33c347104 upstream.

This laptop uses the ALC236 codec with COEF 0x7 and idx 1 to
control the mute LED. Enable the existing quirk for this device.

Signed-off-by: Praful Adiga <praful.adiga@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-25 11:16:50 +02:00
Mohammad Rafi Shaik
66e6d1c928 ASoC: qcom: q6apm-lpass-dais: Fix missing set_fmt DAI op for I2S
commit 33b55b94bc upstream.

The q6i2s_set_fmt() function was defined but never linked into the
I2S DAI operations, resulting DAI format settings is being ignored
during stream setup. This change fixes the issue by properly linking
the .set_fmt handler within the DAI ops.

Fixes: 30ad723b93 ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Cc: stable@vger.kernel.org
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Message-ID: <20250908053631.70978-3-mohammad.rafi.shaik@oss.qualcomm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-25 11:16:48 +02:00
Krzysztof Kozlowski
cc336b242e ASoC: qcom: q6apm-lpass-dais: Fix NULL pointer dereference if source graph failed
commit 68f27f7c77 upstream.

If earlier opening of source graph fails (e.g. ADSP rejects due to
incorrect audioreach topology), the graph is closed and
"dai_data->graph[dai->id]" is assigned NULL.  Preparing the DAI for sink
graph continues though and next call to q6apm_lpass_dai_prepare()
receives dai_data->graph[dai->id]=NULL leading to NULL pointer
exception:

  qcom-apm gprsvc:service:2:1: Error (1) Processing 0x01001002 cmd
  qcom-apm gprsvc:service:2:1: DSP returned error[1001002] 1
  q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: fail to start APM port 78
  q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: ASoC: error at snd_soc_pcm_dai_prepare on TX_CODEC_DMA_TX_3: -22
  Unable to handle kernel NULL pointer dereference at virtual address 00000000000000a8
  ...
  Call trace:
   q6apm_graph_media_format_pcm+0x48/0x120 (P)
   q6apm_lpass_dai_prepare+0x110/0x1b4
   snd_soc_pcm_dai_prepare+0x74/0x108
   __soc_pcm_prepare+0x44/0x160
   dpcm_be_dai_prepare+0x124/0x1c0

Fixes: 30ad723b93 ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Cc: stable@vger.kernel.org
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Message-ID: <20250904101849.121503-2-krzysztof.kozlowski@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-25 11:16:48 +02:00
Mohammad Rafi Shaik
59c4accddf ASoC: qcom: audioreach: Fix lpaif_type configuration for the I2S interface
commit 5f1af203ef upstream.

Fix missing lpaif_type configuration for the I2S interface.
The proper lpaif interface type required to allow DSP to vote
appropriate clock setting for I2S interface.

Fixes: 25ab80db6b ("ASoC: qdsp6: audioreach: add module configuration command helpers")
Cc: stable@vger.kernel.org
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Message-ID: <20250908053631.70978-2-mohammad.rafi.shaik@oss.qualcomm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-25 11:16:48 +02:00
Maciej Strozek
8276c97dcc ASoC: SDCA: Add quirk for incorrect function types for 3 systems
commit 28edfaa10c upstream.

Certain systems have CS42L43 DisCo that claims to conform to version 0.6.28
but uses the function types from the 1.0 spec. Add a quirk as a workaround.

Closes: https://github.com/thesofproject/linux/issues/5515
Cc: stable@vger.kernel.org
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20250901151518.3197941-1-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-25 11:16:48 +02:00
Takashi Sakamoto
778a062c13 ALSA: firewire-motu: drop EPOLLOUT from poll return values as write is not supported
[ Upstream commit aea3493246 ]

The ALSA HwDep character device of the firewire-motu driver incorrectly
returns EPOLLOUT in poll(2), even though the driver implements no operation
for write(2). This misleads userspace applications to believe write() is
allowed, potentially resulting in unnecessarily wakeups.

This issue dates back to the driver's initial code added by a commit
71c3797779 ("ALSA: firewire-motu: add hwdep interface"), and persisted
when POLLOUT was updated to EPOLLOUT by a commit a9a08845e9 ('vfs: do
bulk POLL* -> EPOLL* replacement("").').

This commit fixes the bug.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://patch.msgid.link/20250829233749.366222-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-25 11:16:42 +02:00
Gergo Koteles
bafcd9a090 ALSA: hda: tas2781: reorder tas2563 calibration variables
commit d5f8458e34 upstream.

The tasdev_load_calibrated_data() function expects the calibration data
values in the cali_data buffer as R0, R0Low, InvR0, Power, TLim which
is not the same as what tas2563_save_calibration() writes to the buffer.

Reorder the EFI variables in the tas2563_save_calibration() function
to put the values in the buffer in the correct order.

Fixes: 4fe2385134 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib")
Cc: <stable@vger.kernel.org>
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://patch.msgid.link/20250829160450.66623-2-soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-09 19:02:36 +02:00
Gergo Koteles
5601628904 ALSA: hda: tas2781: fix tas2563 EFI data endianness
commit e5a00dafc7 upstream.

Before conversion to unify the calibration data management, the
tas2563_apply_calib() function performed the big endian conversion and
wrote the calibration data to the device. The writing is now done by the
common tasdev_load_calibrated_data() function, but without conversion.

Put the values into the calibration data buffer with the expected
endianness.

Fixes: 4fe2385134 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib")
Cc: <stable@vger.kernel.org>
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://patch.msgid.link/20250829160450.66623-1-soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-09 19:02:35 +02:00
Aaron Erhardt
1dfa6e6660 ALSA: hda/realtek: Fix headset mic for TongFang X6[AF]R5xxY
commit 051b02b17a upstream.

Add a PCI quirk to enable microphone detection on the headphone jack of
TongFang X6AR5xxY and X6FR5xxY devices.

Signed-off-by: Aaron Erhardt <aer@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250826141054.1201482-1-aer@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-09 19:02:35 +02:00
Takashi Iwai
cfe842782f ALSA: hda/hdmi: Add pin fix for another HP EliteDesk 800 G4 model
commit bcd6659d49 upstream.

It was reported that HP EliteDesk 800 G4 DM 65W (SSID 103c:845a) needs
the similar quirk for enabling HDMI outputs, too.  This patch adds the
corresponding quirk entry.

Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250901115009.27498-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-09 19:02:35 +02:00
Cryolitia PukNgae
0bb3678a24 ALSA: usb-audio: Add mute TLV for playback volumes on some devices
commit 9c6182843b upstream.

Applying the quirk of that, the lowest Playback mixer volume setting
mutes the audio output, on more devices.

Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/2514
Cc: <stable@vger.kernel.org>
Tested-by: Guoli An <anguoli@uniontech.com>
Signed-off-by: Cryolitia PukNgae <cryolitia@uniontech.com>
Link: https://patch.msgid.link/20250822-mixer-quirk-v1-1-b19252239c1c@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2025-09-09 19:02:28 +02:00
Tina Wuest
76ff6437b4 ALSA: usb-audio: Allow Focusrite devices to use low samplerates
[ Upstream commit cc8e91054c ]

Commit 05f254a636 ("ALSA: usb-audio:
Improve filtering of sample rates on Focusrite devices") changed the
check for max_rate in a way which was overly restrictive, forcing
devices to use very high samplerates if they support them, despite
support existing for lower rates as well.

This maintains the intended outcome (ensuring samplerates selected are
supported) while allowing devices with higher maximum samplerates to be
opened at all supported samplerates.

This patch was tested with a Clarett+ 8Pre USB

Fixes: 05f254a636 ("ALSA: usb-audio: Improve filtering of sample rates on Focusrite devices")
Signed-off-by: Tina Wuest <tina@wuest.me>
Link: https://patch.msgid.link/20250901092024.140993-1-tina@wuest.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-09 19:02:15 +02:00
Ajye Huang
1b1a33795a ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
[ Upstream commit 3e7fd1febc ]

Add the missing op in the device description to avoid issues with jack
detection.

Fixes: 6b04629ae9 ("ASoC: SOF: Intel: add initial support for WCL")
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Ajye Huang <ajye_huang@compal.corp-partner.google.com>
Message-ID: <20250826154040.2723998-1-ajye_huang@compal.corp-partner.google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-09 19:02:15 +02:00
Kuninori Morimoto
391203f768 ASoC: rsnd: tidyup direction name on rsnd_dai_connect()
[ Upstream commit 8022629548 ]

commit 2c6b6a3e8b ("ASoC: rsnd: use snd_pcm_direction_name()") uses
snd_pcm_direction_name() instead of original method to get string
"Playback" or "Capture". But io->substream might be NULL in this timing.
Let's re-use original method.

Fixes: 2c6b6a3e8b ("ASoC: rsnd: use snd_pcm_direction_name()")
Reported-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Tested-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-ID: <87zfbmwq6v.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-09 19:02:14 +02:00
Kuninori Morimoto
1d282dcd46 ASoC: soc-core: care NULL dirver name on snd_soc_lookup_component_nolocked()
[ Upstream commit 168873ca17 ]

soc-generic-dmaengine-pcm.c uses same dev for both CPU and Platform.
In such case, CPU component driver might not have driver->name, then
snd_soc_lookup_component_nolocked() will be NULL pointer access error.
Care NULL driver name.

	Call trace:
	 strcmp from snd_soc_lookup_component_nolocked+0x64/0xa4
	 snd_soc_lookup_component_nolocked from snd_soc_unregister_component_by_driver+0x2c/0x44
	 snd_soc_unregister_component_by_driver from snd_dmaengine_pcm_unregister+0x28/0x64
	 snd_dmaengine_pcm_unregister from devres_release_all+0x98/0xfc
	 devres_release_all from device_unbind_cleanup+0xc/0x60
	 device_unbind_cleanup from really_probe+0x220/0x2c8
	 really_probe from __driver_probe_device+0x88/0x1a0
	 __driver_probe_device from driver_probe_device+0x30/0x110
	driver_probe_device from __driver_attach+0x90/0x178
	__driver_attach from bus_for_each_dev+0x7c/0xcc
	bus_for_each_dev from bus_add_driver+0xcc/0x1ec
	bus_add_driver from driver_register+0x80/0x11c
	driver_register from do_one_initcall+0x58/0x23c
	do_one_initcall from kernel_init_freeable+0x198/0x1f4
	kernel_init_freeable from kernel_init+0x1c/0x12c
	kernel_init from ret_from_fork+0x14/0x28

Fixes: 144d6dfc74 ("ASoC: soc-core: merge snd_soc_unregister_component() and snd_soc_unregister_component_by_driver()")
Reported-by: J. Neuschäfer <j.ne@posteo.net>
Closes: https://lore.kernel.org/r/aJb311bMDc9x-dpW@probook
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reported-by: Ondřej Jirman <megi@xff.cz>
Closes: https://lore.kernel.org/r/arxpwzu6nzgjxvsndct65ww2wz4aezb5gjdzlgr24gfx7xvyih@natjg6dg2pj6
Tested-by: J. Neuschäfer <j.ne@posteo.net>
Message-ID: <87ect8ysv8.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-09 19:02:14 +02:00
Shuming Fan
1e46ce777f ASoC: rt1320: fix random cycle mute issue
[ Upstream commit f48d7a1b0b ]

This patch fixed the random cycle mute issue that occurs during long-time playback.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20250807092432.997989-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-04 16:55:31 +02:00
Shuming Fan
7a33a93b3d ASoC: rt721: fix FU33 Boost Volume control not working
[ Upstream commit 633e391d45 ]

This patch fixed FU33 Boost Volume control not working.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20250808055706.1110766-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
2025-09-04 16:55:31 +02:00