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>
[ Upstream commit d26e9f669c ]
Since 8b3a087f7f ("ALSA: usb-audio: Unify virtual type units type to
UAC3 values") usb-audio is using UAC3_CLOCK_SOURCE instead of
bDescriptorSubtype, later refactored with e0ccdef926 ("ALSA: usb-audio:
Clean up check_input_term()") into parse_term_uac2_clock_source().
This breaks the clock source selection for at least my
1397:0003 BEHRINGER International GmbH FCA610 Pro.
Fix by using UAC2_CLOCK_SOURCE in parse_term_uac2_clock_source().
Fixes: 8b3a087f7f ("ALSA: usb-audio: Unify virtual type units type to UAC3 values")
Signed-off-by: René Rebe <rene@exactco.de>
Link: https://patch.msgid.link/20251125.154149.1121389544970412061.rene@exactco.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 249d96b492 ]
Since snd_soc_suspend() is invoked through snd_soc_pm_ops->suspend(),
and snd_soc_pm_ops is associated with the soc_driver (defined in
sound/soc/soc-core.c), and there is no parent-child relationship between
the soc_driver and the DA7213 codec driver, the power management subsystem
does not enforce a specific suspend/resume order between the DA7213 driver
and the soc_driver.
Because of this, the different codec component functionalities, called from
snd_soc_resume() to reconfigure various functions, can race with the
DA7213 struct dev_pm_ops::resume function, leading to misapplied
configuration. This occasionally results in clipped sound.
Fix this by dropping the struct dev_pm_ops::{suspend, resume} and use
instead struct snd_soc_component_driver::{suspend, resume}. This ensures
the proper configuration sequence is handled by the ASoC subsystem.
Cc: stable@vger.kernel.org
Fixes: 431e040065 ("ASoC: da7213: Add suspend to RAM support")
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20251104114914.2060603-1-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1a58d865f4 upstream.
The bus_find_device_by_name() function returns a device pointer with an
incremented reference count, but the original code was missing put_device()
calls in some return paths, leading to reference count leaks.
Fix this by ensuring put_device() is called before function exit after
bus_find_device_by_name() succeeds
This follows the same pattern used elsewhere in the kernel where
bus_find_device_by_name() is properly paired with put_device().
Found via static analysis and code review.
Fixes: 4f8ef33dd4 ("ASoC: soc_sdw_utils: skip the endpoint that doesn't present")
Cc: stable@vger.kernel.org
Signed-off-by: Miaoqian Lin <linmq006@gmail.com>
Link: https://patch.msgid.link/20251029071804.8425-1-linmq006@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 05a1fc5efd upstream.
The PCM stream data in USB-audio driver is transferred over USB URB
packet buffers, and each packet size is determined dynamically. The
packet sizes are limited by some factors such as wMaxPacketSize USB
descriptor. OTOH, in the current code, the actually used packet sizes
are determined only by the rate and the PPS, which may be bigger than
the size limit above. This results in a buffer overflow, as reported
by syzbot.
Basically when the limit is smaller than the calculated packet size,
it implies that something is wrong, most likely a weird USB
descriptor. So the best option would be just to return an error at
the parameter setup time before doing any further operations.
This patch introduces such a sanity check, and returns -EINVAL when
the packet size is greater than maxpacksize. The comparison with
ep->packsize[1] alone should suffice since it's always equal or
greater than ep->packsize[0].
Reported-by: syzbot+bfd77469c8966de076f7@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=bfd77469c8966de076f7
Link: https://lore.kernel.org/690b6b46.050a0220.3d0d33.0054.GAE@google.com
Cc: Lizhi Xu <lizhi.xu@windriver.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20251109091211.12739-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 82420bd4e1 upstream.
After restructuring and splitting the HDMI codec driver code, each
HDMI codec driver contains the own build_controls and build_pcms ops.
A copy-n-paste error put the wrong entries for nvhdmi-mcp driver; both
build_controls and build_pcms are swapped. Unfortunately both
callbacks have the very same form, and the compiler didn't complain
it, either. This resulted in a NULL dereference because the PCM
instance hasn't been initialized at calling the build_controls
callback.
Fix it by passing the proper entries.
Fixes: ad781b550f ("ALSA: hda/hdmi: Rewrite to new probe method")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220743
Link: https://patch.msgid.link/20251106104647.25805-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 632108ec07 ]
In snd_usb_create_streams(), for UAC version 3 devices, the Interface
Association Descriptor (IAD) is retrieved via usb_ifnum_to_if(). If this
call fails, a fallback routine attempts to obtain the IAD from the next
interface and sets a BADD profile. However, snd_usb_mixer_controls_badd()
assumes that the IAD retrieved from usb_ifnum_to_if() is always valid,
without performing a NULL check. This can lead to a NULL pointer
dereference when usb_ifnum_to_if() fails to find the interface descriptor.
This patch adds a NULL pointer check after calling usb_ifnum_to_if() in
snd_usb_mixer_controls_badd() to prevent the dereference.
This issue was discovered by syzkaller, which triggered the bug by sending
a crafted USB device descriptor.
Fixes: 17156f23e9 ("ALSA: usb: add UAC3 BADD profiles support")
Signed-off-by: Haein Lee <lhi0729@kaist.ac.kr>
Link: https://patch.msgid.link/vwhzmoba9j2f.vwhzmob9u9e2.g6@dooray.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6b6eddc63c ]
The probe function enables regulators at the beginning
but fails to disable them in its error handling path.
If any operation after enabling the regulators fails,
the probe will exit with an error, leaving the regulators
permanently enabled, which could lead to a resource leak.
Add a proper error handling path to call regulator_bulk_disable()
before returning an error.
Fixes: 9a397f4736 ("ASoC: cs4271: add regulator consumer support")
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20251105062246.1955-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ee70bacef1 ]
The interrupt handler offloads the microphone detection logic to
nau8821_jdet_work(), which implies a sleep operation. However, before
being able to process any subsequent hotplug event, the interrupt
handler needs to wait for any prior scheduled work to complete.
Move the sleep out of jdet_work by converting it to a delayed work.
This eliminates the undesired blocking in the interrupt handler when
attempting to cancel a recently scheduled work item and should help
reducing transient input reports that might confuse user-space.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20251003-nau8821-jdet-fixes-v1-5-f7b0e2543f09@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 7ddb711b6e upstream.
Optimize the time consumption of profile switching, init_profile saves
the common settings of different profiles, such as the dsp coefficients,
etc, which can greatly reduce the profile switching time comsumption and
remove the repetitive settings.
Fixes: e83dcd139e ("ASoC: tas2781: Add keyword "init" in profile section")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 328b80b29a ]
The ASUS ROG Zephyrus Duo 15 SE (GX551QS) with ALC 289 codec requires specific
pin configuration for proper volume control. Without this quirk, volume
adjustments produce a muffled sound effect as only certain channels attenuate,
leaving bass frequency at full volume.
Testing with hdajackretask confirms these pin tweaks fix the issue:
- Pin 0x17: Internal Speaker (LFE)
- Pin 0x1e: Internal Speaker
Signed-off-by: Adam Holliday <dochollidayxx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 733a763dd8 ]
The problem of having class-D initialization sequence in probe using
regmap_register_patch() is that it will do hardware register writes
immediately after being called as it bypasses regcache. Afterwards, in
aic3x_init() we also perform codec soft reset, rendering class-D init
sequence pointless. This issue is even more apparent when using reset
GPIO line, since in that case class-D amplifier initialization fails
with "Failed to init class D: -5" message as codec is already held in
reset state after requesting the reset GPIO and hence hardware I/O
fails with -EIO errno.
Thus move class-D amplifier initialization sequence from probe function
to aic3x_set_power() just before the usual regcache sync. Use bypassed
regmap_multi_reg_write_bypassed() function to make sure, class-D init
sequence is performed in proper order as described in the datasheet.
Signed-off-by: Primoz Fiser <primoz.fiser@norik.com>
Link: https://patch.msgid.link/20250925085929.2581749-1-primoz.fiser@norik.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 27fa1a8b28 ]
The mclk direction now needs to be specified in endpoint node with
"system-clock-direction-out" property. However some calls to the
set_sysclk callback, related to CPU DAI clock, result in unbalanced
calls to clock API.
The set_sysclk callback in STM32 SAI driver is intended only for mclk
management. So it is relevant to ensure that calls to set_sysclk are
related to mclk only.
Since the master clock is handled only at runtime, skip the calls to
set_sysclk in the initialization phase.
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20250916123118.84175-1-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 25226abc1a ]
MSIOF has TXRST/RXRST to reset FIFO, but it shouldn't be used during SYNC
signal was asserted, because it will be cause of HW issue.
When MSIOF is used as Sound driver, this driver is assuming it is used as
clock consumer mode (= Codec is clock provider). This means, it can't
control SYNC signal by itself.
We need to use SW reset (= reset_control_xxx()) instead of TXRST/RXRST.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Link: https://patch.msgid.link/87cy7fyuug.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d1d6ad7f66 ]
Testing with a Presonus STUDIO 1824c together with
a Behringer ultragain digital ADAT device shows that
using all 3 altno settings works fine.
When selecting sample rate, the driver sets the interface
to the correct altno setting and the correct number of
channels is set.
Selecting the correct altno setting via Ardour, Reaper or
whatever other way to set the sample rate is more convenient
than re-loading the driver module with device_setup to
set altno.
Signed-off-by: Roy Vegard Ovesen <roy.vegard.ovesen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8497324901 ]
If multiple instances of this driver are instantiated and try to send
concurrently then the single static buffer snd_serial_generic_tx_work()
will cause corruption in the data output.
Move the buffer into the per-instance driver data to avoid this.
Signed-off-by: John Keeping <jkeeping@inmusicbrands.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2aec0b6a6b ]
Just add fixed struct size validations for UAC2 and UAC3 effect
units. The descriptor has a variable-length array, so it should be
validated with a proper function later once when the unit is really
parsed and used by the driver (currently only referred partially for
the input terminal parsing).
Link: https://patch.msgid.link/20250821151751.12100-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b2dd1d0d32 ]
When configured for default synchronisation (Rx syncs to Tx) and the
SAI operates in consumer mode (clocks provided externally to Tx), a
synchronisation error occurs on Tx on the first attempt after device
initialisation when the playback stream is started while a capture
stream is already active. This results in channel shift/swap on the
playback stream.
Subsequent streams (ie after that first failing one) always work
correctly, no matter the order, with or without the other stream active.
This issue was observed (and fix tested) on an i.MX6UL board connected
to an ADAU1761 codec, where the codec provides both frame and bit clock
(connected to TX pins).
To fix this, always initialize the 'other' xCR4 and xCR5 registers when
we're starting a stream which is synced to the opposite one, irregardless
of the producer/consumer status.
Fixes: 51659ca069 ("ASoC: fsl-sai: set xCR4/xCR5/xMR for SAI master mode")
Signed-off-by: Maarten Zanders <maarten@zanders.be>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20251024135716.584265-1-maarten@zanders.be
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 75cdae446d ]
The log messages for the PreSonus STUDIO 1810c about
device_setup are not applicable to the 1824c, and should
not be logged when 1824c initializes.
Refactor from if statement to switch statement as there
might be more STUDIO series devices added later.
Fixes: 080564558e ("ALSA: usb-audio: enable support for Presonus Studio 1824c within 1810c file")
Signed-off-by: Roy Vegard Ovesen <roy.vegard.ovesen@gmail.com>
Link: https://patch.msgid.link/aPaYTP7ceuABf8c7@ark
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 659169c4eb ]
The 1824c does not have the A/B switch that the 1810c has,
but instead it has a mono main switch that sums the two
main output channels to mono.
Signed-off-by: Roy Vegard Ovesen <roy.vegard.ovesen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 75cdae446d ("ALSA: usb-audio: don't log messages meant for 1810c when initializing 1824c")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 79a6f2da16 ]
Both mt8195-afe-pcm and mt8365-afe-pcm drivers use devm_pm_runtime_enable()
in probe function, which automatically calls pm_runtime_disable() on device
removal via devres mechanism. However, the remove callbacks explicitly call
pm_runtime_disable() again, resulting in double pm_runtime_disable() calls.
Fix by removing the redundant pm_runtime_disable() calls from remove
functions, letting the devres framework handle it automatically.
Fixes: 2ca0ec01d4 ("ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe")
Fixes: e1991d102b ("ASoC: mediatek: mt8365: Add the AFE driver support")
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Link: https://patch.msgid.link/20251020170440.585-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
