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>
commit 668abe6dc7 upstream.
The quirk table entries should be put in the USB ID order, but some
entries have been put in random places. Re-sort them.
Fixes: bf990c1023 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue")
Fixes: fd28941cff ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless")
Fixes: dfd5fe19db ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless")
Fixes: 4a63e68a29 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.")
Fixes: 7822baa844 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Fixes: 4fb7c24f69 ("ALSA: usb-audio: Add quirk for Fiero SC-01")
Fixes: 2307a0e1ca ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)")
Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7822baa844 upstream.
The RODE NT-USB+ is marketed as a professional usb microphone, however the
usb audio interface is a mess:
[ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd
[ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4
[ 1.503912] usb 1-5: config 1 has no interface number 4
[ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09
[ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 1.519697] usb 1-5: Product: RØDE NT-USB+
[ 1.519699] usb 1-5: Manufacturer: RØDE
[ 1.519700] usb 1-5: SerialNumber: 1D773A1A
[ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82
[ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82
[ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2
Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get.
I have asked Rode support to fix it, but they show no interest.
Signed-off-by: Sean Young <sean@mess.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
I own an external usb Webcam, model NexiGo N930AF, which had low mic volume and
inconsistent sound quality. Video works as expected.
(snip)
[ +0.047857] usb 5-1: new high-speed USB device number 2 using xhci_hcd
[ +0.003406] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17
[ +0.000007] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam
[ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD
[ +0.000004] usb 5-1: SerialNumber: 20201217011
[ +0.003900] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283)
[ +0.025726] usb 5-1: 3:1: cannot get usb sound sample rate freq at ep 0x86
[ +0.071482] usb 5-1: 3:2: cannot get usb sound sample rate freq at ep 0x86
[ +0.004679] usb 5-1: 3:3: cannot get usb sound sample rate freq at ep 0x86
[ +0.051607] usb 5-1: Warning! Unlikely big volume range (=4096), cval->res is probably wrong.
[ +0.000005] usb 5-1: [7] FU [Mic Capture Volume] ch = 1, val = 0/4096/1
Set up quirk cval->res to 16 for 256 levels,
Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate.
Confirmed that happened anyway later due to the backoff mechanism, after 3 failures
All audio stream on device interfaces share the same values,
apart from wMaxPacketSize and tSamFreq :
(snip)
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 3
bAlternateSetting 3
bNumEndpoints 1
bInterfaceClass 1 Audio
bInterfaceSubClass 2 Streaming
bInterfaceProtocol 0
iInterface 0
AudioStreaming Interface Descriptor:
bLength 7
bDescriptorType 36
bDescriptorSubtype 1 (AS_GENERAL)
bTerminalLink 8
bDelay 1 frames
wFormatTag 0x0001 PCM
AudioStreaming Interface Descriptor:
bLength 11
bDescriptorType 36
bDescriptorSubtype 2 (FORMAT_TYPE)
bFormatType 1 (FORMAT_TYPE_I)
bNrChannels 1
bSubframeSize 2
bBitResolution 16
bSamFreqType 1 Discrete
tSamFreq[ 0] 44100
Endpoint Descriptor:
bLength 9
bDescriptorType 5
bEndpointAddress 0x86 EP 6 IN
bmAttributes 5
Transfer Type Isochronous
Synch Type Asynchronous
Usage Type Data
wMaxPacketSize 0x005c 1x 92 bytes
bInterval 4
bRefresh 0
bSynchAddress 0
AudioStreaming Endpoint Descriptor:
bLength 7
bDescriptorType 37
bDescriptorSubtype 1 (EP_GENERAL)
bmAttributes 0x01
Sampling Frequency
bLockDelayUnits 0 Undefined
wLockDelay 0x0000
(snip)
Based on the usb data about manufacturer, SPCA2281B3 is the most likely controller IC
Manufacturer does not provide link for datasheet nor detailed specs.
No way to confirm if the firmware supports any other way of getting the sample rate.
Testing patch provides consistent good sound recording quality and volume range.
(snip)
[ +0.045764] usb 5-1: new high-speed USB device number 2 using xhci_hcd
[ +0.106290] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17
[ +0.000006] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam
[ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD
[ +0.000004] usb 5-1: SerialNumber: 20201217011
[ +0.043700] usb 5-1: set resolution quirk: cval->res = 16
[ +0.002585] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283)
Signed-off-by: Christos Skevis <xristos.thes@gmail.com>
Link: https://lore.kernel.org/r/20231006155330.399393-1-xristos.thes@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a Opencomm2 Headset is connected to a Bluetooth USB dongle,
the audio playback functions properly, but the microphone does not work.
In the dmesg logs, there are messages indicating that the init_pitch
function fails when the capture process begins.
The microphone only functions when the ep pitch control is not set.
Toggling the pitch control off bypasses the init_piatch function
and allows the microphone to work.
Signed-off-by: WhaleChang <whalechang@google.com>
Link: https://lore.kernel.org/r/20231006044852.4181022-1-whalechang@google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft Modern Wireless Headset (appearing on the host as "Microsoft
USB Link") has a playback and a capture mixer volume/switch, but they
are fairly broken. The descriptor reports wrong dB ranges for
playback, and the capture volume/switch don't influence on the actual
recording at all. Moreover, there seem instabilities in the
connection, and at best, we should disable the runtime PM.
So this ended up with a quirk entry for:
- Correct the playback dB range;
I picked up some reasonable values but it's a guess work
- Disable the capture mixer;
it's completely useless and confuses PA/PW
- Suppress get-sample-rate, apply the delay for message handling,
and suppress the auto-suspend
The behavior of the wheel control on the headset is somehow flaky,
too, but it's an issue of HID.
Link: https://bugzilla.suse.com/show_bug.cgi?id=1207129
Link: https://lore.kernel.org/r/20230725092057.15115-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for
HEM devices which supports native DSD. Prior to this change Linux kernel
was not enabling native DSD playback for HEM devices, and as a result,
DSD audio was being converted to PCM "on the fly". HEM devices,
when connected to the system, would only play audio in PCM format,
even if the source material was in DSD format. With the addition of new
VENDOR_FLG in the quircks.c file, the devices are now correctly
recognized, and raw DSD data is transmitted to the device,
allowing for native DSD playback.
Signed-off-by: Lukasz Tyl <ltyl@hem-e.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230614122524.30271-1-ltyl@hem-e.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch provides a basic support for USB MIDI 2.0. As of this
patch, the driver creates a UMP device per MIDI I/O endpoints, which
serves as a dumb terminal to read/write UMP streams.
A new Kconfig CONFIG_SND_USB_AUDIO_MIDI_V2 manages whether to enable
or disable the MIDI 2.0 support. Also, the driver provides a new
module option, midi2_enable, to allow disabling the MIDI 2.0 at
runtime, too. When MIDI 2.0 support is disabled, the driver tries to
fall back to the already existing MIDI 1.0 device (each MIDI 2.0
device is supposed to provide the MIDI 1.0 interface at the altset
0).
For now, the driver doesn't manage any MIDI-CI or other protocol
setups by itself, but relies on the default protocol given via the
group terminal block descriptors.
The MIDI 1.0 messages on MIDI 2.0 device will be automatically
converted in ALSA sequencer in a later patch. As of this commit, the
driver accepts merely the rawmidi UMP accesses.
The driver builds up the topology in the following way:
- Create an object for each MIDI endpoint belonging to the USB
interface
- Find MIDI EP "pairs" that share the same GTB;
note that MIDI EP is unidirectional, while UMP is (normally)
bidirectional, so two MIDI EPs can form a single UMP EP
- A UMP endpoint object is created for each I/O pair
- For remaining "solo" MIDI EPs, create unidirectional UMP EPs
- Finally, parse GTBs and fill the protocol bits on each UMP
So the driver may support multiple UMP Endpoints in theory, although
most devices are supposed to have a single UMP EP that can contain up
to 16 groups -- which should be large enough.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We're going to create rawmidi objects for MIDI 2.0 in a different code
from the current code for USB-MIDI 1.0. As a preliminary work, this
patch adds the number of rawmidi objects to keep globally in a
USB-audio card instance, so that it can be referred from both MIDI 1.0
and 2.0 code.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tascam's Model 12 is a mixer which can also operate as a USB audio
interface. The audio interface uses explicit feedback but it seems that
it does not correctly handle missing isochronous frames.
When injecting an xrun (or doing anything else that pauses the playback
stream) the feedback rate climbs (for example, at 44,100Hz nominal, I
see a stable rate around 44,099 but xrun injection sees this peak at
around 44,135 in most cases) and glitches are heard in the audio stream
for several seconds - this is significantly worse than the single glitch
expected for an underrun.
While the stream does normally recover and the feedback rate returns to
a stable value, I have seen some occurrences where this does not happen
and the rate continues to increase while no audio is heard from the
output. I have not found a solid reproduction for this.
This misbehaviour can be avoided by totally resetting the stream state
by switching the interface to alt 0 and back before restarting the
playback stream.
Add a new quirk flag which forces the endpoint and interface to be
reconfigured whenever the stream is stopped, and use this for the Tascam
Model 12.
Separate interfaces are used for the playback and capture endpoints, so
resetting the playback interface here will not affect the capture stream
if it is running. While there are two endpoints on the interface,
these are the OUT data endpoint and the IN explicit feedback endpoint
corresponding to it and these are always stopped and started together.
Signed-off-by: John Keeping <john@metanate.com>
Link: https://lore.kernel.org/r/20221129130100.1257904-1-john@metanate.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB-audio driver matches per interface, and as default, it
registers the card instance at the very first instance. This can be a
problem for the devices that have multiple interfaces to be probed, as
the udev rule isn't applied properly for the later appearing
interfaces. Although we introduced the delayed_register option and
the quirks for covering those shortcomings, it's nothing but a
workaround for specific devices.
This patch is an another attempt to fix the problem in a more generic
way. Now the driver checks the whole USB device descriptor at the
very first time when an interface is attached to a sound card. It
looks at each matching interface in the descriptor and remembers the
last matching one. The snd_card_register() is invoked only when this
last interface is probed.
After this change, the quirks for the delayed registration become
superfluous, hence they are removed along with the patch. OTOH, the
delayed_register option is still kept, as it might be useful for some
corner cases (e.g. a special driver overtakes the interface probe from
the standard driver, and the last interface probe may miss).
Link: https://lore.kernel.org/r/20220904161247.16461-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the delayed registration is specified via either delayed_register
option or the quirk, we delay the invocation of snd_card_register()
until the given interface. But if a wrong value has been set there
and there are more interfaces over the given interface number,
snd_card_register() call would be missing for those interfaces.
This patch catches up those missing calls by fixing the comparison of
the interface number. Now the call is skipped only if the processed
interface is less than the given interface, instead of the exact
match.
Fixes: b70038ef4f ("ALSA: usb-audio: Add delayed_register option")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082
Link: https://lore.kernel.org/r/20220831125901.4660-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch applies the same quirks used for SC-01 at firmware v1.1.0 to
the ones running v1.0.0, with respect to hard-coded sample rates.
I got two more units and successfully tested the patch series with both
firmwares.
The support is now complete (not accounting ASIO).
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-2-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fiero SC-01 is a USB sound card with two mono inputs and a single
stereo output. The inputs are composed into a single stereo stream.
The device uses a vendor-provided driver on Windows and does not work
at all without it. The driver mostly provides ASIO functionality, but
also alters the way the sound card is queried for sample rates and
clocks.
ALSA queries those failing with an EPIPE (same as Windows 10 does).
Presumably, the vendor-provided driver does not query it at all, simply
matching by VID:PID. Thus, I consider this a buggy firmware and adhere
to a set of fixed endpoint quirks instead.
The soundcard has an internal clock. Implicit feedback mode is required
for the playback.
I have updated my device to v1.1.0 from a Windows 10 VM using a vendor-
provided binary prior to the development, hoping for it to just begin
working. The device provides no obvious way to downgrade the firmware,
and regardless, there's no binary available for v1.0.0 anyway.
Thus, I will be getting another unit to extend the patch with support
for that. Expected to be a simple copy-paste of the existing one,
though.
There were no previous reports of that device in context of Linux
anywhere. Other issues have been reported though, but that's out of the
scope.
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-1-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both Behringer UMC 202 HD and 404 HD need explicit quirks to enable
the implicit feedback mode and start the playback stream primarily.
The former seems fixing the stuttering and the latter is required for
a playback-only case.
Note that the "clock source 41 is not valid" error message still
appears even after this fix, but it should be only once at probe.
The reason of the error is still unknown, but this seems to be mostly
harmless as it's a one-off error and the driver retires the clock
setup and it succeeds afterwards.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220624101132.14528-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a Jieli Technology USB Webcam is connected, the video part works
well, but the mic sound is speeded up. On dmesg there are messages
about different rates from the runtime rates, warnings about volume
resolution and lastly, the log is filled, every 5 seconds, with
retire_capture_urb error messages.
The mic works only when ep packet size is set to wMaxPacketSize (normal
sound and no more retire_capture_urb error messages). Skipping reading
sample rate, fixes the messages about different rates and forcing a volume
resolution, fixes warnings about volume range. I have arbitrarily choosed
the value (16): I read in a comment that there should be no more than 255
levels, so 4096 (max volume) / 16 = 0-255.
Signed-off-by: Marco Giunta <giun7a@gmail.com>
Link: https://lore.kernel.org/r/20211018162552.12082-1-giun7a@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Shciit Hel device responds to the ctl message for the mic capture
switch with a timeout of -EPIPE:
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
This seems safe to ignore as the device works properly with the control
message quirk, so add it to the quirk table so all is good.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-usb@vger.kernel.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Scarlett device series from Focusrite Novation seem requiring the
sample rate validations as we've done for MOTU devices; otherwise the
driver probes invalid audioformat entries that contain the sample
rates that actually don't work, and this may result in an incomplete
setup as reported recently.
This patch adds the needed quirk flag for enabling the sample rate
validation for Focusrite Novation devices.
Fixes: fe773b8711 ("ALSA: usb-audio: workaround for iface reset issue")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214493
Link: https://lore.kernel.org/r/20211004074050.28241-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent quirk for WALKMAN (commit 7af5a14371: "ALSA: usb-audio:
Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other
devices and is worth to be put into the common quirk flags.
This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a
quirk table entry for the device.
Link: https://lore.kernel.org/r/20210824055720.9240-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer code has a flag ignore_ctl_error for ignoring the errors
returned from the device wrt mixer accesses, and this is set from the
entries in mixer_maps.c, as well as ignore_ctl_error module option.
Those can be well integrated into the new quirk_flags field, too.
Link: https://lore.kernel.org/r/20210729074404.19728-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We apply some delay for the control messages on certain devices as a
workaround, and this can be moved into the quirk_flags as well.
Currently there are three different delay periods (1ms, 5ms and 20ms),
so three different quirk bits are assigned for them.
Link: https://lore.kernel.org/r/20210729073855.19043-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
