| Age | Commit message (Collapse) | Author |
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We did not delay after the second strobe signal, so another immediately
following access could potentially corrupt the written value.
This is a purely speculative fix with no supporting evidence, but after
taking out the spinlocks around the writes, it seems plausible that a
modern processor could be actually too fast. Also, it's just cleaner to
be consistent.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-7-oswald.buddenhagen@gmx.de>
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A side effect of making the dock monitoring interrupt-driven was that
we'd be very quick to program a freshly connected dock. However, for
unclear reasons, the dock does not work when we do that - despite the
FPGA netlist upload going just fine. We work around this by adding a
delay before programming the dock; for safety, the value is several
times as much as was determined empirically.
Note that a badly timed dock hot-plug would have triggered the problem
even before the referenced commit - but now it would happen 100% instead
of about 3% of the time, thus making it impossible to work around by
re-plugging.
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-6-oswald.buddenhagen@gmx.de>
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The FPGA access through the GPIO port does not interfere with other
sound processor register access, so there is no need to subject it to
emu_lock. And after moving all FPGA access out of the interrupt handler,
it does not need to be IRQ-safe, either.
What's more, attaching the dock causes a firmware upload, which takes
several seconds. We really don't want to disable IRQs for this long, and
even less also have someone else spin with IRQs disabled waiting for us.
Therefore, use a mutex for FPGA access locking.
This makes the code somewhat more noisy, as we need to wrap bigger
sections into the mutex, as it needs to enclose the spinlocks.
The latter has the "side effect" of fixing dock FPGA programming in a
corner case: a really badly timed mixer access right between entering
FPGA programming mode and uploading the netlist would mess up the
protocol.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-5-oswald.buddenhagen@gmx.de>
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The actual event processing was already done by workqueue items. We can
move the event dispatching there as well, rather than doing it already
in the interrupt handler callback.
This change has a rather profound "side effect" on the reliability of
the FPGA programming: once we enter programming mode, we must not issue
any snd_emu1010_fpga_{read,write}() calls until we're done, as these
would badly mess up the programming protocol. But exactly that would
happen when trying to program the dock, as that triggers GPIO interrupts
as a side effect. This is mitigated by deferring the actual interrupt
handling, as workqueue items are not re-entrant.
To avoid scheduling the dispatcher on non-events, we now explicitly
ignore GPIO IRQs triggered by "uninteresting" pins, which happens a lot
as a side effect of calling snd_emu1010_fpga_{read,write}().
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-4-oswald.buddenhagen@gmx.de>
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Pulled out of the next patch to improve its legibility.
As the function is now available, call it directly from
snd_emu10k1_emu1010_init(), thus making the MicroDock firmware loading
synchronous - there isn't really a reason not to. Note that this does
not affect the AudioDocks of rev1 cards, as these have no independent
power supplies, and thus come up only a while after the main card is
initialized.
As a drive-by, adjust the priorities of two messages to better reflect
their impact.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-3-oswald.buddenhagen@gmx.de>
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While there are two separate IRQ status bits for dock attach and detach,
the hardware appears to mix them up more or less randomly, making them
useless for tracking what actually happened. It is much safer to check
the dock status separately and proceed based on that, as the old polling
code did.
Note that the code assumes that only the dock can be hot-plugged - if
other option card bits changed, the logic would break.
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-2-oswald.buddenhagen@gmx.de>
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Some data for testing is immutable. In the case, the const qualifier is
available for any loader to place it to read-only segment.
Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240425233653.218434-1-o-takashi@sakamocchi.jp>
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Don't populate the read-only array buf_samples on the stack at
run time, instead make it static const.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Acked-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240425160754.114716-1-colin.i.king@gmail.com>
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Instead of reiterating the list, use list_for_each_entry_safe()
that allows to continue without starting over.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Message-ID: <20240424145020.1057216-1-andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add laptop using CS35L41 HDA.
This laptop does not have _DSD, so require entries in property
configuration table for cs35l41_hda driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Message-ID: <20240423162303.638211-3-sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This laptop does not have the correct _DSD settings, so needs to
obtain its configuration from the configuration table.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Message-ID: <20240423162303.638211-2-sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Volume step (dB/step) modification to fix format error
which shown in amixer control.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/b1f546ad16dc4c7abb7daa7396e8345c@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Static checkers complain that the silicon_uid variable passed by
pointer to cs35l56_read_silicon_uid() could later be used
uninitialised when calling cs_amp_get_efi_calibration_data().
cs35l56_read_silicon_uid() must have succeeded to call
cs_amp_get_efi_calibration_data() and that would have populated the
variable.
However, initialise the value so we are not haunted by it forevermore.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240422103211.236063-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Clang warns (or errors with CONFIG_WERROR):
sound/usb/mixer_scarlett2.c:3697:6: error: variable 'err' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3707:9: note: uninitialized use occurs here
3707 | return err;
| ^~~
sound/usb/mixer_scarlett2.c:3697:2: note: remove the 'if' if its condition is always true
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3688:9: note: initialize the variable 'err' to silence this warning
3688 | int err;
| ^
| = 0
1 error generated.
Initialize ret to zero to ensure ret is initialized in all paths within
scarlett2_ag_target_ctl_get(), which matches the style of other
functions in this driver.
Fixes: e30ea5340c25 ("ALSA: scarlett2: Add autogain target controls")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Message-ID: <20240419-alsa-scarlett2-fix-wsometimes-uninitialized-v1-1-e2ace8642e08@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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WSA881x codecs do not retain the state while clock is stopped, so mark
this with clk_stop_mode1 flag.
Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20240419140012.91384-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Although the purpose of dummy seq client is a direct pass-through,
it's sometimes helpful for debugging if it can convert to a certain
UMP MIDI version. This patch adds an option to specify the UMP event
conversion. As default, it skips the conversion and does
passthrough, while user can pass ump=1 or ump=2 to enforce the
conversion to UMP MIDI1 or MIDI2 format.
Message-ID: <20240419101105.15571-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The conversion from MIDI2 to MIDI1 UMP messages had a leftover
artifact (superfluous bit shift), and this resulted in the bogus type
check, leading to empty outputs. Let's fix it.
Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events")
Cc: <stable@vger.kernel.org>
Link: https://github.com/alsa-project/alsa-utils/issues/262
Message-ID: <20240419100442.14806-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Haier Boyue G42 with ALC269VC cannot detect the MIC of headset,
the line out and internal speaker until
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Message-ID: <20240419082159.476879-1-aichao@kylinos.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When using davinci-mcasp as CPU DAI with simple-card, there are some
conditions that cause simple-card to finish registering a sound card before
davinci-mcasp finishes registering all sound components. This creates a
non-working sound card from userspace with no problem indication apart
from not being able to play/record audio on a PCM stream. The issue
arises during simultaneous probe execution of both drivers. Specifically,
the simple-card driver, awaiting a CPU DAI, proceeds as soon as
davinci-mcasp registers its DAI. However, this process can lead to the
client mutex lock (client_mutex in soc-core.c) being held or davinci-mcasp
being preempted before PCM DMA registration on davinci-mcasp finishes.
This situation occurs when the probes of both drivers run concurrently.
Below is the code path for this condition. To solve the issue, defer
davinci-mcasp CPU DAI registration to the last step in the audio part of
it. This way, simple-card CPU DAI parsing will be deferred until all
audio components are registered.
Fail Code Path:
simple-card.c: probe starts
simple-card.c: simple_dai_link_of: simple_parse_node(..,cpu,..) returns EPROBE_DEFER, no CPU DAI yet
davinci-mcasp.c: probe starts
davinci-mcasp.c: devm_snd_soc_register_component() register CPU DAI
simple-card.c: probes again, finish CPU DAI parsing and call devm_snd_soc_register_card()
simple-card.c: finish probe
davinci-mcasp.c: *dma_pcm_platform_register() register PCM DMA
davinci-mcasp.c: probe finish
Cc: stable@vger.kernel.org
Fixes: 9fbd58cf4ab0 ("ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller")
Signed-off-by: Joao Paulo Goncalves <joao.goncalves@toradex.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Reviewed-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240417184138.1104774-1-jpaulo.silvagoncalves@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When creating controls attached to widgets, there are a lot of rules if
they get their name prefixed with widget name or not. Due to that
controls ended up with weirdly looking names like "ssp0_fe DSP Volume",
while topology set it to "DSP Volume".
Fix this by setting no_wname_in_kcontrol_name to true in avs topology
widgets which disables unwanted behaviour.
Fixes: be2b81b519d7 ("ASoC: Intel: avs: Parse control tuples")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240418142621.2487478-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The headset mic requires a fixup to be properly detected/used.
As a reference, this specific model from 2021 reports
the following devices:
https://alsa-project.org/db/?f=1a5ddeb0b151db8fe051407f5bb1c075b7dd3e4a
Signed-off-by: Mauro Carvalho Chehab <mchehab@kernel.org>
Cc: <stable@vger.kernel.org>
Message-ID: <b92a9e49fb504eec8416bcc6882a52de89450102.1713370457.git.mchehab@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Many modern codecs support rates up to 768kHz (including DSD1024). Add
support for rates up to 768kHz to the loopback driver.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-4-pavel.hofman@ivitera.com>
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Many modern codecs support 705.6kHz and 768kHz sample rates. Current HW
params fail to set 705.6kHz and 768kHz sample rates as these are not in the
known-rates list.
Add these new rates to the known-rates list to allow them.
Also add defines in pcm.h so that drivers can use it.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-3-pavel.hofman@ivitera.com>
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The snd-aloop loopback driver does not modify or access the actual samples
in any way, defines no volume or mute controls, it's strictly bitperfect.
Therefore DSD formats can be supported without any modification.
Add all DSD formats to the list of supported formats.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-2-pavel.hofman@ivitera.com>
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change HDA & AMP configuration from ALC287_FIXUP_CS35L41_I2C_2 to
ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD for ThinkBook 16P Gen4
models to fix volumn control issue (cannot fully mute).
Signed-off-by: Huayu Zhang <zhanghuayu1233@qq.com>
Fixes: 6214e24cae9b ("ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops")
Message-ID: <tencent_37EB880C5E5BD99D21C16B288115C4545F06@qq.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added the correct pin table for Asus GU605M and GA403U, enabling all
speakers to be controlled with the master.
Updated quirks for GU605M and GA403U by including the pin table patch
in the chain.
Co-developed-by: Luke D. Jones <luke@ljones.dev>
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Signed-off-by: Vitalii Torshyn <vitaly.torshyn@gmail.com>
Message-ID: <20240411125803.18539-1-vitaly.torshyn@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These laptops do not have _DSD and must be added by configuration
table, however, the initial entries for them are incorrect:
Neither laptop contains a Speaker ID GPIO.
This issue would not affect audio playback, but may affect which files
are loaded when loading firmware.
Fixes: b67a7dc418aa ("ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-8-sbinding@opensource.cirrus.com>
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In every case the 'dir' argument to cs35l41_request_firmware_file() is passed
the string "cirrus/", so this is a redundant argument and can be removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-7-sbinding@opensource.cirrus.com>
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The original mechanism for applying calibration assumed that the
calibration data would be ordered the same as the amp instances.
However, for some 4 amp laptops, this is not the case.
To ensure that the correct calibration is applied to the correct amp,
the calibration data contains a unique id, which matches a unique id
inside the CS35L41. This can be used to match to the correct data
entry. This mechanism is available inside the shared module cs-amp-lib.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-6-sbinding@opensource.cirrus.com>
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Currently, all PC systems are set to use VBSTMON for DSP1RX5_SRC,
however, this is required only for external boost systems.
Internal boost systems require VPMON instead of VBSTMON to be the
input to DSP1RX5_SRC.
All systems require DSP1RX6_SRC to be set to VBSTMON.
Also fix incorrect comment for DACPCM1_SRC to use DSP1TX1.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-5-sbinding@opensource.cirrus.com>
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Add 4 laptops using CS35L41 HDA.
None of these laptops have _DSD, so require entries in property
configuration table for cs35l41_hda driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-4-sbinding@opensource.cirrus.com>
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Add support for 2 new HP Omen models without _DSD into configuration
table.
These laptops use the PCM Gain setting for the tuning setting file.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-3-sbinding@opensource.cirrus.com>
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Some systems requires different max PCM Gains settings than the default.
The current default value, when running firmware is 17.5 dB, which is
used for all systems. Some systems require lower values.
Value when running without firmware is 4.5 dB and remains unchanged.
Since the gain value is dependent on Tuning and Firmware, it can
change, so it cannot be saved in _DSD. Instead we can store it inside
a configuration binary file alongside the Firmware and Tuning files.
The gain value increments in steps of 1 dB, with value 0 representing
0.5 dB. The max value is 20, which corresponds to 20.5 dB.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-2-sbinding@opensource.cirrus.com>
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Add new vendor_id and subsystem_id to support new Lenovo laptop
ThinkPad ICE-1
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240411091823.1644-1-shenghao-ding@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It is recommended that on Lunar Lake the PIO (immediate command response)
is used instead of CORB/RIRB for commands/verbs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-6-peter.ujfalusi@linux.intel.com>
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It is recommended that on Lunar Lake the PIO (immediate command response)
is used instead of CORB/RIRB for commands/verbs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-5-peter.ujfalusi@linux.intel.com>
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Set the use_pio_for_commands flag in case AZX_DCAPS_PIO_COMMANDS quirk is
enabled.
When the PIO command mode is used we can re-use the existing
azx_single_send_cmd() / azx_single_get_response() functions safely as the
CORB DMA is not going to be enabled in snd_hdac_bus_init_cmd_io().
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-4-peter.ujfalusi@linux.intel.com>
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In case the use_pio_for_commands flag is set we must not enable the
CORB DMA to make sure that it is not interfering with the immediate
command mode.
Convert the snd_hdac_bus_send_cmd/snd_hdac_bus_get_response as wrappers to
call either the PIO or CORB based command handling depending on the
use_pio_for_commands flag.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-3-peter.ujfalusi@linux.intel.com>
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Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to
be able to select PIO mode as alternative for CORB based command sending
while retaining the RIRB functionality to receive unsolicited responses.
This mode differs from the azx single_cmd mode when RIRB is disabled.
The mixed mode is needed on Lunar Lake family because it is recommended to
use Immediate Command Response (PIO mode) instead of CORB for HDA commands.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-2-peter.ujfalusi@linux.intel.com>
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The Vocaster Two has a Bluetooth module with a volume control. Add a
corresponding ALSA mixer control.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <b78687f7243142a4466f63c0aee9742b44ee395d.1710264833.git.g@b4.vu>
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The Scarlett 4th Gen and Vocaster interfaces allow the autogain target
dBFS value(s) to be configured. Add Mean and Peak Target controls for
4th Gen, and a Hot Target control for Vocaster.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <33d7f6dc965ab09522361ec99745a0685e4b8272.1710264833.git.g@b4.vu>
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Add Focusrite Vocaster One and Two USB IDs, notification arrays,
config sets, and device info data.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <5fb48555a8db7bb322b25784b165829357cd6e42.1710264833.git.g@b4.vu>
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Add filter and compressor DSP controls for the Vocaster interfaces.
Mark scarlett2_notify_input_dsp() as __always_unused until it gets
used when the Vocaster callback function array is added.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <a45316f79600b862dae38da24f13def638b06476.1710264833.git.g@b4.vu>
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Add controls for the input mute switches that the Vocaster interfaces
have. Mark scarlett2_notify_input_mute() as __always_unused until it
gets used when the Vocaster callback function array is added.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <3b384b4e759241bd06f0c223e9f4f00467d88318.1710264833.git.g@b4.vu>
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The autogain status texts are different for Vocaster vs. Scarlett 4th
Gen, so make them configurable per-config-set.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <b1adcd3dc48117d4ebe16812eeb7f1dbf1ede472.1710264833.git.g@b4.vu>
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Remove the #define SCARLETT2_MAX_GAIN_DB and replace with a
per-config-set TLV as the Vocaster has a maximum gain of 70dB vs the
4th Gen 69dB.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <ade8e18ce38927ea0224946ec7cfea23ad3793d8.1710264833.git.g@b4.vu>
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The 4th Gen Scarlett interfaces added software-controllable input gain
along with channel select, channel link, auto-gain, and "safe" mode.
Vocaster has software-controllable input gain and auto-gain but not
channel select, channel link, or safe mode.
Add a device info field safe_input_count to indicate how many channels
have a safe mode control, and use the presence of the input select and
input link switch configuration parameters to determine if those
controls should be created.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <167f04a37d0fb23f3077705df835adbc4f2b6a8e.1710264833.git.g@b4.vu>
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Update scarlett2_usb_get_config() to support 32-bit values which are
needed by the upcoming Vocaster support.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <ee35dce0172b2aa3fec8163ab8f35bdc35a141bd.1710264833.git.g@b4.vu>
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scarlett2_usb_set_config() was using size = 0 as a signal to use the
parameter buffer. Replace that with an explicit indication (pbuf = 1),
as the upcoming Vocaster support has a config item written via the
parameter buffer with size = 1 rather than the implicit size of 8.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <50a7d85bb04f9a7f13f667c70a706826c8d3ef93.1710264833.git.g@b4.vu>
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The location pointed to by gen4_write_addr and gen4_write_addr + 1 is
officially known as the parameter buffer. Update the code to match.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <aa36ecb8d3ce67387b5edf6c900f0b8a509241ce.1710264833.git.g@b4.vu>
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