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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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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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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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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>
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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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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Adds calls to disable regmap cache-only after a successful return from
cs35l56_wait_for_firmware_boot().
This is to prepare for a change in the shared ASoC module that will
leave regmap in cache-only mode after cs35l56_system_reset(). This is
to prevent register accesses going to the hardware while it is
rebooting.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240408101803.43183-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is no need for it to be 32 samples - 3 will do just fine (which is
the interpolator's epsilon). The old size was presumably meant to
compensate for the cache's presence, but we're now handling that
properly.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-17-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Compensate for the cache lag of 64 frames, and actually populate the
cache. Without these, the playback would start with garbage (which
would be (mostly?) masqueraded by the note's attack phase).
Note that we set the starting address only 61 frames ahead, to
compensate for the interpolator's epsilon. Unlike for PCM playback, we
don't even need to manually silence-fill the first frames in the cache,
because we insert some silence in front of each sample anyway.
A challenge are extremely short samples with a loop end below the cache
size, because a) we'd have to wrap the current address to be within the
loop and b) automatic pre-filling of the cache with the right data does
not work in this case.
We could pre-fill the cache manually, but that's slow, requires
additional code for each sample width, and is made even more complex by
the driver's virtual address space having no contiguous mapping for the
CPU.
We could have the engine fill the cache piece-wise (which is really what
happens when playback is running), but that would also be complex, and
we'd need to wait for the engine to handle each piece, so it wouldn't be
that much faster than the manual fill.
For the case of requiring only one loop iteration prior to reaching the
cache size, we could leverage the engine's looping mechanism around
CCR_CACHELOOPFLAG, but this special case doesn't seem worth the
complexity.
So we just unroll the loop as far as necessary to be able to play back
the sample without any fiddling.
Pedantically, this would be incorrect for loop-until-release samples
with a low loop end which are released very quickly, but that would be
relatively harmless, is not a plausible use case in the first place, and
SoundFont sample mode 3 isn't actually implemented anyway (it's
conflated with mode 1, infinite looping).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-16-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of repeatedly checking the sample width, assign a size shift
centrally.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-14-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The offsets are counted in samples, not in bytes.
While the code block is being rewritten, also move it up a bit, to avoid
churn in a subsequent patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-13-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This de-duplicates the code slightly. But the real reason is that it
moves the code up, which the next patch will depend on.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-12-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Samples are byte-sized in this mode, and thus the offset calculation
needs no shifting.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-11-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The hardware supports S16LE and U8 samples, while U16LE and S8 (which
the driver implicitly claims to support) require sign flipping.
Note that this matters only for the GUS patch loader, as the implemented
SoundFont v2.01 spec is limited to S16LE.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert some checks in snd_emu10k1_sample_new() back into assertions (as
they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*,
2008-08-08)), and move them into the low-level memory access functions
they protect.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-9-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This does several closely related things:
- Move the code from the drivers into the SoundFont loader, which
de-duplicates it.
- Sort of explain the weird "recalculate address offset" feature. Note
that I don't think it actually makes any sense - the calling user
space code should do that. The background is certainly that the source
data (the SoundFont format) uses pointers into a single wave block
(and the API allows doing the same for on-board ROM), but the API
expects the wave data from user space to be pre-chopped into
individual patches anyway.
- Make sure that the specified offsets actually lie within the supplied
wave data. Note that we don't validate ROM offsets, so one can play
back anything within the sound card's address space.
- In load_guspatch(), don't call the sample_new callback anymore when
the patch size is zero, as was already the case in load_data(). The
callbacks would instantly return in that case anyway; these checks are
now removed.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-7-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in
the GUS patch loader. It has not worked on emu10k1 since before ALSA hit
mainline, yet nobody appears to have complained. And as it isn't super
easy to implement, just admit defeat and clean up the code.
If somebody wanted to resurrect the feature, the emu8k driver could
serve as a template, but the code would be quite different. But
arguably, this should be done in user space in the first place, as this
doesn't represent a hardware feature (somewhat ironically, the actual
GUS driver has no synth support, and therefore no GUS patch loader).
Note that instead of properly rejecting affected samples, we continue to
just pretend that the feature wasn't requested. This is extremely
questionable behavior, but avoids that possibly unused instruments
suddenly prevent loading the entire file, which would break backwards
compatibility. But at least we log a warning now.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-6-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Calibrated data was written into an incorrect register, which cause
speaker protection sometimes malfuctions
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240406132010.341-1-shenghao-ding@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add support for HP SnowWhite laptops with CS35L51 amplifiers on I2C
bus connected to Realtek codec.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Message-ID: <20240405210635.22193-1-vitalyr@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Set of changes targeting the avs-driver only. No new features, patchset
either fixes or fortifies existing code.
Patchset starts off with a fix for debugbility on ICL+ platforms which I
have forgotten to fixup when providing support for these initially.
The next two address copier module initialization, most importantly,
silence the gcc 'field-spanning write' false-positive.
The following four:
6/13 ASoC: Intel: avs: Replace risky functions with safer variants
7/13 ASoC: Intel: avs: Fix potential integer overflow
8/13 ASoC: Intel: avs: Test result of avs_get_module_entry()
9/13 ASoC: Intel: avs: Remove dead code
address problems found out by Coverity static analysis tool.
The last two worth mentioning are: recommendation from the firmware team
to wake subsystem from D0ix when starting any pipeline -and- shielding
against invalid period/buffer sizes. Audio format shall be taken into
consideration when calculating either of these.
Amadeusz Sławiński (2):
ASoC: Intel: avs: Restore stream decoupling on prepare
ASoC: Intel: avs: Add assert_static to guarantee ABI sizes
Cezary Rojewski (11):
ASoC: Intel: avs: Fix debug-slot offset calculation
ASoC: Intel: avs: Silence false-positive memcpy() warnings
ASoC: Intel: avs: Fix config_length for config-less copiers
ASoC: Intel: avs: Fix ASRC module initialization
ASoC: Intel: avs: Replace risky functions with safer variants
ASoC: Intel: avs: Fix potential integer overflow
ASoC: Intel: avs: Test result of avs_get_module_entry()
ASoC: Intel: avs: Remove dead code
ASoC: Intel: avs: Wake from D0ix when starting streaming
ASoC: Intel: avs: Init debugfs before booting firmware
ASoC: Intel: avs: Rule invalid buffer and period sizes out
sound/soc/intel/avs/avs.h | 1 +
sound/soc/intel/avs/cldma.c | 2 +-
sound/soc/intel/avs/core.c | 4 +--
sound/soc/intel/avs/icl.c | 12 ++++++---
sound/soc/intel/avs/loader.c | 6 +++--
sound/soc/intel/avs/messages.h | 47 ++++++++++++++++++++++++++++++++--
sound/soc/intel/avs/path.c | 13 ++++------
sound/soc/intel/avs/pcm.c | 34 +++++++++++++++++++++++-
sound/soc/intel/avs/probes.c | 14 ++++++----
9 files changed, 109 insertions(+), 24 deletions(-)
--
2.25.1
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Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
This set of patches factors out some repeated code to clean up
firmware control read/write functions, and removes some redundant
control notification code.
base-commit: f193957b0fbbba397c8bddedf158b3bf7e4850fc
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Any control that the driver is updating should be marked as SYSTEM and
therefore will not have an ALSA control to notify.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Using the cs_dsp_coeff_lock_and_[read|write]_ctrl() wrappers tidies
the calling functions as it does not need to manage the DSP pwr_lock.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fixes the realtek quirk to initialise the Cirrus amp correctly and adds
related quirk for missing DSD properties. This model laptop has slightly
updated internals compared to the previous version with Realtek Codec
ID of 0x1caf.
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Message-ID: <20240402015126.21115-1-luke@ljones.dev>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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microphone
This patch addresses an issue with the Panasonic CF-SZ6's existing quirk,
specifically its headset microphone functionality. Previously, the quirk
used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design
of a single 3.5mm jack for both mic and audio output effectively. The
device uses pin 0x19 for the headset mic without jack detection.
Following verification on the CF-SZ6 and discussions with the original
patch author, i determined that the update to
ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change
is custom-designed for the CF-SZ6's unique hardware setup, which includes
a single 3.5mm jack for both mic and audio output, connecting the headset
microphone to pin 0x19 without the use of jack detection.
Fixes: 0fca97a29b83 ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk")
Signed-off-by: I Gede Agastya Darma Laksana <gedeagas22@gmail.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This fixes the sound not working from internal speakers on
Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID
have been added to cs35l41_hda_property.c and patch_realtek.c.
Signed-off-by: Christian Bendiksen <christian@bendiksen.me>
Cc: <stable@vger.kernel.org>
Message-ID: <20240401122603.6634-1-christian@bendiksen.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As already anticipated in the original commit, playback was broken for
very short samples. I just didn't expect it to be an actual problem,
because we're talking about less than 1.5 milliseconds here. But clearly
such wavetable samples do actually exist.
The problem was that for such short samples we'd set the current
position beyond the end of the loop, so we'd run off the end of the
sample and play garbage.
This is a bigger (more audible) problem than the original one, which was
that we'd start playback with garbage (whatever was still in the cache),
which would be mostly masked by the note's attack phase.
So revert to the old behavior for now. We'll subsequently fix it
properly with a bigger patch series.
Note that this isn't a full revert - the dead code is not re-introduced,
because that would be silly.
Fixes: df335e9a8bcb ("ALSA: emu10k1: fix synthesizer sample playback position and caching")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These ASUS laptops use the Realtek HDA codec combined with a number of
CS35L56 amplifiers.
The SSID of the GA403U matches a previous ASUS laptop - we can tell them
apart because they use different codecs.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Message-ID: <20240329112803.23897-1-simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules
to be loaded on boot so that Serial Multi Instantiate can add the
devices to the bus and begin the driver init sequence.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240328121355.18972-1-simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds the existing fixup to certain TF platforms implementing
the ALC274 codec with a headset jack. It fixes/activates the inactive
microphone of the headset.
Signed-off-by: Christoffer Sandberg <cs@tuxedo.de>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240328102757.50310-1-wse@tuxedocomputers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The debug message "Playback action not supported: action" is not useful,
because the action was previously printed, and the list of supported
actions are intentional.
Remove the debug statement from the default switch case.
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <8b9546db6c92dea4476a7247a88d56248c2ba8c2.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Sometimes it is useful to examine the timing of kcontrol events.
Add debug statements to each kcontrol.
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <18ff4b0caab90a2dacf907e62346fd5079a9eb1a.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The rcabin.profile_cfg_id, cur_prog, cur_conf, force_fwload_status
variables are acccessible from multiple threads and therefore require
locking.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <e35b867f6fe5fa1f869dd658a0a1f2118b737f57.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A)
register. Unfortunately the tas2563 does not have DVC_LVL, but has
INT_MASK0 in 0x1A, which has been misused so far.
Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer
volume control") the volume of the tas2781 amplifiers can be controlled
by the master volume, so this digital gain kcontrol is not needed.
Remove it.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <741fc21db994efd58f83e7aef38931204961e5b2.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Initialization is completed before adding the component as that can
start the process of the device binding and trigger actions that check
init_done.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240325145510.328378-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The system and amplifier names influence which firmware and tuning files
are downloaded to the device; log these values to aid end-user system
support.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240325142937.257869-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add audio quirks to fix speaker output and headset detection on some new
Clevo models:
- L240TU (ALC245)
- PE60SNE-G (ALC1220)
- V350SNEQ (ALC245)
Co-authored-by: Jeremy Soller <jeremy@system76.com>
Signed-off-by: Tim Crawford <tcrawford@system76.com>
Message-ID: <20240319212726.62888-1-tcrawford@system76.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cirrus amps support for this laptop was added in patch:
33e5e648e631 ("ALSA: hda: cs35l41: Support additional HP Envy Models")
This patch adds fixes for wrong pincfgs, wrong DAC selection and
mute/micmute LEDs.
Signed-off-by: Anthony I Gilea <i@cpp.in>
Message-ID: <e2a7aaed-e9d7-4d36-8abf-b71dfd32a0ff@cpp.in>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we tested the headphone playback on 2 LG machines, if we set
the volume to the max value or near to the max value, the sound is too
loud, it could even bring harm to listeners.
A workaround is to decrease the max volume to a reasonable value for
the headphone's amplifier, then the users couldn't set the volume
bigger than that value from the userspace.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Message-ID: <20240318011128.156023-1-hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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