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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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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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Add hwdep read op so flash segments can be read.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <800d20a801e8c59c2905c82ecae5676cd4f31429.1710264833.git.g@b4.vu>
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After scarlett2_usb() sends a command, it seems that we should wait
for an ACK before attempting to read the response. Not doing that
didn't seem necessary previously but seems to be causing occasional
issues with 4th Gen devices.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <452d1263c40fa8eba1cfb24e2055e40a84cbc437.1710264833.git.g@b4.vu>
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So that more forward declarations won't be required when we add
handling of the ACK notification, move the initialisation functions to
after the notification functions.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <0922071cb8be99a2394705de27b917d1e4e46f3f.1710264833.git.g@b4.vu>
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Both drivers provide both sample_new and sample_free, and it makes no
sense to pretend that they could not. In fact, load_data() would already
crash if sample_new was null. So remove the remaining null checks.
Contrary to that, the emu10k1 driver actually has a null sample_reset,
though I'm not convinced that this inconsistency is justified.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-18-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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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Resulting from more reverse engineering in the course of debugging.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-15-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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In load_data(), make the validation of and skipping over the main info
block match that in load_guspatch().
In load_guspatch(), add checking that the specified patch length matches
the actually supplied data, like load_data() already did.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-8-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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The field is explicitly documented to be initialized by the driver
(which it actually is). Also, using patch_info.size would be actually
wrong for 16-bit data, as one field counts samples, while the other
counts bytes.
load_guspatch() already did it right.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-5-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both bounds had off-by-one errors.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-4-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The `client` parameter was not used, so eliminate it from the call
chain.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-3-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We forgot to remember the wavetable /proc entry, so we'd fail to free it
at module unload.
This matters only when only the synth module is unloaded, as unloading
the card driver would tear down the sub-entry anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-2-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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In cases where the sof driver is unable to find the firmware and/or
topology file [1], it exits without releasing the i915 runtime
pm wakeref [2]. This results in dmesg warnings[3] during
suspend/resume or driver unbind. Add remove_late() to the failure path
of sof_init_environment so that i915 wakeref is released appropriately
[1]
[ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found.
[ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles
[ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested):
[ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri
[ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg
[ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed.
[ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from:
[ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/
[ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2
[2]
ref_tracker: 0000:00:02.0@ffff9b8511b6a378 has 1/5 users at
track_intel_runtime_pm_wakeref.part.0+0x36/0x70 [i915]
__intel_runtime_pm_get+0x51/0xb0 [i915]
intel_runtime_pm_get+0x17/0x20 [i915]
intel_display_power_get+0x2f/0x70 [i915]
i915_audio_component_get_power+0x23/0x120 [i915]
snd_hdac_display_power+0x89/0x130 [snd_hda_core]
hda_codec_i915_init+0x3f/0x50 [snd_sof_intel_hda]
hda_dsp_probe_early+0x170/0x250 [snd_sof_intel_hda_common]
snd_sof_device_probe+0x224/0x320 [snd_sof]
sof_pci_probe+0x15b/0x220 [snd_sof_pci]
hda_pci_intel_probe+0x30/0x70 [snd_sof_intel_hda_common]
local_pci_probe+0x4c/0xb0
pci_device_probe+0xcc/0x250
really_probe+0x18e/0x420
__driver_probe_device+0x7e/0x170
driver_probe_device+0x23/0xa0
[3]
[ 484.105070] ------------[ cut here ]------------
[ 484.108238] thunderbolt 0000:00:0d.2: PM: pci_pm_suspend_late+0x0/0x50 returned 0 after 0 usecs
[ 484.117106] i915 0000:00:02.0: i915 raw-wakerefs=1 wakelocks=1 on cleanup
[ 484.792005] WARNING: CPU: 2 PID: 2405 at drivers/gpu/drm/i915/intel_runtime_pm.c:444 intel_runtime_pm_driver_release+0x6c/0x80
Tested-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Lucas De Marchi <lucas.demarchi@intel.com>
Link: https://github.com/thesofproject/linux/pull/4878
Signed-off-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Link: https://msgid.link/r/20240404184813.134566-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Before ACP firmware loading, DSP interrupts are not expected.
Sometimes after reboot, it's observed that before ACP firmware is loaded
false DSP interrupt is reported.
Registering the interrupt handler before acp initialization causing false
interrupts sometimes on reboot as ACP reset is not applied.
Correct the sequence by invoking acp initialization sequence prior to
registering interrupt handler.
Fixes: 738a2b5e2cc9 ("ASoC: SOF: amd: Add IPC support for ACP IP block")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240404041717.430545-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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During probe the DMIC/SSP offload is enabled and it is not reversed on
remove.
Add a remove wrapper for LNL to disable the offload for DMIC and SSP
similarly to what is done during probe.
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>
Link: https://msgid.link/r/20240403111839.27259-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Modpost warns about missing module description, add it.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240402130640.3310999-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We removed the configuration of ES8326_ADC_SCALE
in es8326_jack_detect_handler because user changed
the configuration by snd_controls
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-5-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We got a headphone detection issue after suspend and resume.
And we fixed it by modifying the configuration at es8326_suspend
and invoke es8326_irq at es8326_resume.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-4-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We got a digital microphone feature issue. And we fixed it by modifying
the clock table. Also, we changed the marco ES8326_CLK_ON declaration
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-3-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We got an error report about headphone type detection and button detection.
We fixed the headphone type detection error by adjusting the debounce timer
configuration. And we fixed the button detection error by disabling the
button detection feature when the headphone are unplugged and enabling it
when headphone are plugged in.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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For shutting up spurious KMSAN uninit-value warnings, just replace
kmalloc() calls with kzalloc() for the buffers used for
communications. There should be no real issue with the original code,
but it's still better to cover.
Reported-by: syzbot+7fb05ccf7b3d2f9617b3@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/r/00000000000084b18706150bcca5@google.com
Message-ID: <20240402063628.26609-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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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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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Stefan Binding
Peter Ujfalusi
Geoffrey D. Bennett
Oswald Buddenhagen
Shenghao Ding
Vitaly Rodionov
Chaitanya Kumar Borah
Vijendar Mukunda
Amadeusz Sławiński
Zhang Yi
Takashi Iwai
Luke D. Jones
I Gede Agastya Darma Laksana