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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
Calibrated data was written into an incorrect register, which cause
speaker protection sometimes malfuctions
Fixes: 5be27f1e3e ("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>
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>
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
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Merge tag 'asoc-fix-v6.9-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
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: 738a2b5e2c ("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>
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>
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We solved some issues related to headphone detection.And for using
the same configuration in different power conditions,we modified the
clock table
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>
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>
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>
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>
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>
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: 0fca97a29b ("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>
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>
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: df335e9a8b ("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>
As described in the added code comment, a reference to .exit.text is ok
for drivers registered via module_platform_driver_probe(). Make this
explicit to prevent the following section mismatch warning
WARNING: modpost: sound/oss/dmasound/dmasound_paula: section mismatch in reference: amiga_audio_driver+0x8 (section: .data) -> amiga_audio_remove (section: .exit.text)
that triggers on an allmodconfig W=1 build.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Message-ID: <c216a129aa88f3af5c56fe6612a472f7a882f048.1711748999.git.u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
If acp_init() fails, acp pci driver probe should return error.
Add acp_init() function return value check logic.
Fixes: e61b415515 ("ASoC: amd: acp: refactor the acp init and de-init sequence")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20240329053815.2373979-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since we put dvc_tlv static variable to a header file it's copied to
each module that includes the header. But not all of them are actually
used it.
Fix this W=1 build warning:
include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not
used [-Wunused-const-variable=]
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/
Fixes: ae065d0ce9 ("ALSA: hda/tas2781: remove digital gain kcontrol")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <0e461545a2a6e9b6152985143e50526322e5f76b.1711665731.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Fix a set of problematic locking sequences and update error messages,
tested on SOF/SoundWire platforms.
In snd_soc_info_volsw(), mask is generated by figuring out the index of
the most significant bit set in max and converting the index to a
bitmask through bit shift 1. Unintended wraparound occurs when max is an
integer value with msb bit set. Since the bit shift value 1 is treated
as an integer type, the left shift operation will wraparound and set
mask to 0 instead of all 1's. In order to fix this, we type cast 1 as
`1ULL` to prevent the wraparound.
Fixes: 7077148fb5 ("ASoC: core: Split ops out of soc-core.c")
Signed-off-by: Stephen Lee <slee08177@gmail.com>
Link: https://msgid.link/r/20240326010131.6211-1-slee08177@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
