The following error occurs when trying to restore a previously saved
ALSA mixer state (tested on a Rock 5B board):
$ alsactl --no-ucm -f /tmp/asound.state store hw:Analog
$ alsactl --no-ucm -I -f /tmp/asound.state restore hw:Analog
alsactl: set_control:1475: Cannot write control '2:0:0:ALC Capture Target Volume:0' : Invalid argument
According to ES8316 datasheet, the register at address 0x2B, which is
related to the above mixer control, contains by default the value 0xB0.
Considering the corresponding ALC target bits (ALCLVL) are 7:4, the
control is initialized with 11, which is one step above the maximum
value allowed by the driver:
ALCLVL | dB gain
-------+--------
0000 | -16.5
0001 | -15.0
0010 | -13.5
.... | .....
0111 | -6.0
1000 | -4.5
1001 | -3.0
1010 | -1.5
.... | .....
1111 | -1.5
The tests performed using the VU meter feature (--vumeter=TYPE) of
arecord/aplay confirm the specs are correct and there is no measured
gain if the 1011-1111 range would have been mapped to 0 dB:
dB gain | VU meter %
--------+-----------
-6.0 | 30-31
-4.5 | 35-36
-3.0 | 42-43
-1.5 | 50-51
0.0 | 50-51
Increment the max value allowed for ALC Capture Target Volume control,
so that it matches the hardware default. Additionally, update the
related TLV to prevent an artificial extension of the dB gain range.
Fixes: b8b88b7087 ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These IDs are for AD102, AD103, AD104, AD106, and AD107 gpus with
audio functions that are largely similar to the existing ones.
Tested audio using gnome-settings, over HDMI, DP-SST and DP-MST
connections on AD106 gpu.
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230517090736.15088-1-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two functions are defined and used in pcm_oss.c but also optionally
used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS
is disabled, this causes a warning as the functions are not static
and have no prototype:
sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes]
sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes]
Avoid this by making the prototypes unconditional.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_cs46xx_download_image() was originally called from dsp_spos.c, but
is now local to cs46xx_lib.c. Mark it as 'static' to avoid a warning
about it lacking a declaration, and '__maybe_unused' to avoid a warning
about it being unused when CONFIG_SND_CS46XX_NEW_DSP is disabled:
sound/pci/cs46xx/cs46xx_lib.c:534:5: error: no previous prototype for 'snd_cs46xx_download_image'
Fixes: 89f157d9e6 ("[ALSA] cs46xx - Fix PM resume")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230516195046.550584-1-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
get_line_out_pfx() may trigger an Oops by overflowing the static array
with more than 8 channels. This was reported for MacBookPro 12,1 with
Cirrus codec.
As a workaround, extend for the 9.1 channels and also fix the
potential Oops by unifying the code paths accessing the same array
with the proper size check.
Reported-by: Olliver Schinagl <oliver@schinagl.nl>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/64d95eb0-dbdb-cff8-a8b1-988dc22b24cd@schinagl.nl
Link: https://lore.kernel.org/r/20230516184412.24078-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More fixes that came in since the merge window, the bulk of which are
for the SOF code, I suspect as a result of the wide usage, active
development and large code size rather than huge quality problems.
There's also a couple of MAINTAINERS updates and some new device quirks.
-----BEGIN PGP SIGNATURE-----
iQEzBAABCgAdFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAmRjoe4ACgkQJNaLcl1U
h9BNiwf/SRv0HUMuaLHV/a7TUoLyFZbYIR7OoYNWZcGcNb01C12sEcjaFm38p7wD
sjzAAECQyfLgulLIaqc68lcOzF5+QfM28Gqfhw9mKVf9lLQaOvgzJfZ0gZG/CIRk
8MpqsFYF+XRsvmJy8xxS8PMaOCiBMJd31EFHt5dYr/rXJMmJnK7jwpGW5Gsf1o/N
naISPkeDIbjsmSalTyBXOfgebS/tWvcW4/Gsx6Rs37kH8e1xP72gOFjRy5bQXiVk
3o0eHVK7kw7oUacBUjAgxn1mWkeyhSFl235Nh6R89L6XDNeYr254SRFd3Y9kxeDq
Wx/H3RLjNhr8yA12XGJhrQ4/bK6Ktw==
=dYo0
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v6.4-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
More fixes that came in since the merge window, the bulk of which are
for the SOF code, I suspect as a result of the wide usage, active
development and large code size rather than huge quality problems.
There's also a couple of MAINTAINERS updates and some new device quirks.
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
With additional testing with multiple links and multiple DAI types, we
found a couple of mistakes with refcounts, base address, missing
initialization.
A new helper was also added due to a change in the SoundWire
programming sequences, with the host driver in charge of setting up
the DMA channel mapping instead of the firmware.
The memory allocated for the tuples array assumes that there's 1
instance of all tokens already. So for those tokens that have multiple
instances in topology, we need to exclude the initial instance that has
already been accounted for.
Fixes: 4fdef47a44 ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230515085200.17094-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Using the same token ID for both input and output format pin index
results in collisions and incorrect pin index getting parsed from
topology.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@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: Péter Ujfalusi <peter.ujfalusi@linux.intel.com
Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230515104403.32207-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Flush the SoundWire interrupt handler work instead of cancelling it.
When a SoundWire interrupt is triggered the pm_runtime is held
until the work has completed. It's therefore unsafe to cancel
the work, it must be flushed.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com
Link: https://lore.kernel.org/r/20230512144237.739000-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org
Topology could have more instances of the tokens being searched for than
the number of sets that need to be copied. Stop copying token after the
limit of number of token instances has been reached. This worked before
only by chance as we had allocated more size for the tuples array than
the number of actual tokens being parsed.
Fixes: 7006d20e5e ("ASoC: SOF: Introduce IPC3 ops")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512114630.24439-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
If there are failures in DSP runtime resume, the device state will not
reach active and this makes it impossible e.g. to retrieve a possible
DSP panic dump via "exception" debugfs node. If
CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in
cache is stale. If debugfs cache is not used, the region simply cannot
be read.
To allow debugging these scenarios, update the debugfs cache contents in
resume error handler. User-space can then later retrieve DSP panic and
other state via debugfs (requires SOF debugfs cache to be enabled in
build).
Reported-by: Curtis Malainey <cujomalainey@chromium.org
Link: https://github.com/thesofproject/linux/issues/4274
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
When devm runs function in the "remove" path for a device it runs them
in the reverse order. That means that if you have parts of your driver
that aren't using devm or are using "roll your own" devm w/
devm_add_action_or_reset() you need to keep that in mind.
The mt8186 audio driver didn't quite get this right. Specifically, in
mt8186_init_clock() it called mt8186_audsys_clk_register() and then
went on to call a bunch of other devm function. The caller of
mt8186_init_clock() used devm_add_action_or_reset() to call
mt8186_deinit_clock() but, because of the intervening devm functions,
the order was wrong.
Specifically at probe time, the order was:
1. mt8186_audsys_clk_register()
2. afe_priv->clk = devm_kcalloc(...)
3. afe_priv->clk[i] = devm_clk_get(...)
At remove time, the order (which should have been 3, 2, 1) was:
1. mt8186_audsys_clk_unregister()
3. Free all of afe_priv->clk[i]
2. Free afe_priv->clk
The above seemed to be causing a use-after-free. Luckily, it's easy to
fix this by simply using devm more correctly. Let's move the
devm_add_action_or_reset() to the right place. In addition to fixing
the use-after-free, code inspection shows that this fixes a leak
(missing call to mt8186_audsys_clk_unregister()) that would have
happened if any of the syscon_regmap_lookup_by_phandle() calls in
mt8186_init_clock() had failed.
Fixes: 55b423d562 ("ASoC: mediatek: mt8186: support audio clock control in platform driver")
Signed-off-by: Douglas Anderson <dianders@chromium.org
Link: https://lore.kernel.org/r/20230511092437.1.I31cceffc8c45bb1af16eb613e197b3df92cdc19e@changeid
Signed-off-by: Mark Brown <broonie@kernel.org
The commands in sof_ipc_dai_config.flags are encoded as bits:
1 (bit0) - hw_params
2 (bit1) - hw_free
4 (bit2) - pause
These are commands, they cannot be combined as one would assume, for
example
3 (bit0 | bit1) is invalid.
This can happen right at the second start of a stream as at the end of the
first stream we set the hw_free command (bit1) and on the second start we
would OR on top of it the hw_params (bit0).
Fixes: b66bfc3a98 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230512110317.5180-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
When an error occurs, we need to make sure the device can pm_runtime
suspend instead of keeping it active.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
When an error occurs, we need to make sure the device can pm_runtime
suspend instead of keeping it active.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
When a firmware IPC error happens during a pm_runtime suspend, we
ignore the error and suspend anyways. However, the code
unconditionally increases the runtime_pm counter. This results in a
confusing configuration where the code will suspend, resume but never
suspend again due to the use of pm_runtime_get_noresume().
The intent of the counter increase was to prevent entry in D3, but if
that transition to D3 is already started it cannot be stopped. In
addition, there's no point in that case in trying to prevent anything,
the firmware error is handled and the next resume will re-initialize
the firmware completely.
This patch changes the logic to prevent suspend when the device is
pm_runtime active and has a use_count > 0.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
These registers enable the HDaudio DMA hardware to split/merge data
from different PDIs, possibly on different links.
This capability exists for all types of HDaudio extended links, but
for now is only required for SoundWire. In the SSP/DMIC case, the IP
is programmed by the DSP firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
We defined the values but never initialized it for SoundWire/SSP, fix
this miss.
A Fixes: tag is not provided as instance_offset was not used so far,
so nothing was really broken. This patch is only required for the
SoundWire support in the following patch.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
We mix the use of hlink->ml_addr and the 'ml_addr' parameter. It's the
same thing, let's align on using the parameter.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
The base_ptr value needs to be derived from the remap_addr pointer,
not the ml_addr. This base_ptr was used only in debug logs that were
so far not contributed upstream so the issue was not detected. It
needs to be fixed for SoundWire support on LunarLake.
Fixes: 17c9b6ec35 ("ASoC: SOF: Intel: hda-mlink: add structures to parse ALT links")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Same functionality as for DMIC/SSP with different ID.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
In hindsight it was a very bad idea to use the same refcount for
Extended and 'legacy' HDaudio multi-links. The existing solution only
powers-up the first sublink, which causes SoundWire and SSP tests to
fail when more than one DAI is used concurrently. Solving this problem
requires per-sublink refcounting, as suggested in this patch.
The existing refcounting remains for 'legacy' HdAudio links, mainly to
avoid changing the obscure programming sequence in
snd_hdac_ext_bus_link_put().
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
There's yet another laptop that needs the fixup to enable mute and
micmute LEDs. So do it accordingly.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230512083417.157127-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Pavilion 15 line has B&O top speakers similar to the x360 and
applying the same profile produces good sound. Without this, the
sound would be tinny and underpowered without either applying
model=alc295-hp-x360 or booting another OS first.
Signed-off-by: Ryan Underwood <nemesis@icequake.net>
Fixes: 563785edfc ("ALSA: hda/realtek - Add quirk entry for HP Pavilion 15")
Link: https://lore.kernel.org/r/ZF0mpcMz3ezP9KQw@icequake.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed
48k sample rate like other Line6 models. This patch adds the
corresponding entry to line6_parse_audio_format_rate_quirk().
Reported-by: John Humlick <john@humlick.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230512075858.22813-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code was supposed to return an error code if init_stream()
failed, but it instead freed dg00x->rx_stream and returned success.
This potentially leads to a use after free.
Fixes: 9a08067ec3 ("ALSA: firewire-digi00x: support AMDTP domain")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://lore.kernel.org/r/c224cbd5-d9e2-4cd4-9bcf-2138eb1d35c6@kili.mountain
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply a workaround for what appears to be a hardware quirk.
The problem seems to happen when enabling "whole chip power" (bit D7
register R6) for the very first time after the chip receives power. If
either "output" (D4) or "DAC" (D3) aren't powered on at that time,
playback becomes very distorted later on.
This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10:
https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset-issue/480229
I suspect this happens only when using an external MCLK signal (which
is the case for both of these boards).
Here are some experiments run on a Google Chameleon v3. These were run
in userspace using a wrapper around the i2cset utility:
ssmset() {
i2cset -y 0 0x1a $(($1*2)) $2
}
For each of the following sequences, we apply power to the ssm2603
chip, set the configuration registers R0-R5 and R7-R8, run the selected
sequence, and check for distortions on playback.
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x87 # out, dac
ssmset 0x06 0x07 # chip
OK
(disable MCLK)
ssmset 0x09 0x01 # core
ssmset 0x06 0x1f # chip
ssmset 0x06 0x07 # out, dac
(enable MCLK)
OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x1f # chip
ssmset 0x06 0x07 # out, dac
NOT OK
ssmset 0x06 0x1f # chip
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # out, dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x0f # chip, out
ssmset 0x06 0x07 # dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x17 # chip, dac
ssmset 0x06 0x07 # out
NOT OK
For each of the following sequences, we apply power to the ssm2603
chip, run the selected sequence, issue a reset with R15, configure
R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check
for distortions.
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
OK
(disable MCLK)
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
(enable MCLK after reset)
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x17 # chip, dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x0f # chip, out
NOT OK
ssmset 0x06 0x07 # chip, out, dac
NOT OK
Signed-off-by: Paweł Anikiel <pan@semihalf.com
Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org
These models use 2 CS35L41 amplifiers using SPI for down-facing
speakers.
alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the
down-facing speakers.
Pin configs are needed to enable headset mic detection.
Note that these models lack the ACPI _DSD properties needed to
initialize the amplifiers. They can be added during boot to get working
sound out of the speakers:
https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff
Signed-off-by: Alexandru Sorodoc <ealex95@gmail.com>
Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for HP EliteBook 835/845/845W/865 G10 laptops
with CS35L41 amplifiers on I2C/SPI bus connected to Realtek codec.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230510142227.32945-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the CPU supplies bit/frame clocks, the system clock (clk_i2s)
is divided to produce the bit clock. This is a simple 1/N divider
with a fairly limited range, so for a given system clock frequency
only a few sample rates can be produced. Usually a wider range of
sample rates is supported by varying the system clock frequency.
The old calculation method was not very robust and could easily
produce the wrong clock rate, especially with non-standard rates.
For example, if the system clock is 1.99x the target bit clock
rate, the divider would be calculated as 1 instead of the more
accurate 2.
Instead, use a more accurate method that considers two adjacent
divider settings and selects the one that produces the least error
versus the requested rate. If the error is 5% or higher then the
rate setting is rejected to prevent garbled audio.
Skip divider calculation when the codec is supplying both the bit
and frame clock; in that case, the divider outputs are unused and
we don't want to constrain the sample rate.
Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com
Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org
There is error message when defer probe happens:
fsl-micfil-dai 30ca0000.micfil: Unbalanced pm_runtime_enable!
Fix the error handler with pm_runtime_enable and add
fsl_micfil_remove() for pm_runtime_disable.
Fixes: 47a70e6fc9 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com
Link: https://lore.kernel.org/r/1683540996-6136-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org
On slow CPU (FPGA/QEMU emulated) printing overrun messages from
interrupt handler to uart console may leads to more overrun errors.
So use dev_err_ratelimited to limit the number of error messages.
Signed-off-by: Maxim Kochetkov <fido_max@inbox.ru
Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru
Signed-off-by: Mark Brown <broonie@kernel.org
Here are collections of small fixes for rc1.
The only (LOC-wise) dominant change was ASoC Qualcomm fix, but most
of it was merely a code shuffling.
Another significant change here is for ALSA PCM core; it received a
revert and a series of fixes for PCM auto-silencing where it caused
a regression in the previous PR for rc1.
Others are all small: ASoC Intel fixes, various quirks for ASoC AMD,
HD-audio and USB-audio, the continued legacy emu10k1 code cleanup,
and some documentation updates.
-----BEGIN PGP SIGNATURE-----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=jd8l
-----END PGP SIGNATURE-----
Merge tag 'sound-fix-6.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes for rc1.
The only (LOC-wise) dominant change was ASoC Qualcomm fix, but most of
it was merely a code shuffling.
Another significant change here is for ALSA PCM core; it received a
revert and a series of fixes for PCM auto-silencing where it caused a
regression in the previous PR for rc1.
Others are all small: ASoC Intel fixes, various quirks for ASoC AMD,
HD-audio and USB-audio, the continued legacy emu10k1 code cleanup, and
some documentation updates"
* tag 'sound-fix-6.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
ALSA: pcm: use exit controlled loop in snd_pcm_playback_silence()
ALSA: pcm: simplify top-up mode init in snd_pcm_playback_silence()
ALSA: pcm: playback silence - move silence variable updates to separate function
ALSA: pcm: playback silence - remove extra code
ALSA: pcm: fix playback silence - correct incremental silencing
ALSA: pcm: fix playback silence - use the actual new_hw_ptr for the threshold mode
ALSA: pcm: Revert "ALSA: pcm: rewrite snd_pcm_playback_silence()"
ALSA: hda/realtek: Fix mute and micmute LEDs for an HP laptop
ALSA: caiaq: input: Add error handling for unsupported input methods in `snd_usb_caiaq_input_init`
ALSA: usb-audio: Add quirk for Pioneer DDJ-800
ALSA: hda/realtek: support HP Pavilion Aero 13-be0xxx Mute LED
ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet
ASoC: amd: yc: Add Asus VivoBook Pro 14 OLED M6400RC to the quirks list for acp6x
ASoC: codecs: wcd938x: fix accessing regmap on unattached devices
ALSA: docs: Fix code block indentation in ALSA driver example
ALSA: docs: Extend module parameters description
ALSA: hda/realtek: Add quirk for ASUS UM3402YAR using CS35L41
ALSA: emu10k1: use more existing defines instead of open-coded numbers
ASoC: amd: yc: Add ASUS M3402RA into DMI table
ALSA: hda/realtek: Add quirk for ThinkPad P1 Gen 6
...
We already know that `frames` is greater than zero, because we just
checked it. So we don't need to check the loop condition on the first
iteration.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230505155244.2312199-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Inline the remaining call of snd_pcm_playback_hw_avail(). This makes
the top-up branch more congruent with the thresholded one, and allows
simplifying the handling of the corner cases.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230505155244.2312199-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code tracking the added samples in thresholded mode and the code
tracking the just played samples in top-up mode are semantically
identical, so factor it out to a common function to enhance readability.
Co-developed-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230505155244.2312199-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The removed condition handles de facto only one situation where
runtime->silence_filled variable is equal to runtime->buffer_size,
because this variable cannot go over the buffer size. This case is
implicitly caught by the required comparison of the noise distance
with the threshold.
Suggested-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230505155244.2312199-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a826ddba6 ("[ALSA] pcm core: fix silence_start calculations")
came with exactly the right commit message, but the patch just made
things broken in a different way: We'd fill at a too low address if the
area was already partially zeroed, so we'd under-fill. This affected
both thresholded mode (where it was somewhat less likely) and top-up
mode (where it would be the case consistently).
Co-developed-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230505155244.2312199-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_pcm_playback_hw_avail() function uses runtime->status->hw_ptr.
Unfortunately, in case when we call this function from snd_pcm_update_hw_ptr0(),
this variable contains the previous hardware pointer. Use the new_hw_ptr
argument to calculate hw_avail (filled samples by the user space) to
correct the threshold comparison.
The new_hw_ptr argument may also be set to ULONG_MAX which means the
initialization phase. In this case, use runtime->status->hw_ptr.
Suggested-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230505155244.2312199-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's another laptop that needs the fixup to enable mute and micmute
LEDs. So do it accordingly.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230505125925.543601-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A small set of fixes and device quirks that have come in during the
merge window, the Qualcomm fix seems quite large but it's mainly code
motion so looks larger than it is.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAmRU+zUTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0IPPB/457WnFOGXv/3FSOfTU4KVznNPOS9NM
7MfpsaI+ZtwhmHUZq/8Vd9+qcrKp4W6oJQa8z95oU+H3UQHmO8S9I4Ae+rqfzKSo
t7Gu/NTWZ89Vhib9w+ACXclFD0wwz5WxRuvZKQhW5iNMpY3S/ha7lBh9KK5Csb6+
JFiCsZcw4uWSdrH7hizZUQwZSZVyk3yoUDVDGOwUhCZ5ZS9N7/I5LQbI77Q1vadS
9H1xtKt6UA8Dq2a9yF7LEvNBb16TfJjGv46BviUdqs/fLGMCROU1wBYDoomAnS1I
iW0OGUwWGie/BAaDj3H1cwH9oEpUBm9wrjuRGf6MSuG/lrJeZOQ4zqIs
=Opld
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v6.4-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
A small set of fixes and device quirks that have come in during the
merge window, the Qualcomm fix seems quite large but it's mainly code
motion so looks larger than it is.
Smatch complains that:
snd_usb_caiaq_input_init() warn: missing error code 'ret'
This patch adds a new case to handle the situation where the
device does not support any input methods in the
`snd_usb_caiaq_input_init` function. It returns an `-EINVAL` error code
to indicate that no input methods are supported on the device.
Fixes: 523f1dce37 ("[ALSA] Add Native Instrument usb audio device support")
Signed-off-by: Ruliang Lin <u202112092@hust.edu.cn>
Reviewed-by: Dongliang Mu <dzm91@hust.edu.cn>
Acked-by: Daniel Mack <daniel@zonque.org>
Link: https://lore.kernel.org/r/20230504065054.3309-1-u202112092@hust.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
