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commit 269f399dc19f0e5c51711c3ba3bd06e0ef6ef403 upstream.
Otherwise bit clock remains running writing invalid data to the DAC.
Signed-off-by: Matus Gajdos <matuszpd@gmail.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20230712124934.32232-1-matuszpd@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0659400f18c0e6c0c69d74fe5d09e7f6fbbd52a2 upstream.
The HP Laptop 15s-eq2xxx uses ALC236 codec and controls the mute LED using
COEF 0x07 index 1. No existing quirk covers this configuration.
Adds a new quirk and enables it for the device.
Signed-off-by: Luka Guzenko <l.guzenko@web.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230718161241.393181-1-l.guzenko@web.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 69ea4c9d02b7947cdd612335a61cc1a02e544ccd upstream.
This was the ALC283 depop procedure.
Maybe this procedure wasn't suitable with new codec.
So, let us remove it. But HP 15z-fc000 must do 3k pull low. If it
reboot with plugged headset,
it will have errors show don't find codec error messages. Run 3k pull
low will solve issues.
So, let AMD chipset will run this for workarround.
Fixes: 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Reported-by: Joseph C. Sible <josephcsible@gmail.com>
Closes: https://lore.kernel.org/r/CABpewhE4REgn9RJZduuEU6Z_ijXNeQWnrxO1tg70Gkw=F8qNYg@mail.gmail.com/
Link: https://lore.kernel.org/r/4678992299664babac4403d9978e7ba7@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a46d37012a5be1737393b8f82fd35665e4556eee upstream.
If the second component fails to initialize, cleanup the first on.
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Cc: stable@kernel.org
Fixes: f1b5bf07365d ("ASoC: mt2701/mt8173: replace platform to component")
Signed-off-by: Ricardo Ribalda Delgado <ribalda@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230612-mt8173-fixup-v2-1-432aa99ce24d@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f9c058d14f4fe23ef523a7ff73734d51c151683c upstream.
After reordering the irq probe, the error path was not properly done.
Lets fix it.
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Cc: stable@kernel.org
Fixes: 4cbb264d4e91 ("ASoC: mediatek: mt8173: Enable IRQ when pdata is ready")
Signed-off-by: Ricardo Ribalda Delgado <ribalda@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230612-mt8173-fixup-v2-2-432aa99ce24d@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 89dbb335cb6a627a4067bc42caa09c8bc3326d40 ]
snd_jack_report() is supposed to be callable from an IRQ context, too,
and it's indeed used in that way from virtsnd driver. The fix for
input_dev race in commit 1b6a6fc5280e ("ALSA: jack: Access input_dev
under mutex"), however, introduced a mutex lock in snd_jack_report(),
and this resulted in a potential sleep-in-atomic.
For addressing that problem, this patch changes the relevant code to
use the object get/put and removes the mutex usage. That is,
snd_jack_report(), it takes input_get_device() and leaves with
input_put_device() for assuring the input_dev being assigned.
Although the whole mutex could be reduced, we keep it because it can
be still a protection for potential races between creation and
deletion.
Fixes: 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex")
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Closes: https://lore.kernel.org/r/cf95f7fe-a748-4990-8378-000491b40329@moroto.mountain
Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230706155357.3470-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 60413129ee2b38a80347489270af7f6e1c1de4d0 ]
When using the codec through the generic audio graph card, there are at
least two calls of es8316_set_dai_sysclk(), with the effect of limiting
the allowed sample rates according to the MCLK/LRCK ratios supported by
the codec:
1. During audio card setup, to set the initial MCLK - see
asoc_simple_init_dai().
2. Before opening a stream, to update MCLK, according to the stream
sample rate and the multiplication factor - see
asoc_simple_hw_params().
In some cases the initial MCLK might be set to a frequency that doesn't
match any of the supported ratios, e.g. 12287999 instead of 12288000,
which is only 1 Hz below the supported clock, as that is what the
hardware reports. This creates an empty list of rate constraints, which
is further passed to snd_pcm_hw_constraint_list() via
es8316_pcm_startup(), and causes the following error on the very first
access of the sound card:
$ speaker-test -D hw:Analog,0 -F S16_LE -c 2 -t wav
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument
Note that all subsequent retries succeed thanks to the updated MCLK set
at point 2 above, which uses a computed frequency value instead of a
reading from the hardware registers. Normally this would have mitigated
the issue, but es8316_pcm_startup() executes before the 2nd call to
es8316_set_dai_sysclk(), hence it cannot make use of the updated
constraints.
Since es8316_pcm_hw_params() performs anyway a final validation of MCLK
against the stream sample rate and the supported MCLK/LRCK ratios, fix
the issue by ensuring that sysclk_constraints list is only set when at
least one supported sample rate is autodetected by the codec.
Fixes: b8b88b70875a ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-3-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6f073429037cd79d7311cd8236311c53f5ea8f01 ]
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: b8b88b70875a ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ]
Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the
DMI table, so that active-high jack-detect will work properly on
these laptops.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ]
In the error path, a of_node_put() for platform is missing.
Just add it.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ]
In the BE hw_params configuration, the existing code checks if any of the
existing FEs are prepared, running, paused or suspended - and skips the
configuration in those cases. This allows multiple calls of hw_params
which the ALSA state machine supports.
This check is not handled for the prepare stage, which can lead to the
same BE being prepared multiple times. This patch adds a check similar to
that of the hw_params, with the main difference being that the suspended
state is allowed: the ALSA state machine allows a transition from
suspended to prepared with hw_params skipped.
This problem was detected on Intel IPC4/SoundWire devices, where the BE
dailink .prepare stage is used to configure the SoundWire stream with a
bank switch. Multiple .prepare calls lead to conflicts with the .trigger
operation with IPC4 configurations. This problem was not detected earlier
on Intel devices, HDaudio BE dailinks detect that the link is already
prepared and skip the configuration, and for IPC3 devices there is no BE
trigger.
Link: https://github.com/thesofproject/sof/issues/7596
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6e7a6d4797ef521c0762914610ed682e102b9d36 ]
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230523154605.4284-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 7ca4c8d4d3f41c2cd9b4cf22bb829bf03dac0956 upstream.
Headset microphone on this platform does not work without
ALC897_FIXUP_HEADSET_MIC_PIN fixup.
Signed-off-by: RenHai <kean0048@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230602003604.975892-1-kean0048@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 527c356b51f3ddee02c9ed5277538f85e30a2cdc upstream.
Add a quirk for HP Slim Desktop S01 to fixup headset MIC no presence.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230526094704.14597-1-aichao@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ]
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
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ]
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
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ]
Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for
Glenfly Gpus.
- In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware
limitation, need to increase hdac interrupt interval.
- In azx_first_init, enable polling mode for Glenfly Gpu. When the codec
complete the command, it sends interrupt and writes response entries to
memory, howerver, the write requests sometimes are not actually
synchronized to memory when driver handle hdac interrupt on Glenfly Gpus.
If the RIRB status is not updated in the interrupt handler,
azx_rirb_get_response keeps trying to recevie a response from rirb until
1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue.
- In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need
driver to do actual clean-ups for the linked codec when switch from one
codec to another.
Signed-off-by: jasontao <jasontao@glenfly.com>
Signed-off-by: Reaper Li <reaperlioc@glenfly.com>
Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4ca110cab46561cd74a2acd9b447435acb4bec5f upstream.
Lenovo M70/M90 Gen4 are equipped with ALC897, and they need
ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work.
The previous quirk for M70/M90 is for Gen3.
Signed-off-by: Bin Li <bin.li@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230524113755.1346928-1-bin.li@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 81302b1c7c997e8a56c1c2fc63a296ebeb0cd2d0 upstream.
It's reported that the recording started right after the driver probe
doesn't work properly, and it turned out that this is related with the
codec auto-suspend. Namely, after the probe phase, the usage count
goes zero, and the auto-suspend is programmed, but the codec is kept
still active until the auto-suspend expiration. When an application
(e.g. alsactl) updates the mixer values at this moment, the values are
cached but not actually written. Then, starting arecord thereafter
also results in the silence because of the missing unmute.
The root cause is the handling of "lazy update" mode; when a mixer
value is updated *after* the suspend, it should update only the cache
and exits. At the resume, the cached value is written to the device,
in turn. The problem is that the current code misinterprets the state
of auto-suspend as if it were already suspended.
Although we can add the check of the actual device state after
pm_runtime_get_if_in_use() for catching the missing state, this won't
suffice; the second call of regmap_update_bits_check() will skip
writing the register because the cache has been already updated by the
first call. So we'd need fixes in two different places.
OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with
pm_runtime_get_if_active() (with ign_usage_count=true). This change
implies that the driver takes the pm refcount if the device is still
in ACTIVE state and continues the processing. A small caveat is that
this will leave the auto-suspend timer. But, since the timer callback
itself checks the device state and aborts gracefully when it's active,
this won't be any substantial problem.
Long story short: we address the missing register-write problem just
by replacing the pm_runtime_*() call in snd_hda_keep_power_up().
Fixes: fc4f000bf8c0 ("ALSA: hda - Fix unexpected resume through regmap code path")
Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.com
Suggested-by: Cezary Rojewski <cezary.rojewski@intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230518113520.15213-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7843380d07bbeffd3ce6504e73cf61f840ae76ca upstream.
This quirk is necessary for surround and other DSP effects to work
with the onboard ca0132 based audio chipset for the EVGA X299 dark
mainboard.
Signed-off-by: Adam Stylinski <kungfujesus06@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=67071
Link: https://lore.kernel.org/r/ZGopOe19T1QOwizS@eggsbenedict.adamsnet
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a4671b7fba59775845ee60cfbdfc4ba64300211b upstream.
Add quirk for GU603 with 0x1c62 variant of codec.
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230505235824.49607-2-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 90670ef774a8b6700c38ce1222e6aa263be54d5f upstream.
Add a quirk for HP EliteDesk 805 to fixup ALC3867 headset MIC no sound.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230506022653.2074343-1-aichao@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit dc4f2ccaedddb489a83e7b12ebbdc347272aacc9 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3b44ec8c5c44790a82f07e90db45643c762878c6 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 359b4315471181f108723c61612d96e383e56179 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 92553ee03166ef8fa978e7683f9f4af30c9c4e6b ]
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: 563785edfcef ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c0e72058d5e21982e61a29de6b098f7c1f0db498 ]
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: 9a08067ec318 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0d727e1856ef22dd9337199430258cb64cbbc658 ]
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: 523f1dce3743 ("[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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 357ad4d898286b94aaae0cb7e3f573459e5b98b9 upstream.
We observed: 'dmasound_setup' defined but not used error with
COMPILER=gcc ARCH=m68k DEFCONFIG=allmodconfig build.
Fix it by adding __maybe_unused to dmasound_setup.
Error(s):
sound/oss/dmasound/dmasound_core.c:1431:12: error: 'dmasound_setup' defined but not used [-Werror=unused-function]
Fixes: 9dd7c46346ca ("sound/oss/dmasound: fix build when drivers are mixed =y/=m")
Signed-off-by: Miles Chen <miles.chen@mediatek.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20220414091940.2216-1-miles.chen@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 1c34890273a020d61d6127ade3f68ed1cb21c16a ]
of_node_put() should have been done directly after
mqs_priv->regmap = syscon_node_to_regmap(gpr_np);
otherwise it creates a reference leak on the success path.
To fix this, of_node_put() is moved to the correct location, and change
all the gotos to direct returns.
Fixes: a9d273671440 ("ASoC: fsl_mqs: Fix error handling in probe")
Signed-off-by: Liliang Ye <yll@hust.edu.cn>
Reviewed-by: Dan Carpenter <error27@gmail.com>
Link: https://lore.kernel.org/r/20230403152647.17638-1-yll@hust.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 39db65a0a17b54915b269d3685f253a4731f344c ]
The driver is able to work fine without relying on a mandatory interrupt
being assigned to the I2C device. This is only needed when making use of
the jack-detect support.
However, the following warning message is always emitted when there is
no such interrupt available:
es8316 0-0011: Failed to get IRQ 0: -22
Do not attempt to request an IRQ if it is not available/valid. This also
ensures the rather misleading message is not displayed anymore.
Also note the IRQ validation relies on commit dab472eb931bc291 ("i2c /
ACPI: Use 0 to indicate that device does not have interrupt assigned").
Fixes: 822257661031 ("ASoC: es8316: Add jack-detect support")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20230328094901.50763-1-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1cf2aa665901054b140eb71748661ceae99b6b5a ]
Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then
disabling it immediately after requesting it.
This fixes a possible race where the IRQ might trigger between requesting
and disabling it; and this also leads to a small code cleanup.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211003132255.31743-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 39db65a0a17b ("ASoC: es8316: Handle optional IRQ assignment")
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 9dd7c46346ca4390f84a7ea9933005eb1b175c15 upstream.
When CONFIG_DMASOUND_ATARI=m and CONFIG_DMASOUND_Q40=y (or vice versa),
dmasound_core.o can be built without dmasound_deinit() being defined,
causing a build error:
ERROR: modpost: "dmasound_deinit" [sound/oss/dmasound/dmasound_atari.ko] undefined!
Modify dmasound_core.c and dmasound.h so that dmasound_deinit() is
always available.
The mixed modes (=y/=m) also mean that several variables and structs
have to be declared in all cases.
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Suggested-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Reported-by: kernel test robot <lkp@intel.com>
Link: lore.kernel.org/r/202204032138.EFT9qGEd-lkp@intel.com
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20220405234118.24830-1-rdunlap@infradead.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit e38c5e80c3d293a883c6f1d553f2146ec0bda35e ]
The Acer Iconia One 7 B1-750 tablet mostly works fine with the defaults
for an Bay Trail CR tablet. Except for the internal mic, instead of
an analog mic on IN3 a digital mic on DMIC1 is uses.
Add a quirk with these settings for this tablet.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20230322145332.131525-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 86a24e99c97234f87d9f70b528a691150e145197 upstream.
dma_request_slave_channel() may return NULL which will lead to
NULL pointer dereference error in 'tmp_chan->private'.
Correct this behaviour by, first, switching from deprecated function
dma_request_slave_channel() to dma_request_chan(). Secondly, enable
sanity check for the resuling value of dma_request_chan().
Also, fix description that follows the enacted changes and that
concerns the use of dma_request_slave_channel().
Fixes: 706e2c881158 ("ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End")
Co-developed-by: Natalia Petrova <n.petrova@fintech.ru>
Signed-off-by: Nikita Zhandarovich <n.zhandarovich@fintech.ru>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://lore.kernel.org/r/20230417133242.53339-1-n.zhandarovich@fintech.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f342ac00da1064eb4f94b1f4bcacbdfea955797a upstream.
The BIOS botches this one completely - it says the 2nd S/PDIF output is
used, while in fact it's the 1st one. This is tested on DP45SG, but I'm
assuming it's valid for the other boards in the series as well.
Also add some comments regarding the pins.
FWIW, the codec is apparently still sold by Tempo Semiconductor, Inc.,
where one can download the documentation.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197826-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fb4a624f88f658c7b7ae124452bd42eaa8ac7168 upstream.
Smatch Warns:
sound/firewire/tascam/tascam-stream.c:493 snd_tscm_stream_start_duplex()
warn: missing unwind goto?
The direct return will cause the stream list of "&tscm->domain" unemptied
and the session in "tscm" unfinished if amdtp_domain_start() returns with
an error.
Fix this by changing the direct return to a goto which will empty the
stream list of "&tscm->domain" and finish the session in "tscm".
The snd_tscm_stream_start_duplex() function is called in the prepare
callback of PCM. According to "ALSA Kernel API Documentation", the prepare
callback of PCM will be called many times at each setup. So, if the
"&d->streams" list is not emptied, when the prepare callback is called
next time, snd_tscm_stream_start_duplex() will receive -EBUSY from
amdtp_domain_add_stream() that tries to add an existing stream to the
domain. The error handling code after the "error" label will be executed
in this case, and the "&d->streams" list will be emptied. So not emptying
the "&d->streams" list will not cause an issue. But it is more efficient
and readable to empty it on the first error by changing the direct return
to a goto statement.
The session in "tscm" has been begun before amdtp_domain_start(), so it
needs to be finished when amdtp_domain_start() fails.
Fixes: c281d46a51e3 ("ALSA: firewire-tascam: support AMDTP domain")
Signed-off-by: Xu Biang <xubiang@hust.edu.cn>
Reviewed-by: Dan Carpenter <error27@gmail.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230406132801.105108-1-xubiang@hust.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e98e7a82bca2b6dce3e03719cff800ec913f9af7 upstream.
snd_cs8427_iec958_active() would always delete
SNDRV_CTL_ELEM_ACCESS_INACTIVE, even though the function has an
argument `active`.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201219.2197811-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c17f8fd31700392b1bb9e7b66924333568cb3700 upstream.
Like the other boards from the D*45* series, this one sets up the
outputs not quite correctly.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197826-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b09c551c77c7e01dc6e4f3c8bf06b5ffa7b06db5 upstream.
Due to two copy/pastos, closing the MIC or EFX capture device would
make a running ADC capture hang due to unsetting its interrupt handler.
In principle, this would have also allowed dereferencing dangling
pointers, but we're actually rather thorough at disabling and flushing
the ints.
While it may sound like one, this actually wasn't a hypothetical bug:
PortAudio will open a capture stream at startup (and close it right
away) even if not asked to. If the first device is busy, it will just
proceed with the next one ... thus killing a concurrent capture.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197923-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fa4e7a6fa12b1132340785e14bd439cbe95b7a5a upstream.
It's been reported that the recent kernel can't probe the PCM devices
on Roland VS-100 properly, and it turned out to be a regression by the
recent addition of the bit shift range check for the format bits.
In the old code, we just did bit-shift and it resulted in zero, which
is then corrected to the standard PCM format, while the new code
explicitly returns an error in such a case.
For addressing the regression, relax the check and fallback to the
standard PCM type (with the info output).
Fixes: 43d5ca88dfcd ("ALSA: usb-audio: Fix potential out-of-bounds shift")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217084
Link: https://lore.kernel.org/r/20230324075005.19403-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
