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[ Upstream commit bd55842ed9 ]
The PCM memory allocation helpers have a sanity check against too many
buffer allocations. However, the check is performed without a proper
lock and the allocation isn't serialized; this allows user to allocate
more memories than predefined max size.
Practically seen, this isn't really a big problem, as it's more or
less some "soft limit" as a sanity check, and it's not possible to
allocate unlimitedly. But it's still better to address this for more
consistent behavior.
The patch covers the size check in do_alloc_pages() with the
card->memory_mutex, and increases the allocated size there for
preventing the further overflow. When the actual allocation fails,
the size is decreased accordingly.
Reported-by: BassCheck <bass@buaa.edu.cn>
Reported-by: Tuo Li <islituo@gmail.com>
Link: https://lore.kernel.org/r/CADm8Tek6t0WedK+3Y6rbE5YEt19tML8BUL45N2ji4ZAz1KcN_A@mail.gmail.com
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230703112430.30634-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 46cdff2369 ]
Set spec->en_3kpull_low default to true.
Then fillback ALC236 and ALC257 to false.
Additional note: this addresses a regression caused by the previous
fix 69ea4c9d02 ("ALSA: hda/realtek - remove 3k pull low procedure").
The previous workaround was applied too widely without necessity,
which resulted in the pop noise at PM again. This patch corrects the
condition and restores the old behavior for the devices that don't
suffer from the original problem.
Fixes: 69ea4c9d02 ("ALSA: hda/realtek - remove 3k pull low procedure")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217732
Link: https://lore.kernel.org/r/01e212a538fc407ca6edd10b81ff7b05@realtek.com
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1f848f121 ]
When the tdm lane mask is computed, the driver currently fills the 1st lane
before moving on to the next. If the stream has less channels than the
lanes can accommodate, slots will be disabled on the last lanes.
Unfortunately, the HW distribute channels in a different way. It distribute
channels in pair on each lanes before moving on the next slots.
This difference leads to problems if a device has an interface with more
than 1 lane and with more than 2 slots per lane.
For example: a playback interface with 2 lanes and 4 slots each (total 8
slots - zero based numbering)
- Playing a 8ch stream:
- All slots activated by the driver
- channel #2 will be played on lane #1 - slot #0 following HW placement
- Playing a 4ch stream:
- Lane #1 disabled by the driver
- channel #2 will be played on lane #0 - slot #2
This behaviour is obviously not desirable.
Change the way slots are activated on the TDM lanes to follow what the HW
does and make sure each channel always get mapped to the same slot/lane.
Fixes: 1a11d88f49 ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20230809171931.1244502-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1f4a08fed4 ]
The variable codec->regmap is often protected by the lock
codec->regmap_lock when is accessed. However, it is accessed without
holding the lock when is accessed in snd_hdac_regmap_sync():
if (codec->regmap)
In my opinion, this may be a harmful race, because if codec->regmap is
set to NULL right after the condition is checked, a null-pointer
dereference can occur in the called function regcache_sync():
map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c
To fix this possible null-pointer dereference caused by data race, the
mutex_lock coverage is extended to protect the if statement as well as the
function call to regcache_sync().
[ Note: the lack of the regmap_lock itself is harmless for the current
codec driver implementations, as snd_hdac_regmap_sync() is only for
PM runtime resume that is prohibited during the codec probe.
But the change makes the whole code more consistent, so it's merged
as is -- tiwai ]
Reported-by: BassCheck <bass@buaa.edu.cn>
Signed-off-by: Tuo Li <islituo@gmail.com>
Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e51df4f81b upstream.
In commit 2cb1e0259f ("ASoC: cs42l51: re-hook of_match_table
pointer"), 9 years ago, some random guy fixed the cs42l51 after it was
split into a core part and an I2C part to properly match based on a
Device Tree compatible string.
However, the fix in this commit is wrong: the MODULE_DEVICE_TABLE(of,
....) is in the core part of the driver, not the I2C part. Therefore,
automatic module loading based on module.alias, based on matching with
the DT compatible string, loads the core part of the driver, but not
the I2C part. And threfore, the i2c_driver is not registered, and the
codec is not known to the system, nor matched with a DT node with the
corresponding compatible string.
In order to fix that, we move the MODULE_DEVICE_TABLE(of, ...) into
the I2C part of the driver. The cs42l51_of_match[] array is also moved
as well, as it is not possible to have this definition in one file,
and the MODULE_DEVICE_TABLE(of, ...) invocation in another file, due
to how MODULE_DEVICE_TABLE works.
Thanks to this commit, the I2C part of the driver now properly
autoloads, and thanks to its dependency on the core part, the core
part gets autoloaded as well, resulting in a functional sound card
without having to manually load kernel modules.
Fixes: 2cb1e0259f ("ASoC: cs42l51: re-hook of_match_table pointer")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Petazzoni <thomas.petazzoni@bootlin.com>
Link: https://lore.kernel.org/r/20230713112112.778576-1-thomas.petazzoni@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 89dbb335cb ]
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 1b6a6fc528 ("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: 1b6a6fc528 ("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 60413129ee ]
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: b8b88b7087 ("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 6f07342903 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e123036be3 ]
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 040b5a046a ]
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 f63550e2b1 ]
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 c51e431052 ]
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 81302b1c7c 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: fc4f000bf8 ("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>
[ Upstream commit 92553ee031 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0d727e1856 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
