commit 88e98af9f4 upstream.
When suspended, the DMA channel may enter PAUSE state if dmaengine_pause()
is supported by DMA.
At this state, dmaengine_synchronize() should not be called, otherwise
the DMA channel can't be resumed successfully.
Fixes: e8343410dd ("ALSA: dmaengine: Synchronize dma channel after drop()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/1721198693-27636-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d7063c0873 upstream.
Samsung Galaxy Book Pro 360 (13" 2022 NT935QDB-KC71S) with codec SSID
144d:c1a4 requires the same workaround to enable the speaker amp
as other Samsung models with the ALC298 codec.
Signed-off-by: Seunghun Han <kkamagui@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240718080908.8677-1-kkamagui@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8fc1e8b230 upstream.
Positivo SU C1400 is equipped with ALC256, and it needs
ALC269_FIXUP_ASPIRE_HEADSET_MIC quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240712180642.22564-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6a7db25aad ]
When dmaengine supports pause function, in suspend state,
dmaengine_pause() is called instead of dmaengine_terminate_async(),
In end of playback stream, the runtime->state will go to
SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen
at this time, application will not resume playback stream, the
stream will be closed directly, the dmaengine_terminate_async()
will not be called before the dmaengine_synchronize(), which
violates the call sequence for dmaengine_synchronize().
This behavior also happens for capture streams, but there is no
SNDRV_PCM_STATE_DRAINING state for capture. So use
dmaengine_tx_status() to check the DMA status if the status is
DMA_PAUSED, then call dmaengine_terminate_async() to terminate
dmaengine before dmaengine_synchronize().
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718851218-27803-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 524d3f1263 ]
Set driver name to "HDMI". This simplifies the code and gets rid of
the following error messages:
ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e'
Signed-off-by: Primoz Fiser <primoz.fiser@norik.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c5dcf8ab10 ]
The minimum period size was enforced to 64 as older devices integrating
McASP with EDMA used an internal FIFO of 64 samples.
With UDMA based platforms this internal McASP FIFO is optional, as the
DMA engine internally does some buffering which is already accounted for
when registering the platform. So we should read the actual FIFO
configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to
64.
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e8343410dd ]
Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e3209a1827 ]
When headphones are plugged in, they appear absent; when they are removed,
they appear present.
Add a specific entry in bytcr_rt5640 for this device
Signed-off-by: Thomas GENTY <tomlohave@gmail.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 6db03b1929 upstream.
The internal mic boost on the VAIO models VJFE-CL and VJFE-IL is too high.
Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine
to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240705141012.5368-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b46953029c upstream.
HP 250 G7 has a mute LED that can be made to work using quirk
ALC269_FIXUP_HP_LINE1_MIC1_LED. Enable already existing quirk.
Signed-off-by: Nazar Bilinskyi <nbilinskyi@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240709080546.18344-1-nbilinskyi@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 89b32ccb12 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 90f3feb241 ]
priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 86a4338629 upstream.
The internal mic boost on the N14AP7 is too high. Fix this by applying the
ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240605153923.2837-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4a63bd179f upstream.
Currently ALSA timer doesn't have the lower limit of the start tick
time, and it allows a very small size, e.g. 1 tick with 1ns resolution
for hrtimer. Such a situation may lead to an unexpected RCU stall,
where the callback repeatedly queuing the expire update, as reported
by fuzzer.
This patch introduces a sanity check of the timer start tick time, so
that the system returns an error when a too small start size is set.
As of this patch, the lower limit is hard-coded to 100us, which is
small enough but can still work somehow.
Reported-by: syzbot+43120c2af6ca2938cc38@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/r/000000000000fa00a1061740ab6d@google.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240514182745.4015-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ backport note: the error handling is changed, as the original commit
is based on the recent cleanup with guard() in commit beb45974dd
-- tiwai ]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7078ac4fd1 ]
TAS2552 is a Smartamp with I/V sense data, add TX path
to support capturing I/V data.
Fixes: 38803ce7b5 ("ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://msgid.link/r/20240518033515.866-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a85ed162f0 ]
For DSP_A, data is a BCK cycle behind LRCK trigger edge. For DSP_B, this
delay doesn't exist. Fix the delay configuration to match the standard.
Fixes: 52fcd65414 ("ASoC: mediatek: mt8192: support tdm in platform driver")
Signed-off-by: Hsin-Te Yuan <yuanhsinte@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://lore.kernel.org/r/20240509-8192-tdm-v1-1-530b54645763@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ea60ab9572 ]
In kirkwood_dma_hw_params() mv_mbus_dram_info() returns NULL if
CONFIG_PLAT_ORION macro is not defined.
Fix this bug by adding NULL check.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: bb6a40fc5a ("ASoC: kirkwood: Fix reference to PCM buffer address")
Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru>
Link: https://msgid.link/r/20240328173337.21406-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0cb3b7fd53 ]
Topology files that are propagated to the world and utilized by the
skylake-driver carry shortcomings in their SectionGraphs.
Since commit daa480bde6 ("ASoC: soc-core: tidyup for
snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe
failures for Intel boards have been partially addressed by commit
a22ae72b86 ("ASoC: soc-core: disable route checks for legacy devices")
and its follow up but only skl_nau88l25_ssm4567.c is patched. Fix the
problem for the rest of the boards.
Link: https://lore.kernel.org/all/20200309192744.18380-1-pierre-louis.bossart@linux.intel.com/
Fixes: daa480bde6 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-2-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e8a6a5ad73 ]
The documentation for device_get_named_child_node() mentions this
important point:
"
The caller is responsible for calling fwnode_handle_put() on the
returned fwnode pointer.
"
Add fwnode_handle_put() to avoid a leaked reference.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153033.38500-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bda16500dd ]
Volume step (dB/step) modification to fix format error
which shown in amixer control.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/b1f546ad16dc4c7abb7daa7396e8345c@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 103abab975 ]
The codec leaves tie combo jack's sleeve/ring2 to floating status
default. It would cause electric noise while connecting the active
speaker jack during boot or shutdown.
This patch requests a gpio to control the additional jack circuit
to tie the contacts to the ground or floating.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://msgid.link/r/20240408091057.14165-1-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 87988a534d upstream.
In snd_card_disconnect(), we set card->shutdown flag at the beginning,
call callbacks and do sync for card->power_ref_sleep waiters at the
end. The callback may delete a kctl element, and this can lead to a
deadlock when the device was in the suspended state. Namely:
* A process waits for the power up at snd_power_ref_and_wait() in
snd_ctl_info() or read/write() inside card->controls_rwsem.
* The system gets disconnected meanwhile, and the driver tries to
delete a kctl via snd_ctl_remove*(); it tries to take
card->controls_rwsem again, but this is already locked by the
above. Since the sleeper isn't woken up, this deadlocks.
An easy fix is to wake up sleepers before processing the driver
disconnect callbacks but right after setting the card->shutdown flag.
Then all sleepers will abort immediately, and the code flows again.
So, basically this patch moves the wait_event() call at the right
timing. While we're at it, just to be sure, call wait_event_all()
instead of wait_event(), although we don't use exclusive events on
this queue for now.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218816
Cc: <stable@vger.kernel.org>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240510101424.6279-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 39381fe739 upstream.
The commit 81033c6b58 ("ALSA: core: Warn on empty module")
introduced a WARN_ON() for a NULL module pointer passed at snd_card
object creation, and it also wraps the code around it with '#ifdef
MODULE'. This works in most cases, but the devils are always in
details. "MODULE" is defined when the target code (i.e. the sound
core) is built as a module; but this doesn't mean that the caller is
also built-in or not. Namely, when only the sound core is built-in
(CONFIG_SND=y) while the driver is a module (CONFIG_SND_USB_AUDIO=m),
the passed module pointer is ignored even if it's non-NULL, and
card->module remains as NULL. This would result in the missing module
reference up/down at the device open/close, leading to a race with the
code execution after the module removal.
For addressing the bug, move the assignment of card->module again out
of ifdef. The WARN_ON() is still wrapped with ifdef because the
module can be really NULL when all sound drivers are built-in.
Note that we keep 'ifdef MODULE' for WARN_ON(), otherwise it would
lead to a false-positive NULL module check. Admittedly it won't catch
perfectly, i.e. no check is performed when CONFIG_SND=y. But, it's no
real problem as it's only for debugging, and the condition is pretty
rare.
Fixes: 81033c6b58 ("ALSA: core: Warn on empty module")
Reported-by: Xu Yang <xu.yang_2@nxp.com>
Closes: https://lore.kernel.org/r/20240520170349.2417900-1-xu.yang_2@nxp.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Xu Yang <xu.yang_2@nxp.com>
Link: https://lore.kernel.org/r/20240522070442.17786-1-tiwai@suse.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d18ca8635d upstream.
When using davinci-mcasp as CPU DAI with simple-card, there are some
conditions that cause simple-card to finish registering a sound card before
davinci-mcasp finishes registering all sound components. This creates a
non-working sound card from userspace with no problem indication apart
from not being able to play/record audio on a PCM stream. The issue
arises during simultaneous probe execution of both drivers. Specifically,
the simple-card driver, awaiting a CPU DAI, proceeds as soon as
davinci-mcasp registers its DAI. However, this process can lead to the
client mutex lock (client_mutex in soc-core.c) being held or davinci-mcasp
being preempted before PCM DMA registration on davinci-mcasp finishes.
This situation occurs when the probes of both drivers run concurrently.
Below is the code path for this condition. To solve the issue, defer
davinci-mcasp CPU DAI registration to the last step in the audio part of
it. This way, simple-card CPU DAI parsing will be deferred until all
audio components are registered.
Fail Code Path:
simple-card.c: probe starts
simple-card.c: simple_dai_link_of: simple_parse_node(..,cpu,..) returns EPROBE_DEFER, no CPU DAI yet
davinci-mcasp.c: probe starts
davinci-mcasp.c: devm_snd_soc_register_component() register CPU DAI
simple-card.c: probes again, finish CPU DAI parsing and call devm_snd_soc_register_card()
simple-card.c: finish probe
davinci-mcasp.c: *dma_pcm_platform_register() register PCM DMA
davinci-mcasp.c: probe finish
Cc: stable@vger.kernel.org
Fixes: 9fbd58cf4a ("ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller")
Signed-off-by: Joao Paulo Goncalves <joao.goncalves@toradex.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Reviewed-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240417184138.1104774-1-jpaulo.silvagoncalves@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2e93a29b48 upstream.
DSPK configuration is wrong for 16-bit playback and this happens because
the client config is always fixed at 24-bit in hw_params(). Fix this by
updating the client config to 16-bit for the respective playback.
Fixes: 327ef64702 ("ASoC: tegra: Add Tegra186 based DSPK driver")
Cc: stable@vger.kernel.org
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://msgid.link/r/20240405104306.551036-1-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit c4e51e424e ]
For shutting up spurious KMSAN uninit-value warnings, just replace
kmalloc() calls with kzalloc() for the buffers used for
communications. There should be no real issue with the original code,
but it's still better to cover.
Reported-by: syzbot+7fb05ccf7b3d2f9617b3@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/r/00000000000084b18706150bcca5@google.com
Message-ID: <20240402063628.26609-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c158cf9147 ]
The documentation for device_get_named_child_node() mentions this
important point:
"
The caller is responsible for calling fwnode_handle_put() on the
returned fwnode pointer.
"
Add fwnode_handle_put() to avoid a leaked reference.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Fixes: 08c2a4bc9f ("ALSA: hda: move Intel SoundWire ACPI scan to dedicated module")
Message-ID: <20240426152731.38420-1-pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6db26f9ea4 ]
Amlogic sound cards do create a lot of pcm interfaces, possibly more than
8. Some pcm interfaces are internal (like DPCM backends and c2c) and not
exposed to userspace.
Those interfaces still increase the number passed to snd_find_free_minor(),
which eventually exceeds 8 causing -EBUSY error on card registration if
CONFIG_SND_DYNAMIC_MINORS=n and the interface is exposed to userspace.
select CONFIG_SND_DYNAMIC_MINORS for Amlogic cards to avoid the problem.
Fixes: 7864a79f37 ("ASoC: meson: add axg sound card support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426134150.3053741-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bf5e4887ee ]
So far, the formatters have been reset/enabled using the .prepare()
callback. This was done in this callback because walking the formatters use
a mutex so it could not be done in .trigger(), which is atomic by default.
It turns out there is a problem on capture path of the AXG series.
The FIFO may get out of sync with the TDM decoder if the IP are not enabled
in a specific order. The FIFO must be enabled before the formatter starts
producing data. IOW, we must deal with FE before the BE. The .prepare()
callback is called on the BEs before the FE so it is not OK for the AXG.
The .trigger() callback order can be configured, and it deals with the FE
before the BEs by default. To solve our problem, we just need to start and
stop the formatters from the .trigger() callback. It is OK do so now that
the links have been made 'nonatomic' in the card driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e138233e56 ]
Non atomic operations need to be performed in the trigger callback
of the TDM interfaces. Those are BEs but what matters is the nonatomic
flag of the FE in the DPCM context. Just set nonatomic for everything so,
at least, it is clear.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b11d26660d ]
With the AXG audio subsystem, there is a possible random channel shift on
TDM capture, when the slot number per lane is more than 2, and there is
more than one lane used.
The problem has been there since the introduction of the axg audio support
but such scenario is pretty uncommon. This is why there is no loud
complains about the problem.
Solving the problem require to make the links non-atomic and use the
trigger() callback to start FEs and BEs in the appropriate order.
This was tried in the past and reverted because it caused the block irq to
sleep while atomic. However, instead of reverting, the solution is to call
snd_pcm_period_elapsed() in a non atomic context.
Use the bottom half of a threaded IRQ to do so.
Fixes: 6dc4fa179f ("ASoC: meson: add axg fifo base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426152946.3078805-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9e6f39535c ]
Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240227150826.573581-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: b11d26660d ("ASoC: meson: axg-fifo: use threaded irq to check periods")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 23fb6bc269 ]
When pcm_runtime is adding platform components it will scan all
registered components. In case of DPCM FE/BE some DAI links will
configure dummy platform. However both dummy codec and dummy platform
are using "snd-soc-dummy" as component->name. Dummy codec should be
skipped when adding platforms otherwise there'll be overflow and UBSAN
complains.
Reported-by: Zhipeng Wang <zhipeng.wang_1@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240305065606.3778642-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a486439d2 ]
Miglia Harmony Audio (OXFW970) has a quirk to put the number of
accumulated quadlets in CIP payload into the dbc field of CIP header.
This commit handles the quirk in the packet processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 1576f263ee upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit fc563aa900 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit aae86cfd87 ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: b69de265bd ("ASoC: rt711: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ee28777164 ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 23adeb7056 ("ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 310a5caa4e ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 02fb23d727 ("ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 051e0840ff upstream.
The dreamcastcard->timer could schedule the spu_dma_work and the
spu_dma_work could also arm the dreamcastcard->timer.
When the snd_pcm_substream is closing, the aica_channel will be
deallocated. But it could still be dereferenced in the worker
thread. The reason is that del_timer() will return directly
regardless of whether the timer handler is running or not and
the worker could be rescheduled in the timer handler. As a result,
the UAF bug will happen. The racy situation is shown below:
(Thread 1) | (Thread 2)
snd_aicapcm_pcm_close() |
... | run_spu_dma() //worker
| mod_timer()
flush_work() |
del_timer() | aica_period_elapsed() //timer
kfree(dreamcastcard->channel) | schedule_work()
| run_spu_dma() //worker
... | dreamcastcard->channel-> //USE
In order to mitigate this bug and other possible corner cases,
call mod_timer() conditionally in run_spu_dma(), then implement
PCM sync_stop op to cancel both the timer and worker. The sync_stop
op will be called from PCM core appropriately when needed.
Fixes: 198de43d75 ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Duoming Zhou <duoming@zju.edu.cn>
Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d397b6e561 upstream.
Headset Mic will no show at resume back.
This patch will fix this issue.
Fixes: d7f32791a9 ("ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4713d48a372e47f98bba0c6120fd8254@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit a39d51ff1f ]
If a usb audio device sets more bits than the amount of channels
it could write outside of the map array.
Signed-off-by: Johan Carlsson <johan.carlsson@teenage.engineering>
Fixes: 04324ccc75 ("ALSA: usb-audio: add channel map support")
Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c062166995 ]
Realtek codec on HP Envy laptop series are heavily modified by vendor.
Therefore, need intervention to make it work properly. The patch fixes:
- B&O soundbar speakers (between lid and keyboard) activation
- Enable LED on mute button
- Add missing process coefficient which affects the output amplifier
- Volume control synchronization between B&O soundbar and side speakers
- Unmute headset output on several HP Envy models
- Auto-enable headset mic when plugged
This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285
The only unsolved problem is output amplifier of all built-in speakers
is too weak, which causes volume of built-in speakers cannot be loud
as vendor's proprietary driver due to missing _DSD parameter in the
firmware. The solution is currently on research. Expected to has another
patch in the future.
Potential fix to related issues, need test before close those issues:
- https://bugzilla.kernel.org/show_bug.cgi?id=189331
- https://bugzilla.kernel.org/show_bug.cgi?id=216632
- https://bugzilla.kernel.org/show_bug.cgi?id=216311
- https://bugzilla.kernel.org/show_bug.cgi?id=213507
Signed-off-by: Athaariq Ardhiansyah <foss@athaariq.my.id>
Message-ID: <20240310140249.3695-1-foss@athaariq.my.id>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 59c6a3a43b ]
According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.
The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.
Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.
For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.
Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
