This is a quick follow up for 5.14: a fix for a very recently
introduced regression on ASoC Intel Atom driver, and another
trivial HD-audio quirk for HP laptops.
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Merge tag 'sound-5.14-rc7-2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull more sound fixes from Takashi Iwai:
"This is a quick follow up for 5.14: a fix for a very recently
introduced regression on ASoC Intel Atom driver, and another trivial
HD-audio quirk for HP laptops"
* tag 'sound-5.14-rc7-2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: intel: atom: Fix breakage for PCM buffer address setup
ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8
The commit 2e6b836312 ("ASoC: intel: atom: Fix reference to PCM
buffer address") changed the reference of PCM buffer address to
substream->runtime->dma_addr as the buffer address may change
dynamically. However, I forgot that the dma_addr field is still not
set up for the CONTINUOUS buffer type (that this driver uses) yet in
5.14 and earlier kernels, and it resulted in garbage I/O. The problem
will be fixed in 5.15, but we need to address it quickly for now.
The fix is to deduce the address again from the DMA pointer with
virt_to_phys(), but from the right one, substream->runtime->dma_area.
Fixes: 2e6b836312 ("ASoC: intel: atom: Fix reference to PCM buffer address")
Reported-and-tested-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@vger.kernel.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/2048c6aa-2187-46bd-6772-36a4fb3c5aeb@redhat.com
Link: https://lore.kernel.org/r/20210819152945.8510-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mic has lots of noises if mic boost is enabled. So disable mic boost
to get crystal clear audio capture.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hopefully the last PR for 5.14: here includes only a few regression
fixes and trivial device quirks.
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Merge tag 'sound-5.14-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Only a few regression fixes and trivial device quirks"
* tag 'sound-5.14-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/via: Apply runtime PM workaround for ASUS B23E
ALSA: hda: Fix hang during shutdown due to link reset
ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop
ALSA: oxfw: fix functioal regression for silence in Apogee Duet FireWire
ALSA: hda - fix the 'Capture Switch' value change notifications
The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio
setup as the Precision models, so requires the same quirk to enable woofer
output. Tested on my own 9510.
Signed-off-by: Kristin Paget <kristin@tombom.co.uk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OXFW 971 has no function to use the value in syt field of received
isochronous packet for playback timing generation. In kernel prepatch for
v5.14, ALSA OXFW driver got change to send NO_INFO value in the field
instead of actual timing value. The change brings Apogee Duet FireWire to
generate no playback sound, while output meter moves.
As long as I investigate, _any_ value in the syt field takes the device to
generate sound. It's reasonable to think that the device just ignores data
blocks in packet with NO_INFO value in its syt field for audio data
processing.
This commit adds a new flag for the quirk to fix regression.
Fixes: 029ffc4294 ("ALSA: oxfw: perform sequence replay for media clock recovery")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210812022839.42043-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original code in the cap_put_caller() function does not
handle correctly the positive values returned from the passed
function for multiple iterations. It means that the change
notifications may be lost.
Fixes: 352f7f914e ("ALSA: hda - Merge Realtek parser code to generic parser")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213851
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210811161441.1325250-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP ProBook 650 G8 Notebook PC is using ALC236 codec which is
using 0x02 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210810100846.65844-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix c4824ae7db ("ALSA: pcm: Fix mmap capability check")
restricts the mmap capability only to the drivers that properly set up
the buffers, but it caused a regression for a few drivers that manage
the buffer on its own way.
For those with UNKNOWN buffer type (i.e. the uninitialized / unused
substream->dma_buffer), just assume that the driver handles the mmap
properly and blindly trust the hardware info bit.
Fixes: c4824ae7db ("ALSA: pcm: Fix mmap capability check")
Reported-and-tested-by: Jeff Woods <jwoods@fnordco.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5him0gpghv.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS GV301QH sound appears to work well with the quirk for
ALC294_FIXUP_ASUS_DUAL_SPK.
Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210807025805.27321-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here is a collection of small fixes:
- A few regression fixes (PCM core fixes, USB-audio fixes)
- Follow up fixes for the USB-audio mixer changes in this cycle
- A long-standing ALSA sequencer race bug fix
- Usual device-specific quirks for HD- and USB-audio
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Merge tag 'sound-5.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- A few regression fixes (PCM core fixes, USB-audio fixes)
- Follow up fixes for the USB-audio mixer changes in this cycle
- A long-standing ALSA sequencer race bug fix
- Usual device-specific quirks for HD- and USB-audio"
* tag 'sound-5.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: seq: Fix racy deletion of subscriber
ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS
ALSA: pcm - fix mmap capability check for the snd-dummy driver
ALSA: usb-audio: Avoid unnecessary or invalid connector selection at resume
ALSA: hda/realtek: add mic quirk for Acer SF314-42
ALSA: usb-audio: Add registration quirk for JBL Quantum 600
ALSA: hda/realtek: Fix headset mic for Acer SWIFT SF314-56 (ALC256)
ALSA: usb-audio: Fix superfluous autosuspend recovery
ALSA: usb-audio: fix incorrect clock source setting
ALSA: scarlett2: Fix line out/speaker switching notifications
ALSA: scarlett2: Correct channel mute status after mute button pressed
ALSA: scarlett2: Fix Direct Monitor control name for 2i2
ALSA: scarlett2: Fix Mute/Dim/MSD Mode control names
Quite a lot of fixes here, the biggest set being for the cs42l42 driver
which is reasonably old but has seen a sudden uptick in activity.
There's also some fixes for correctly referencing PCM buffer addresses
and the removal of some driver-local bodges that had been done for the
lack of prefix handling in DAPM which were broken by the core handling
that as expected.
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Merge tag 'asoc-fix-v5.14-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.14
Quite a lot of fixes here, the biggest set being for the cs42l42 driver
which is reasonably old but has seen a sudden uptick in activity.
There's also some fixes for correctly referencing PCM buffer addresses
and the removal of some driver-local bodges that had been done for the
lack of prefix handling in DAPM which were broken by the core handling
that as expected.
I2S always has two LRCLK phases and both CH1 and CH2 of the RX
must be enabled (corresponding to the low and high phases of LRCLK.)
The selection of the valid data channels is done by setting the DAC
CHA_SEL and CHB_SEL. CHA_SEL is always the first (left) channel,
CHB_SEL depends on the number of active channels.
Previously for mono ASP CH2 was not enabled, the result was playing
mono data would not produce any audio output.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 621d65f3b8 ("ASoC: cs42l42: Provide finer control on playback path")
Link: https://lore.kernel.org/r/20210805161111.10410-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The lowest valid SCLK corresponds to 44.1 kHz at 16-bit. Sample
rates less than this would produce SCLK below the minimum when using
a normal I2S frame. A constraint must be applied to prevent this.
The constraint is not applied if the machine driver sets SCLK, to
allow setups where the host generates additional bits per LRCLK
phase to increase the SCLK frequency. In these cases the machine
driver would always have to inform this driver of the actual SCLK,
and it must select a legal SCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210805161111.10410-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An I2S frame starts on the falling edge of LRCLK so ASP_STP must
be 0.
At the same time, move other format settings in the same register
from cs42l42_pll_config() to cs42l42_set_dai_fmt() where you'd
expect to find them, and merge into a single write.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210805161111.10410-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Both SCLK and PLL clocks must be running to drive the glitch-free mux
behind MCLK_SRC_SEL and complete the switchover.
This patch moves the writing of MCLK_SRC_SEL to when the PLL is started
and stopped, so that it only transitions while the PLL is running.
The unconditional write MCLK_SRC_SEL=0 in cs42l42_mute_stream() is safe
because if the PLL is not running MCLK_SRC_SEL is already 0.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 43fc357199 ("ASoC: cs42l42: Set clock source for both ways of stream")
Link: https://lore.kernel.org/r/20210805161111.10410-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver was defining two ALSA controls that both change the same
register field for the wind noise filter corner frequency. The filter
response has two corners, at different frequencies, and the duplicate
controls most likely were an attempt to be able to set the value using
either of the frequencies.
However, having two controls changing the same field can be problematic
and it is unnecessary. Both frequencies are related to each other so
setting one implies exactly what the other would be.
Removing a control affects user-side code, but there is currently no
known use of the removed control so it would be best to remove it now
before it becomes a problem.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210803160834.9005-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The underlying register field has inverted sense (0 = enabled) so
the control definition must be marked as inverted.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210803160834.9005-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It turned out that the current implementation of the port subscription
is racy. The subscription contains two linked lists, and we have to
add to or delete from both lists. Since both connection and
disconnection procedures perform the same order for those two lists
(i.e. src list, then dest list), when a deletion happens during a
connection procedure, the src list may be deleted before the dest list
addition completes, and this may lead to a use-after-free or an Oops,
even though the access to both lists are protected via mutex.
The simple workaround for this race is to change the access order for
the disconnection, namely, dest list, then src list. This assures
that the connection has been established when disconnecting, and also
the concurrent deletion can be avoided.
Reported-and-tested-by: folkert <folkert@vanheusden.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210801182754.GP890690@belle.intranet.vanheusden.com
Link: https://lore.kernel.org/r/20210803114312.2536-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checking that two values don't have common bits makes no sense,
strict equality is meant.
Fixes: f3b433e469 ("ASoC: SOF: Implement Probe IPC API")
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210802151749.15417-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous Kconfig cleanup added simplifications but also introduced
a new one by moving a boolean to a tristate. This leads to randconfig
problems.
This patch moves the select operations in the SOUNDWIRE_LINK_BASELINE
option. The INTEL_SOUNDWIRE config remains a tristate for backwards
compatibility with older configurations but is essentially an on/off
switch.
Fixes: cf5807f5f8 ('ASoC: SOF: Intel: SoundWire: simplify Kconfig')
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Tested-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210802151628.15291-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer. The driver needs to refer to substream->runtime->dma_addr
instead for the buffer address.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210731084331.32225-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent code refactoring made the mmap of continuous pages to be
done via the own helper snd_dma_continuous_mmap() with
remap_pfn_range(). There I overlooked that dmab->addr isn't set for
the allocation with SNDRV_DMA_TYPE_CONTINUOUS. This resulted always
in an error at mmap with this buffer type on the system such as
Intel SST Baytrail driver.
This patch fixes the regression by passing the correct address.
Fixes: 30b7ba6972 ("ALSA: core: Add continuous and vmalloc mmap ops")
Reported-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/8d6674da-7d7b-803e-acc9-7de6cb1223fa@redhat.com
Link: https://lore.kernel.org/r/20210801113801.31290-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with the component layer code the nau8824 driver had been doing some
open coded pin manipulation which will have been broken now the core is
fixed to handle this properly, remove the open coding to avoid the issue.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210728234729.10135-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The transition to the managed PCM buffers allowed the dynamically
buffer allocation, while the driver code still assumes the fixed
preallocation buffer and sets up the DMA stuff at the open call.
This needs to be moved to hw_params after the buffer allocation and
setup. Also, the reference to the buffer address has to be corrected
to runtime->dma_addr.
Fixes: b3c0ae75f5 ("ASoC: kirkwood: Use managed DMA buffer allocation")
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Along with the transition to the managed PCM buffers, the driver now
accepts the dynamically allocated buffer, while it still kept the
reference to the old preallocated buffer address. This patch corrects
to the right reference via runtime->dma_addr.
(Although this might have been already buggy before the cleanup with
the managed buffer, let's put Fixes tag to point that; it's a corner
case, after all.)
Fixes: d55894bc27 ("ASoC: uniphier: Use managed buffer allocation")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer. The driver needs to refer to substream->runtime->dma_addr
instead for the buffer address.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer. The address should be retrieved from runtime->dma_addr,
instead of substream->dma_buffer (and shouldn't use virt_to_phys).
Also, remove the line overriding runtime->dma_area superfluously,
which was already set up at the PCM buffer allocation.
Cc: Cezary Rojewski <cezary.rojewski@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for the resume on Lenovo machines seems causing a
regression on others. It's because the change always triggers the
connector selection no matter which widget node type is.
This patch addresses the regression by setting the resume callback
selectively only for the connector widget.
Fixes: 44609fc01f ("ALSA: usb-audio: Check connector value on resume")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897
Link: https://lore.kernel.org/r/20210729185126.24432-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An I2S frame always has a left and right channel slot even if mono
data is being sent. So if channels==1 the actual bitclock frequency
is 2 * snd_soc_params_to_bclk(params).
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2cdba9b045 ("ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called")
Link: https://lore.kernel.org/r/20210729170929.6589-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has no support for left-justified protocol so it should
not have been allowing this to be passed to cs42l42_set_dai_fmt().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210729170929.6589-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC volume is a signed 8-bit number with range -97 to +12,
with -97 being mute. Use a SOC_SINGLE_S8_TLV() to define this
and fix the DECLARE_TLV_DB_SCALE() to have the correct start and
mute flag.
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210729170929.6589-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer Swift SF314-42 laptop is using Realtek ALC255 codec. Add a
quirk so microphone in a headset connected via the right-hand side jack
is usable.
Signed-off-by: Alexander Monakov <amonakov@ispras.ru>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210721170141.24807-1-amonakov@ispras.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_cleanup_component_debugfs will debugfs_remove_recursive
the component->debugfs_root, so adsp doesn't need to also
remove the same entry.
By doing that adsp also creates a race with core component,
which causes a NULL pointer dereference
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the component level pin control functions were added they for some
no longer obvious reason handled adding prefixing of widget names. This
meant that when the lack of prefix handling in the DAPM level pin
operations was fixed by ae4fc53224 (ASoC: dapm: use component
prefix when checking widget names) the one device using the component
level API ended up with the prefix being applied twice, causing all
lookups to fail.
Fix this by removing the redundant prefixing from the component code,
which has the nice side effect of also making that code much simpler.
Reported-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tested-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210726194123.54585-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On some platforms with an external HDaudio codec, the DSDT reports the
presence of SoundWire devices. Pin-mux restrictions and board reworks
usually prevent coexistence between the two types of links, let's
prevent unnecessary operations from starting.
In the case of a single iDISP codec being detected, we still start the
links even if no SoundWire machine configuration was detected, so that
we can double-check what the hardware is and add the missing
configuration if applicable.
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 <bard.liao@intel.com>
Link: https://lore.kernel.org/r/20210726182855.179943-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The indexes of the devices are described within the topology file, it is a
possibility that the topology encodes invalid indexes when DYNAMIC_MINORS
is not enabled in kernel:
#define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */
#define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */
#define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */
#define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */
#define SNDRV_MINOR_PCM_CAPTURE 24 /* 24 - 31 */
If the topology assigns an index greater than 7 for PLAYBACK/CAPTURE PCM
then there will be minor number collision.
As an example:
card0 creates a capture PCM with index 10 -> minor = 34
card1 creates compress device with index 0 -> minor = 34
Card1 will fail to instantiate because the minor for the compress stream is
already taken.
To avoid seemingly mysterious issues with card creation, select the
DYNAMIC_MINORS when the topology is enabled.
The other option would be to try to do out of bound index checks in case of
DYNAMIC_MINOR is not enabled and do not even attempt to create the device
with failing the topology load.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210726182142.179604-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Apparently JBL Quantum 600 has multiple hardware revisions. Apply
registration quirk to another device id as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727093326.1153366-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The issue on Acer SWIFT SF314-56 is that headset microphone doesn't work.
The following quirk fixed headset microphone issue. The fixup was found by trial and error.
Note that the fixup of SF314-54/55 (ALC256_FIXUP_ACER_HEADSET_MIC) was not successful on my SF314-56.
Signed-off-by: Nikos Liolios <liolios.nk@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727030510.36292-1-liolios.nk@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default codec for speaker amp's DAI Link is max98373 and will be
overwritten in probe function if the board id is sof_da7219_mx98360a.
However, the probe function does not do it because the board id is
changed in earlier commit.
Fixes: 1cc04d195d ("ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters")
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210726094525.5748-1-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The change to restore the autosuspend from the disabled state uses a
wrong check: namely, it should have been the exact comparison of the
quirk_type instead of the bitwise and (&). Otherwise it matches
wrongly with the other quirk types.
Although re-enabling the autosuspend for the already enabled device
shouldn't matter much, it's better to fix the unbalanced call.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hr1flh9ov.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tlv320aic31xx driver relies on regcache_sync() to restore the register
contents after going to _BIAS_OFF, for example during system suspend. This
does not work for the jack detection configuration since that is configured
via the same register that status is read back from so the register is
volatile and not cached. This can also cause issues during init if the jack
detection ends up getting set up before the CODEC is initially brought out
of _BIAS_OFF, we will reset the CODEC and resync the cache as part of that
process.
Fix this by explicitly reapplying the jack detection configuration after
resyncing the register cache during power on.
This issue was found by an engineer working off-list on a product
kernel, I just wrote up the upstream fix.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210723180200.25105-1-broonie@kernel.org
Cc: stable@vger.kernel.org
The following scenario describes an echo test for
Samsung USBC Headset (AKG) with VID/PID (0x04e8/0xa051).
We first start a capture stream(USB IN transfer) in 96Khz/24bit/1ch mode.
In clock find source function, we get value 0x2 for clock selector
and 0x1 for clock source.
Kernel-4.14 behavior
Since clock source is valid so clock selector was not set again.
We pass through this function and start a playback stream(USB OUT transfer)
in 48Khz/32bit/2ch mode. This time we get value 0x1 for clock selector
and 0x1 for clock source. Finally clock id with this setting is 0x9.
Kernel-5.10 behavior
Clock selector was always set one more time even it is valid.
When we start a playback stream, we will get 0x2 for clock selector
and 0x1 for clock source. In this case clock id becomes 0xA.
This is an incorrect clock source setting and results in severe noises.
We see wrong data rate in USB IN transfer.
(From 288 bytes/ms becomes 144 bytes/ms) It should keep in 288 bytes/ms.
This earphone works fine on older kernel version load because
this is a newly-added behavior.
Fixes: d2e8f64125 ("ALSA: usb-audio: Explicitly set up the clock selector")
Signed-off-by: chihhao.chen <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/1627100621-19225-1-git-send-email-chihhao.chen@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The values of the line output controls can change when the SW/HW
switches are set to HW, and also when speaker switching is enabled.
These notifications were sent with a mask of only
SNDRV_CTL_EVENT_MASK_INFO. Change the notifications to set the
SNDRV_CTL_EVENT_MASK_VALUE mask bit as well.
When the mute control is updated, the notification was sent with a
mask of SNDRV_CTL_EVENT_MASK_INFO. Change the mask to the correct
value of SNDRV_CTL_EVENT_MASK_VALUE.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/8192e15ba62fa4bc90425c005f265c0de530be20.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the hardware mute button is pressed, private->vol_updated is set
so that the mute status is invalidated. As the channel mute values may
be affected by the global mute value, update scarlett2_mute_ctl_get()
to call scarlett2_update_volumes() if private->vol_updated is set.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/aa18ddbf8d8bd7f31832ab1b6b6057c00b931202.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
