Jeremy Szu [Wed, 19 May 2021 17:03:53 +0000 (01:03 +0800)]
ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs
ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the
fixup, the mute/micmute LEDs work good.
Peter Ujfalusi [Mon, 24 May 2021 20:37:26 +0000 (23:37 +0300)]
ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces
bug plock/pop noise over line out (green jack on the back) which can be
fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine.
Convert the existing entry for the same SSID to chain to apply this fixup
as well.
Takashi Iwai [Tue, 25 May 2021 06:58:01 +0000 (08:58 +0200)]
Merge tag 'asoc-fix-v5.13-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.13
A collection of fixes that have come in since the merge window, mainly
device specific things. The fixes to the generic cards from
Morimoto-san are handling regressions that were introduced in the merge
window on at least the Kontron sl28-var3-ads2.
Hui Wang [Sat, 22 May 2021 04:26:45 +0000 (12:26 +0800)]
ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i
The Lenovo Yoga 9i has bass speaker, but the bass speaker can't work,
that is because there is an i2s amplifier on that speaker, need to
run ideapad_s740_coef() to initialize the amplifier.
And also needs to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK to rename
the speaker's mixer control name, otherwise the PA can't handle them.
Hui Wang [Sat, 22 May 2021 03:47:41 +0000 (11:47 +0800)]
ALSA: hda/realtek: Headphone volume is controlled by Front mixer
On some ASUS and MSI machines, the audio codec is alc1220 and the
Headphone is connected to audio mixer 0xf and DAC 0x5, in theory
the Headphone volume is controlled by DAC 0x5 (Heapdhone Playback
Volume), but somehow it is controlled by DAC 0x2 (Front Playback
Volume), maybe this is a defect on the codec alc1220.
Because of this issue, the PA couldn't switch the headphone and
Lineout correctly, If we apply the quirk CLEVO_P950 to those machines,
the Lineout and Headphone will share the audio mixer 0xc and DAC 0x2,
and generate Headphone+LO mixer, then PA could handle them when
switching between them.
ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci
Use usb_rcvctrlpipe() not usb_sndctrlpipe() for USB control input in
the Scarlett Gen 2 mixer driver. This fixes the device hang during
initialisation when used with the ehci-pci host driver.
Johan Hovold [Fri, 21 May 2021 13:37:42 +0000 (15:37 +0200)]
ALSA: usb-audio: fix control-request direction
The direction of the pipe argument must match the request-type direction
bit or control requests may fail depending on the host-controller-driver
implementation.
Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe().
Takashi Iwai [Tue, 18 May 2021 08:39:39 +0000 (10:39 +0200)]
ALSA: line6: Fix racy initialization of LINE6 MIDI
The initialization of MIDI devices that are found on some LINE6
drivers are currently done in a racy way; namely, the MIDI buffer
instance is allocated and initialized in each private_init callback
while the communication with the interface is already started via
line6_init_cap_control() call before that point. This may lead to
Oops in line6_data_received() when a spurious event is received, as
reported by syzkaller.
This patch moves the MIDI initialization to line6_init_cap_control()
as well instead of the too-lately-called private_init for avoiding the
race. Also this reduces slightly more lines, so it's a win-win
change.
BE hw_params op was recently added for SSP type DAIs.
But sending the DAI_CONFIG IPC during hw_params
is not supported with older firmware. So add an ABI check
to avoid sending the IPC if the firmware ABI is older than
3.18.
Takashi Sakamoto [Tue, 18 May 2021 01:26:12 +0000 (10:26 +0900)]
ALSA: dice: fix stream format for TC Electronic Konnekt Live at high sampling transfer frequency
At high sampling transfer frequency, TC Electronic Konnekt Live
transfers/receives 6 audio data frames in multi bit linear audio data
channel of data block in CIP payload. Current hard-coded stream format
is wrong.
Although many DICE-based devices have a quirk at high sampling transfer
frequency to multiplex double number of PCM frames into data block than
the number in IEC 61883-1/6, the above devices are just compliant to
IEC 61883-1/6.
This commit disables the mode of double_pcm_frames for the models.
Takashi Iwai [Sun, 16 May 2021 16:17:55 +0000 (18:17 +0200)]
ALSA: intel8x0: Don't update period unless prepared
The interrupt handler of intel8x0 calls snd_intel8x0_update() whenever
the hardware sets the corresponding status bit for each stream. This
works fine for most cases as long as the hardware behaves properly.
But when the hardware gives a wrong bit set, this leads to a zero-
division Oops, and reportedly, this seems what happened on a VM.
For fixing the crash, this patch adds a internal flag indicating that
the stream is ready to be updated, and check it (as well as the flag
being in suspended) to ignore such spurious update.
Takashi Sakamoto [Thu, 13 May 2021 12:56:52 +0000 (21:56 +0900)]
ALSA: firewire-lib: fix amdtp_packet tracepoints event for packet_index field
The snd_firewire_lib:amdtp_packet tracepoints event includes index of
packet processed in a context handling. However in IR context, it is not
calculated as expected.
Takashi Sakamoto [Thu, 13 May 2021 12:56:51 +0000 (21:56 +0900)]
ALSA: firewire-lib: fix calculation for size of IR context payload
The quadlets for CIP header is handled as a part of IR context header,
thus it doesn't join in IR context payload. However current calculation
includes the quadlets in IR context payload.
Takashi Sakamoto [Thu, 13 May 2021 12:56:49 +0000 (21:56 +0900)]
ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro
Mackie d.2 has an extension card for IEEE 1394 communication, which uses
BridgeCo DM1000 ASIC. On the other hand, Mackie d.4 Pro has built-in
function for IEEE 1394 communication by Oxford Semiconductor OXFW971,
according to schematic diagram available in Mackie website. Although I
misunderstood that Mackie d.2 Pro would be also a model with OXFW971,
it's wrong. Mackie d.2 Pro is a model which includes the extension card
as factory settings.
This commit fixes entries in Kconfig and comment in ALSA OXFW driver.
Takashi Sakamoto [Thu, 13 May 2021 12:56:48 +0000 (21:56 +0900)]
ALSA: dice: fix stream format at middle sampling rate for Alesis iO 26
Alesis iO 26 FireWire has two pairs of digital optical interface. It
delivers PCM frames from the interfaces by second isochronous packet
streaming. Although both of the interfaces are available at 44.1/48.0
kHz, first one of them is only available at 88.2/96.0 kHz. It reduces
the number of PCM samples to 4 in Multi Bit Linear Audio data channel
of data blocks on the second isochronous packet streaming.
This device requires single register transactions, this will
definely cause problems with the new device ID parsing which uses
regmap_bulk_read but might also show up in the cache sync sometimes.
Add the missing flags to the regmap_config.
This device requires single register transactions, this will
definely cause problems with the new device ID parsing which uses
regmap_bulk_read but might also show up in the cache sync sometimes.
Add the missing flags to the regmap_config.
This device requires single register transactions, this will
definely cause problems with the new device ID parsing which uses
regmap_bulk_read but might also show up in the cache sync sometimes.
Add the missing flags to the regmap_config.
This device requires single register transactions, this will
definely cause problems with the new device ID parsing which
uses regmap_bulk_read but might also show up in the cache sync
sometimes. Add the missing flags to the regmap_config.
Zou Wei [Wed, 12 May 2021 03:12:25 +0000 (11:12 +0800)]
ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
This patch adds missing MODULE_DEVICE_TABLE definition which generates
correct modalias for automatic loading of this driver when it is built
as an external module.
Main issue I'm understanding is name create timing.
We want to create dailink->name via dlc->dai_name.
But in CPU case, this dai_name might be removed by asoc_simple_canonicalize_cpu()
if it CPU was single DAI.
Thus, we need to
A) get dlc->dai_name
B) create dailink->name via dlc->dai_name
C) call asoc_simple_canonicalize_cpu()
Above reverted patch did A->C->B.
My previous v1 patch did B->A->C.
I'm so sorry that I didn't deep test on v1.
I hope v2 patches has no issues on kontron-sl28-var3-ads2.
audio-graph is using cpus->dai_name / codecs->dai_name for
dailink->name.
In graph_parse_node(), xxx->dai_name is got by
snd_soc_get_dai_name(), but it might be removed soon by
asoc_simple_canonicalize_cpu().
The order should be
*1) call snd_soc_get_dai_name()
2) create dailink name
*3) call asoc_simple_canonicalize_cpu()
* are implemented in graph_parse_node().
This patch remove 3) from graph_parse_node()
Hans de Goede [Sat, 8 May 2021 15:01:46 +0000 (17:01 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet
The Lenovo Miix 3-830 tablet has only 1 speaker, has an internal analog
mic on IN1 and uses JD2 for jack-detect, add a quirk to automatically
apply these settings on Lenovo Miix 3-830 tablets.
Hans de Goede [Sat, 8 May 2021 15:01:45 +0000 (17:01 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet
Add a quirk for the Glavey TM800A550L tablet, this BYTCR tablet has no CHAN
package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which
is the default for BYTCR devices.
Takashi Iwai [Tue, 11 May 2021 09:05:00 +0000 (11:05 +0200)]
ALSA: usb-audio: Fix potential out-of-bounce access in MIDI EP parser
The recently introduced MIDI endpoint parser code has an access to the
field without the size validation, hence it might lead to
out-of-bounce access. Add the sanity checks for the descriptor
sizes.
Takashi Iwai [Mon, 10 May 2021 15:06:59 +0000 (17:06 +0200)]
ALSA: usb-audio: Validate MS endpoint descriptors
snd_usbmidi_get_ms_info() may access beyond the border when a
malformed descriptor is passed. This patch adds the sanity checks of
the given MS endpoint descriptors, and skips invalid ones.
This patch adds missing MODULE_DEVICE_TABLE definition which generates
correct modalias for automatic loading of this driver when it is built
as an external module.
The GU502 requires a few steps to make headset i/o works properly:
pincfg, verbs to unmute headphone out and callback to toggle output
between speakers and headphone using jack.
Hui Wang [Fri, 7 May 2021 02:44:52 +0000 (10:44 +0800)]
ALSA: hda/realtek: reset eapd coeff to default value for alc287
Ubuntu users reported an audio bug on the Lenovo Yoga Slim 7 14IIL05,
he installed dual OS (Windows + Linux), if he booted to the Linux
from Windows, the Speaker can't work well, it has crackling noise,
if he poweroff the machine first after Windows, the Speaker worked
well.
Before rebooting or shutdown from Windows, the Windows changes the
codec eapd coeff value, but the BIOS doesn't re-initialize its value,
when booting into the Linux from Windows, the eapd coeff value is not
correct. To fix it, set the codec default value to that coeff register
in the alsa driver.
Hui Wang [Tue, 4 May 2021 07:39:17 +0000 (15:39 +0800)]
ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
Without this change, the DAC ctl's name could be changed only when
the machine has both Speaker and Headphone, but we met some machines
which only has Lineout and Headhpone, and the Lineout and Headphone
share the Audio Mixer0 and DAC0, the ctl's name is set to "Front".
On most of machines, the "Front" is used for Speaker only or Lineout
only, but on this machine it is shared by Lineout and Headphone,
This introduces an issue in the pipewire and pulseaudio, suppose users
want the Headphone to be on and the Speaker/Lineout to be off, they
could turn off the "Front", this works on most of the machines, but on
this machine, the "Front" couldn't be turned off otherwise the
headphone will be off too. Here we do some change to let the ctl's
name change to "Headphone+LO" on this machine, and pipewire and
pulseaudio already could handle "Headphone+LO" and "Speaker+LO".
(https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747)
Takashi Iwai [Tue, 4 May 2021 12:18:32 +0000 (14:18 +0200)]
ALSA: hda/realtek: Add fixup for HP OMEN laptop
HP OMEN dc0019-ur with codec SSID 103c:84da requires the pin config
overrides and the existing mic/mute LED setup. This patch implements
those in the fixup table.
Takashi Iwai [Tue, 4 May 2021 09:18:02 +0000 (11:18 +0200)]
ALSA: hda/realtek: Fix speaker amp on HP Envy AiO 32
HP Envy AiO 32-a12xxx has an external amp that is controlled via GPIO
bit 0x04. However, unlike other devices, this amp seems to shut down
itself after the certain period, hence the OS needs to up/down the bit
dynamically only during the actual playback.
This patch adds the control of the GPIO bit via the existing pcm_hook
mechanism. Ideally it should be triggered at the actual stream start,
but we have only the state change at prepare/cleanup, so use those for
switching the GPIO bit on/off. This should be good enough for the
purpose, and was actually confirmed to work fine.
Takashi Iwai [Tue, 4 May 2021 08:20:57 +0000 (10:20 +0200)]
ALSA: hda/realtek: Fix silent headphone output on ASUS UX430UA
It was reported that the headphone output on ASUS UX430UA (SSID
1043:1740) with ALC295 codec is silent while the speaker works.
After the investigation, it turned out that the DAC assignment has to
be fixed on this machine; unlike others, it expects DAC 0x02 to be
assigned to the speaker pin 0x07 while DAC 0x03 to headphone pin
0x21.
This patch provides a fixup for the fixed DAC/pin mapping for this
device.
Timo Gurr [Mon, 3 May 2021 11:08:22 +0000 (13:08 +0200)]
ALSA: usb-audio: Add dB range mapping for Sennheiser Communications Headset PC 8
The decibel volume range contains a negative maximum value resulting in
pipewire complaining about the device and effectivly having no sound
output. The wrong values also resulted in the headset sounding muted
already at a mixer level of about ~25%.
Sami Loone [Sat, 1 May 2021 10:07:53 +0000 (12:07 +0200)]
ALSA: hda/realtek: ALC285 Thinkpad jack pin quirk is unreachable
In 9bbb94e57df1 ("ALSA: hda/realtek: fix static noise on ALC285 Lenovo
laptops") an existing Lenovo quirk was made more generic by removing a
0x12 pin requirement from the entry. This made the second chance table
Thinkpad jack entry unreachable as the pin configurations became
identical.
Revert the 0x12 pin requirement removal and move Thinkpad jack pin quirk
back to the primary pin table as they can co-exist when more specific
configurations come first.
Add a more targeted pin quirk for Lenovo devices that have 0x12 as
0x40000000.
p = subprocess.Popen(["aplay -t raw -D plughw:1,0 /dev/zero"], shell=True)
subprocess.call(["arecord -Dhw:1,0 --dump-hw-params"], shell=True)
subprocess.call(["arecord -Dhw:1,0 -fdat -d1 /dev/null"], shell=True)
p.kill()
Handling ACP global external interrupt enable register
causing this issue.
This register got updated wrongly when there is active
stream causing interrupts disabled for active stream.
Refactored code to handle enabling and disabling external interrupts.
Just re-order the cx5066_fixups[] entries for HP devices for avoiding
the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
ALSA: hda/realtek: Remove redundant entry for ALC861 Haier/Uniwill devices
The quirk entry for Uniwill ECS M31EI is with the PCI SSID device 0,
which means matching with all. That is, it's essentially equivalent
with SND_PCI_QUIRK_VENDOR(0x1584), which also matches with the
previous entry for Haier W18 applying the very same quirk.
Let's unify them with the single vendor-quirk entry.
Just re-order the alc662_fixup_tbl[] entries for Acer and ASUS devices
for avoiding the oversight of the duplicated or unapplied item in
future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Just re-order the alc269_fixup_tbl[] entries for FSC, Medion, Samsung
and Lemote devices for avoiding the oversight of the duplicated or
unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Just re-order the alc269_fixup_tbl[] entries for Lenovo devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
ALSA: hda/realtek: Re-order ALC269 Sony quirk table entries
Just re-order the alc269_fixup_tbl[] entries for Sony devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
ALSA: hda/realtek: Re-order ALC269 ASUS quirk table entries
Just re-order the alc269_fixup_tbl[] entries for ASUS devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Just re-order the alc269_fixup_tbl[] entries for Dell devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Just re-order the alc269_fixup_tbl[] entries for Acer devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
ALSA: hda/realtek: Re-order ALC269 HP quirk table entries
Just re-order the alc269_fixup_tbl[] entries for HP devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Formerly, some entries were grouped for the actual codec, but this
doesn't seem reasonable to keep in that way. So now we simply keep
the PCI SSID order for the whole.
Also Cc-to-stable for the further patch applications.
Just re-order the alc882_fixup_tbl[] entries for Clevo devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also, user lower hex letters in the entry.
Also Cc-to-stable for the further patch applications.
ALSA: hda/realtek: Re-order ALC882 Sony quirk table entries
Just re-order the alc882_fixup_tbl[] entries for Sony devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
Just re-order the alc882_fixup_tbl[] entries for Acer devices for
avoiding the oversight of the duplicated or unapplied item in future.
No functional changes.
Also Cc-to-stable for the further patch applications.
ALSA: usb-audio: Remove redundant assignment to len
Variable len is set to zero but this value is never read as it is
overwritten with a new value later on, hence it is a redundant
assignment and can be removed.
Cleans up the following clang-analyzer warning:
sound/usb/mixer.c:2713:3: warning: Value stored to 'len' is never read
[clang-analyzer-deadcode.DeadStores].
ALSA: hda/realtek: Add quirk for Intel Clevo PCx0Dx
This applies a SND_PCI_QUIRK(...) to the Clevo PCx0Dx barebones. This
fix enables audio output over the headset jack and ensures that a
microphone connected via the headset combo jack is correctly recognized
when pluged in.
[ Rearranged the list entries in a sorted order -- tiwai ]
Stefan Binding [Mon, 26 Apr 2021 16:37:49 +0000 (17:37 +0100)]
ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye
Cracking noises have been reported on the built-in speaker for certain
Bullseye platforms, when volume is > 80%.
This issue is caused by the specific combination of Codec and AMP in
this platform, and cannot be fixed by the AMP, so indead must be fixed
at codec level, by adding attenuation to the volume.
Tested on DELL Inspiron-3505, DELL Inspiron-3501, DELL Inspiron-3500
ALSA: sb: Fix two use after free in snd_sb_qsound_build
In snd_sb_qsound_build, snd_ctl_add(..,p->qsound_switch...) and
snd_ctl_add(..,p->qsound_space..) are called. But the second
arguments of snd_ctl_add() could be freed via snd_ctl_add_replace()
->snd_ctl_free_one(). After the error code is returned,
snd_sb_qsound_destroy(p) is called in __error branch.
But in snd_sb_qsound_destroy(), the freed p->qsound_switch and
p->qsound_space are still used by snd_ctl_remove().
My patch set p->qsound_switch and p->qsound_space to NULL if
snd_ctl_add() failed to avoid the uaf bugs. But these codes need
to further be improved with the code style.
Merge tag 'asoc-v5.13' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.13
A lot of changes here for quite a quiet release in subsystem terms -
there's been a lot of fixes and cleanups all over the subsystem both
from generic work and from people working on specific drivers.
- More cleanup and consolidation work in the core and the generic card
drivers from Morimoto-san.
- Lots of cppcheck fixes for Pierre-Louis Brossart.
- New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358
accessory detection, and Realtek RT1019, RT1316, RT711 and RT715.
ALSA: emu8000: Fix a use after free in snd_emu8000_create_mixer
Our code analyzer reported a uaf.
In snd_emu8000_create_mixer, the callee snd_ctl_add(..,emu->controls[i])
calls snd_ctl_add_replace(.., kcontrol,..). Inside snd_ctl_add_replace(),
if error happens, kcontrol will be freed by snd_ctl_free_one(kcontrol).
Then emu->controls[i] points to a freed memory, and the execution comes
to __error branch of snd_emu8000_create_mixer. The freed emu->controls[i]
is used in snd_ctl_remove(card, emu->controls[i]).
My patch set emu->controls[i] to NULL if snd_ctl_add() failed to avoid
the uaf.
Marco Felsch [Fri, 23 Apr 2021 13:54:02 +0000 (15:54 +0200)]
ASoC: max98088: fix ni clock divider calculation
The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk.
The max98088 datasheet [2] has no such formula but table-12 equals so
we can assume that it is the same for both devices.
Sami Loone [Sun, 25 Apr 2021 20:37:12 +0000 (22:37 +0200)]
ALSA: hda/realtek: fix static noise on ALC285 Lenovo laptops
Remove a duplicate vendor+subvendor pin fixup entry as one is masking
the other and making it unreachable. Consider the more specific newcomer
as a second chance instead.
The generic entry is made less strict to also match for laptops with
slightly different 0x12 pin configuration. Tested on Lenovo Yoga 6 (AMD)
where 0x12 is 0x40000000.
ALSA: usb-audio: Fix implicit sync clearance at stopping stream
The recent endpoint management change for implicit feedback mode added
a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at
snd_usb_endpoint_stop() to assure no leftover for the feedback from
the already stopped capture stream. This turned out to cause a
regression, however, when full-duplex streams were running and only a
capture was stopped. Because of the above clearance of ep->sync_sink
pointer, no more feedback is done, hence the playback will stall.
This patch fixes the ep->sync_sink clearance to be done only after all
endpoints are released, for addressing the regression.
Mark Brown [Fri, 23 Apr 2021 17:07:54 +0000 (18:07 +0100)]
ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2
A KernelCI bisection identified 59c35c44a9cf89 "ASoC: simple-card: add
simple_parse_node()" as causing simple-card to fail to instantiate on
kontron-sl28-var3-ads2 systems. Since the merge window is expected to
open over the weekend drop that commit and subsequent ones which depend
on it for now in case other systems are affected too.
<3>[ 10.191982] kobject_add_internal failed for (null)-wm8904-hifi with -EEXIST, don't try to register things with the same name in the same directory.
Shuming Fan [Thu, 22 Apr 2021 10:32:35 +0000 (18:32 +0800)]
ASoC: rt711-sdca: add the notification when volume changed
This patch adds the return value when the volume settings were changed.
The userspace application might monitor the kcontrols to check which control changed.
Shuming Fan [Thu, 22 Apr 2021 10:32:20 +0000 (18:32 +0800)]
ASoC: rt711-sdca: change capture switch controls
The DAPM event and mixer control could mute/unmute the capture directly.
That will be confused that capture still works if the user settings is unmute before the capture.
Therefore, this patch uses the variables to record the capture switch status of DAPM and mixer.
Instead of using the clk embedded in the clk_hw (which is meant to go
away), a clock provider which need to interact with its own clock should
request clk reference through the clock provider API.
Instead of using the clk embedded in the clk_hw (which is meant to go
away), a clock provider which need to interact with its own clock should
request clk reference through the clock provider API.
ALSA: usb-audio: Generic application of implicit fb to Roland/BOSS devices
Through the examinations and experiments with lots of Roland and BOSS
USB-audio devices, we found out that the recently introduced
full-duplex operations with the implicit feedback mode work fine for
quite a few devices, while the others need only the capture-side quirk
to enforce the full-duplex mode. The recent commit d86f43b17ed4
("ALSA: usb-audio: Add support for many Roland devices' implicit
feedback quirks") tried to add such quirk entries manually in the
lists, but this turned out to be too many and error-prone, hence it
was reverted again.
This patch is another attempt to cover those missing Roland/BOSS
devices but in a more generic way. It matches the devices with the
vendor ID 0x0582, and checks whether they are with both ASYNC sync
types or ASYNC is only for capture device. In the former case, it's
the device with the implicit feedback mode, and applies accordingly.
In both cases, the capture stream requires always the full-duplex
mode, and we apply the known capture quirk for that, too.
Basically the already existing BOSS device quirk entries become
redundant after this generic matching, so those are removed. Although
the capture_implicit_fb_quirks[] table became empty and superfluous, I
keep it for now, so that people can put a special device easily at any
time later again.
ASoC: tegra: mark runtime-pm functions as __maybe_unused
A reorganization of the driver source led to two of them causing
a compile time warning in some configurations:
tegra/tegra20_spdif.c:36:12: error: 'tegra20_spdif_runtime_resume' defined but not used [-Werror=unused-function]
36 | static int tegra20_spdif_runtime_resume(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra20_spdif.c:27:12: error: 'tegra20_spdif_runtime_suspend' defined but not used [-Werror=unused-function]
27 | static int tegra20_spdif_runtime_suspend(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra30_ahub.c:64:12: error: 'tegra30_ahub_runtime_resume' defined but not used [-Werror=unused-function]
64 | static int tegra30_ahub_runtime_resume(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra30_ahub.c:43:12: error: 'tegra30_ahub_runtime_suspend' defined but not used [-Werror=unused-function]
43 | static int tegra30_ahub_runtime_suspend(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
Mark these functions as __maybe_unused to avoid this kind of warning.
Fixes: b5571449e618 ("ASoC: tegra30: ahub: Remove handing of disabled runtime PM") Fixes: c53b396f0dd4 ("ASoC: tegra20: spdif: Remove handing of disabled runtime PM") Fixes: 80ec4a4cb36d ("ASoC: tegra20: i2s: Remove handing of disabled runtime PM") Fixes: b5f6f781fcb2 ("ASoC: tegra30: i2s: Remove handing of disabled runtime PM") Signed-off-by: Arnd Bergmann <[email protected]> Link: https://lore.kernel.org/r/[email protected] Signed-off-by: Mark Brown <[email protected]>
Configuring number of channels per LRCLK frame by using e.g.
snd_soc_dai_set_tdm_slot before configuring DAI format was being
overwritten by the latter due to a regmap_write which would write over
the whole register.
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