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Pull sound fixes from Takashi Iwai:
"A bit more commits than expected at this time, but likely it's the
last shot before the final.
Many of changes are device-specific fix-ups for various ASoC drivers,
while a few usual HD-audio quirks and a FireWire fix, as well as a
couple of ALSA / ASoC core fixes.
All look nice and small, and nothing to scare much"
* tag 'sound-5.13-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: seq: Fix race of snd_seq_timer_open()
ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8
ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()
ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 840 Aero G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP EliteBook x360 1040 G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2
ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
ASoC: meson: gx-card: fix sound-dai dt schema
ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
ASoC: SOF: reset enabled_cores state at suspend
ASoC: fsl-asoc-card: Set .owner attribute when registering card.
ASoC: topology: Fix spelling mistake "vesion" -> "version"
ASoC: rt5659: Fix the lost powers for the HDA header
ASoC: core: Fix Null-point-dereference in fmt_single_name()
The timer instance per queue is exclusive, and snd_seq_timer_open()
should have managed the concurrent accesses. It looks as if it's
checking the already existing timer instance at the beginning, but
it's not right, because there is no protection, hence any later
concurrent call of snd_seq_timer_open() may override the timer
instance easily. This may result in UAF, as the leftover timer
instance can keep running while the queue itself gets closed, as
spotted by syzkaller recently.
For avoiding the race, add a proper check at the assignment of
tmr->timeri again, and return -EBUSY if it's been already registered.
Reported-by: syzbot+ddc1260a83ed1cbf6fb5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com
Link: https://lore.kernel.org/r/20210610152059.24633-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are 2 issues on this machine, the 1st one is mic's plug/unplug
can't be detected, that is because the mic is set to manual detecting
mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto
detecting mode. The other one is headphone's plug/unplug can't be
detected by pulseaudio, that is because the pulseaudio will use
ucm2/sof-hda-dsp on this machine, and the ucm2 only handle
'Headphone Jack', but on this machine the headphone's pincfg sets the
location to Front, then the alsa mixer name is "Front Headphone Jack"
instead of "Headphone Jack", so override the pincfg to change location
to Left.
BugLink: http://bugs.launchpad.net/bugs/1930188
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210608024600.6198-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the workqueue to queue wake-up event, isochronous context is not
processed, thus it's useless to check context for the workqueue to switch
status of runtime for PCM substream to XRUN. On the other hand, in
software IRQ context of 1394 OHCI, it's needed.
This commit fixes the bug introduced when tasklet was replaced with
workqueue.
Cc: <stable@vger.kernel.org>
Fixes: 2b3d2987d8 ("ALSA: firewire: Replace tasklet with work")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210605091054.68866-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull sound fixes from Takashi Iwai:
"A couple of small fixes are found in the ALSA core side at this time;
a fix in the new LED handling code and a long-standing (and likely no
one would notice) ioctl bug.
The rest are usual HD-audio fixes, mostly device-specific quirks but
also one major regression fix that was introduced in 5.13"
* tag 'sound-5.13-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: update the power_state during the direct-complete
ALSA: timer: Fix master timer notification
ALSA: control led: fix memory leak in snd_ctl_led_register
ALSA: hda: Fix for mute key LED for HP Pavilion 15-CK0xx
ALSA: hda/cirrus: Set Initial DMIC volume to -26 dB
ALSA: hda: Fix a regression in Capture Switch mixer read
ALSA: hda: Add AlderLake-M PCI ID
The patch_realtek.c needs to check if the power_state.event equals
PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and
resume() will be skipped if the codec is already rt_suspended, in this
case, the patch_realtek.c will always get PM_EVENT_ON even the system
is really resumed from S3.
We could set power_state to PMSG_SUSPEND in the prepare(), if other
PM functions are called before complete(), those functions will
override power_state; if no other PM functions are called before
complete(), we could know the suspend() and resume() are skipped since
only S3 pm functions could be skipped by direct-complete, in this case
set power_state to PMSG_RESUME in the complete(). This could guarantee
the first time of calling hda_codec_runtime_resume() after complete()
has the correct power_state.
Fixes: 215a22ed31 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210602145424.3132-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain
the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails
to decrease the refcount to zero, which causes device_release never to
be invoked. This leads to memory leak to some resources, like struct
device_private. In addition, we also free some other similar memory
leaks in snd_ctl_led_init/snd_ctl_led_exit.
Fix this by replacing device_del to device_unregister
in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit.
Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will
call kobject_release and delay the release of kobject, which will cause
use-after-free when the memory backing the kobject is freed at once.
Reported-by: syzbot+08a7d8b51ea048a74ffb@syzkaller.appspotmail.com
Fixes: a135dfb5de ("ALSA: led control - add sysfs kcontrol LED marking layer")
Signed-off-by: Dongliang Mu <mudongliangabcd@gmail.com>
Reviewed-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210602034136.2762497-1-mudongliangabcd@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit to drop the HDA-specific mute-LED control,
e65bf99718 ("ALSA: HDA - remove the custom implementation for the
audio LED trigger"), caused a regression on the mixer element read for
"Capture Switch" when it's built from bind controls. The function
create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly
via snd_hda_gen_add_kctl() instead of add_control(). Although the
commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in
add_control() as default, this code path fell out from the radar. As
a result, now the driver gives -EPERM error because of the lack of the
proper access bit at reading "Capture Switch" element value.
Fix the regression by setting the access bit properly.
Fixes: e65bf99718 ("ALSA: HDA - remove the custom implementation for the audio LED trigger")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634
Link: https://lore.kernel.org/r/20210531180633.27831-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull sound fixes from Takashi Iwai:
"A slightly high volume at this time due to pending ASoC fixes.
While there are a few generic simple-card fixes for regressions, most
of the changes are device-specific fixes: ASoC Intel SOF, codec
clocks, other codec / platform fixes as well as usual HD-audio and
USB-audio"
* tag 'sound-5.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (37 commits)
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook Fury 17 G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook Fury 15 G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook G8
ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
ALSA: usb-audio: scarlett2: snd_scarlett_gen2_controls_create() can be static
ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i
ALSA: hda/realtek: Headphone volume is controlled by Front mixer
ALSA: usb-audio: scarlett2: Improve driver startup messages
ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci
ALSA: usb-audio: fix control-request direction
ASoC: qcom: lpass-cpu: Use optional clk APIs
ASoC: cs35l33: fix an error code in probe()
ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware
ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency
ASoC: cs42l52: Minor tidy up of error paths
ASoC: cs35l32: Add missing regmap use_single config
ASoC: cs35l34: Add missing regmap use_single config
ASoC: cs42l73: Add missing regmap use_single config
ASoC: cs53l30: Add missing regmap use_single config
...
The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after
the DAPM widgets were added, and these powers were set by the JD settings
"RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function,
these powers were ignored to prevent them controlled by DAPM.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Message-Id: <15fced51977b458798ca4eebf03dafb9@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
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.
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1206
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210522034741.13415-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.
Use info level instead of error for notifications.
Display the VID/PID so device_setup is targeted to the right device.
Display "enabled" message to easily confirm that the driver is loaded.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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().
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: stable@vger.kernel.org # 5.10
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20210521133742.18098-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull sound fixes from Takashi Iwai:
"All small device-specific fixes here: a series of FireWire audio
fixes, UAF and other fixes in USB-audio and co spotted by fuzzer,
and a few HD-audio quirks as usual"
* tag 'sound-5.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: line6: Fix racy initialization of LINE6 MIDI
ALSA: dice: fix stream format for TC Electronic Konnekt Live at high sampling transfer frequency
ALSA: dice: disable double_pcm_frames mode for M-Audio Profire 610, 2626 and Avid M-Box 3 Pro
ALSA: intel8x0: Don't update period unless prepared
ALSA: hda/realtek: Add some CLOVE SSIDs of ALC293
ALSA: firewire-lib: fix amdtp_packet tracepoints event for packet_index field
ALSA: firewire-lib: fix calculation for size of IR context payload
ALSA: firewire-lib: fix check for the size of isochronous packet payload
ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro
ALSA: dice: fix stream format at middle sampling rate for Alesis iO 26
ALSA: hda/realtek: Add fixup for HP Spectre x360 15-df0xxx
ALSA: usb-audio: Fix potential out-of-bounce access in MIDI EP parser
ALSA: usb-audio: Validate MS endpoint descriptors
ALSA: hda: fixup headset for ASUS GU502 laptop
ALSA: hda/realtek: reset eapd coeff to default value for alc287
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.
Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com
Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com
Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo
Cc: Hyeonggon Yoo <42.hyeyoo@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210518083939.1927-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA dice driver detects jumbo payload at high sampling transfer frequency
for below models:
* Avid M-Box 3 Pro
* M-Audio Profire 610
* M-Audio Profire 2626
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518012510.37126-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Sergey Senozhatsky <senozhatsky@chromium.org>
Link: https://lore.kernel.org/r/s5h5yzi7uh0.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
