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>
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>
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.
Cc: <stable@vger.kernel.org>
Fixes: fd6f4b0dc1 ("ALSA: bebob: Add skelton for BeBoB based devices")
Fixes: ec4dba5053 ("ALSA: oxfw: Add support for Behringer/Mackie devices")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 commit fixes hardcoded stream formats.
Cc: <stable@vger.kernel.org>
Fixes: 28b208f600 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi Mark, Guillaume
I'm so sorry to bother you again and again.
These are v2 of simple-card / audio-graph re-cleanup.
KernelCI had reported that below patches broke kontron-sl28-var3-ads2
sound card probing.
434392271a "ASoC: simple-card: add simple_link_init()"
59c35c44a9 "ASoC: simple-card: add simple_parse_node()"
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.
Link: https://lore.kernel.org/r/87cztzcq56.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87h7k0i437.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/20210423175318.13990-1-broonie@kernel.org
Link: https://lore.kernel.org/r/3ca62063-41b4-c25b-a7bc-8a8160e7b684@collabora.com
Kuninori Morimoto (4):
ASoC: simple-card: add simple_parse_node()
ASoC: simple-card: add simple_link_init()
ASoC: audio-graph: tidyup graph_dai_link_of_dpcm()
ASoC: audio-graph: tidyup graph_parse_node()
sound/soc/generic/audio-graph-card.c | 57 ++++-----
sound/soc/generic/simple-card.c | 168 +++++++++++++--------------
2 files changed, 112 insertions(+), 113 deletions(-)
--
2.25.1
cs42l42 does not support standard burst transfers so the use_single_read
and use_single_write flags must be set in the regmap config.
Because of this bug, the patch:
commit 0a0eb567e1 ("ASoC: cs42l42: Minor error paths fixups")
broke cs42l42 probe() because without the use_single_* flags it causes
regmap to issue a burst read.
However, the missing use_single_* could cause problems anyway because the
regmap cache can attempt burst transfers if these flags are not set.
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210511132855.27159-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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()
Reported-by: "kernelci.org bot" <bot@kernelci.org>
Fixes: 8859f809c7 ("ASoC: audio-graph: add graph_parse_node()")
Fixes: e51237b8d3 ("ASoC: audio-graph: add graph_link_init()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/87cztyawzr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Fixes: eb596e0fd1 ("ALSA: usb-audio: generate midi streaming substream names from jack names")
Link: https://lore.kernel.org/r/20210511090500.2637-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
BugLink: http://bugs.launchpad.net/bugs/1925057
Suggested-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210507024452.8300-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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)
BugLink: http://bugs.launchpad.net/bugs/804178
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210504091802.13200-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212933
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210504082057.6913-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
