Geoffrey D. Bennett [Fri, 21 May 2021 08:20:12 +0000 (17:50 +0930)]
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().
Fixes: 93db51d06b32 ("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>
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.
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.
Cc: <stable@vger.kernel.org> Fixes: f1f0f330b1d0 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20210518012612.37268-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Cc: <stable@vger.kernel.org> Fixes: f11453c7cc01 ("ALSA: firewire-lib: use 16 bytes IR context header to separate CIP header") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20210513125652.110249-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Cc: <stable@vger.kernel.org> Fixes: fd6f4b0dc167 ("ALSA: bebob: Add skelton for BeBoB based devices") Fixes: ec4dba5053e1 ("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>
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 commit fixes hardcoded stream formats.
Cc: <stable@vger.kernel.org> Fixes: 28b208f600a3 ("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>
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.
Daniel Cordova A [Fri, 7 May 2021 17:31:16 +0000 (12:31 -0500)]
ALSA: hda: fixup headset for ASUS GU502 laptop
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.
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 ]
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com> Co-developed-by: Werner Sembach <wse@tuxedocomputers.com> Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210427153025.451118-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
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.
Fixes: 607184cb1635 ("ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button") Signed-off-by: Sami Loone <sami@loone.fi> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/YIXS+GT/dGI/LtK6@yoga Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <arnd@arndb.de> Link: https://lore.kernel.org/r/20210422133418.1757893-1-arnd@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
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.
ASoC: Intel: KMB: Fix random noise at the HDMI output
Random noise could be heard when playing audio to the HDMI output.
This is due to the IEC conversion is invoked in the external loop.
As a result, this additional loop takes up a lot of the processing
cycle.
hdmi_reformat_iec958() process the conversion using an internal loop,
it is safe to move it out from the external loop to avoid unnecessary
processing cycle been spent. Furthermore, ALSA IEC958 plugin works in
32bit format only.
"endpoint" and "port" are always exists, but there is no guarantee
for "ports". This patch checks "ports" if exists, otherwise,
it might set un-expected settings.
Colin Ian King [Tue, 20 Apr 2021 13:47:19 +0000 (14:47 +0100)]
ALSA: usb: midi: don't return -ENOMEM when usb_urb_ep_type_check fails
Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL)
the error return path returns -ENOMEM via the exit label "error". Other
uses of the same error exit label set the err variable to -ENOMEM but this
is not being used. I believe the original intent was for the error exit
path to return the value in err rather than the hard coded -ENOMEM, so
return this rather than the hard coded -ENOMEM.
ASoC: simple-card-utils: Allocate link info structure on heap
struct link_info can grow fairly large and may cause the stack frame
size to be exceeded when allocated on the stack. Some architectures
such as 32-bit ARM, RISC-V or PowerPC have small stack frames where
this causes a compiler warning, so allocate these structures on the
heap instead of the stack.
Fixes: 343e55e71877 ("ASoC: simple-card-utils: Increase maximum number of links to 128") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reported-by: Stephen Rothwell <sfr@canb.auug.org.au> Reported-by: kernel test robot <lkp@intel.com> Signed-off-by: Thierry Reding <treding@nvidia.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20210419164117.1422242-1-thierry.reding@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook 845 G8
On HP EliteBook 845 G8, the audio LEDs can be enabled by
ALC285_FIXUP_HP_MUTE_LED. So use it accordingly.
In addition to that, the mic captures lots of noises, so also limits the
mic boost. The quality of capture audio becomes crystal clear after
limiting the mic boost.
Mark Brown [Mon, 19 Apr 2021 17:18:45 +0000 (18:18 +0100)]
Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward@gatech.edu>:
The last two patches in this series fix a longstanding issue that prevented
the ALC3263 codec from using a headset mic. This codec can be found on Dell
systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343.
In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use
legacy HD Audio just to avoid this issue:
This may allow that ACPI quirk to be removed. Either way, the other systems
mentioned above do not support this quirk and already use the ASoC driver,
so this fix is necessary for headset mic support on those systems.
Note: there is likely other handling for this codec that only exists in the
HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824fa
("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that
does not seem to be resolved in the ASoC driver, to give an example.
Other patches in this series are not specific to the ALC3263. These patches
set the correct combo jack configuration when headphones are inserted, and
fix a misaligned value set in the DMIC2 Configuration Default register.
David Ward [Sun, 18 Apr 2021 13:46:58 +0000 (09:46 -0400)]
ASoC: rt286: Generalize support for ALC3263 codec
The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350
and Venue 11 Pro 7140. They require the same handling for the combo jack to
work with a headset: GPIO pin 6 must be set.
The HDA driver always sets this pin on the ALC3263, which it distinguishes
by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028
(Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to
determine if Dell is the system vendor.
David Ward [Sun, 18 Apr 2021 13:46:56 +0000 (09:46 -0400)]
ASoC: rt298: Configure combo jack for headphones
During jack detection, the combo jack is configured for a CTIA headset, and
then for an OMTP headset, while sensing the mic connection. If a mic is not
found in either case, the combo jack should be re-configured for headphones
only. This is consistent with the HDA driver behavior.
David Ward [Sun, 18 Apr 2021 13:46:55 +0000 (09:46 -0400)]
ASoC: rt286: Configure combo jack for headphones
During jack detection, the combo jack is configured for a CTIA headset, and
then for an OMTP headset, while sensing the mic connection. If a mic is not
found in either case, the combo jack should be re-configured for headphones
only. This is consistent with the HDA driver behavior.
ASoC: sigmadsp: Disable cache mechanism for readbacks
The ALSA control readback functionality only works for non-volatile
controls, i.e. control values that does not change on their own without
driver interaction.
This doesn't work for readbacks since the DSP firmware updates the
control value. Disable the cache mechanism in the driver if the control
name matches the prefix used for readbacks to ensure that the control
value is valid.
Mark Brown [Mon, 19 Apr 2021 15:51:43 +0000 (16:51 +0100)]
Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward@gatech.edu>:
The last two patches in this series fix a longstanding issue that prevented
the ALC3263 codec from using a headset mic. This codec can be found on Dell
systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343.
In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use
legacy HD Audio just to avoid this issue:
This may allow that ACPI quirk to be removed. Either way, the other systems
mentioned above do not support this quirk and already use the ASoC driver,
so this fix is necessary for headset mic support on those systems.
Note: there is likely other handling for this codec that only exists in the
HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824fa
("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that
does not seem to be resolved in the ASoC driver, to give an example.
Other patches in this series are not specific to the ALC3263. These patches
set the correct combo jack configuration when headphones are inserted, and
fix a misaligned value set in the DMIC2 Configuration Default register.
David Ward [Sun, 18 Apr 2021 13:46:54 +0000 (09:46 -0400)]
ASoC: rt286: Fix upper byte in DMIC2 configuration
This HDA verb sets the upper byte of the Configuration Default register, so
only an 8-bit value should be used. For the rt298, the same fix was applied
in commit f8f2dc4a7127 ("ASoC: rt298: fix wrong setting of gpio2_en").
ALSA: usb-audio: Re-apply implicit feedback mode to Pioneer devices
Pioneer devices are supposed to be working with the implicit feedback
mode, but so far the attempt to apply the implicit feedback caused
issues, hence we explicitly skipped the implicit feedback mode for
them. Recently, Geraldo discovered that the device actually works if
you skip the generic matching of the sync EPs for the capture stream.
That is, we should apply the implicit feedback setup for the playback
like other similar devices, while we need to return 1 from
audioformat_capture_quirk() so that no further matching will be done.
And, later on, Olivia reported later that the fiddling with the
capture quirk alone doesn't suffice for the test with speaker-test
program. This seems to be a similar case like the recently fixed BOSS
devices. Indeed, the problem could be addressed by setting
playback_first flag, which indicates that the playback URBs have to be
sent out at first even in the implicit feedback mode.
This patch implements the application of the implicit feedback to
Pioneer devices as described in the above. The former
skip_pioneer_sync_ep() was dropped, and instead we provide
is_pioneer_implicit_fb() to check the Pioneer devices that need the
implicit feedback. In the audioformat_implicit_fb_quirk(), simply
apply the implicit fb for playback and set chip->playback_first flag
if matching, and in audioformat_capture_quirk()(), it returns 1 for
skipping the generic EP sync handling.
sound/soc/codecs/rt5682.c:2404:42: style: Boolean result is used in
bitwise operation. Clarify expression with
parentheses. [clarifyCondition]
(pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST));
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210416191144.27006-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/lpass-rx-macro.c:1626:9: warning: Identical condition
and return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/codecs/lpass-rx-macro.c:1623:6: note: If condition 'ret' is
true, the function will return/exit
if (ret)
^
sound/soc/codecs/lpass-rx-macro.c:1626:9: note: Returning identical
expression 'ret'
return ret;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210416191144.27006-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/soc-core.c:2784:6: style: Condition '!num_routes' is always
false [knownConditionTrueFalse]
if (!num_routes) {
^
sound/soc/soc-core.c:2777:17: note: Assuming that condition
'num_routes<0' is not redundant
if (num_routes < 0 || num_routes & 1) {
^
sound/soc/soc-core.c:2783:2: note: Compound assignment '/=', assigned
value is 0
num_routes /= 2;
^
sound/soc/soc-core.c:2784:6: note: Condition '!num_routes' is always
false
if (!num_routes) {
^
The documentation for of_property_count_string reads
"
* Returns the number of strings on
* success, -EINVAL if the property does not exist, -ENODATA if property
* does not have a value, and -EILSEQ if the string is not null-terminated
* within the length of the property data.
"
Since the case for num_routes == 0 is not possible, let's remove this
test.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210416191144.27006-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/soc-core.c:2237:13: warning: %x in format string (no. 1)
requires 'unsigned int *' but the argument type is 'signed
int *'. [invalidScanfArgType_int]
} else if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
^
sound/soc/soc-core.c:2237:13: warning: %x in format string (no. 2)
requires 'unsigned int *' but the argument type is 'signed
int *'. [invalidScanfArgType_int]
} else if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210416191144.27006-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
value that is never used. [unreadVariable]
acpi_status status = AE_OK;
^
sound/soc/soc-acpi.c:37:21: style: Variable 'status' is assigned a Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210416191144.27006-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Add case statement to set sample-rate for the DJM-750 Pioneer
mixer. This was included as part of another patch but I think it has
been archived on Patchwork and hasn't been merged.
Luke D Jones [Mon, 19 Apr 2021 03:04:11 +0000 (15:04 +1200)]
ALSA: hda/realtek: GA503 use same quirks as GA401
The GA503 has almost exactly the same default setup as the GA401
model with the same issues. The GA401 quirks solve all the issues
so we will use the full quirk chain.
projects / linux-platform-drivers-x86.git / log