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[ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ]
According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has
the following bit field definitions:
| BITS | FIELD | RW | RESET | DEFINITION |
| 15 | RSVD | RO | 0x0 | Reserved |
| 14 | RSVD | RW | 0x1 | Reserved |
| 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode |
| 11:10 | RSVD | RO | 0x0 | Reserved |
| 9:8 | LBI_RESP | RW | 0x1 | Integrator Response |
| 7:6 | RSVD | RO | 0x0 | Reserved |
| 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode |
| 4:1 | RSVD | RO | 0x0 | Reserved |
| 0 | EN | RW | 0x0 | Enable/Disable AVC |
The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED. It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0. While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.
The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'. Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.
Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.
[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf
Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ]
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ]
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 87263968516fb9507d6215d53f44052627fae8d8 upstream.
When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt()
function rewrites the number of slots previously set by the
fsl_ssi_set_dai_tdm_slot() function to 2 by default.
To fix this, let's use the saved slot count value or, if TDM
is not used and the slot count is not set, proceed as before.
Fixes: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2bf44e0ee95f39cc54ea1b942f0a027e0181ca4e upstream.
Recently we found the micmute led init state is not correct after
freshly installing the ubuntu linux on a Lenovo AIO machine. The
internal mic is not muted, but the micmute led is on and led mode is
'follow mute'. If we mute internal mic, the led is keeping on, then
unmute the internal mic, the led is off. And from then on, the
micmute led will work correctly.
So the micmute led init state is not correct. The led is controlled
by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the
patch_realtek, the gpio data is set to 0x4 initially and the led is
on with this data. In the hda_generic, the led_value is set to
0 initially, suppose users set the 'capture switch' to on from
user space and the micmute led should change to be off with this
operation, but the check "if (val == spec->micmute_led.led_value)" in
the call_micmute_led_update() will skip the led setting.
To guarantee the led state will be set by the 1st time of changing
"Capture Switch", set -1 to the init led_value.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210312041408.3776-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 80cffd2468ddb850e678f17841fc356930b2304a upstream.
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4ec5b96775a88dd9b1c3ba1d23c43c478cab95a2 upstream.
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 06abcb18b3a021ba1a3f2020cbefb3ed04e59e72 upstream.
Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fec60c3bc5d1713db2727cdffc638d48f9c07dc3 upstream.
Dell AE515 sound bar (413c:a506) spews the error messages when the
driver tries to read the current sample frequency, hence it needs to
be on the list in snd_usb_get_sample_rate_quirk().
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211551
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304083021.2152-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5ff9dde42e8c72ed8102eb8cb62e03f9dc2103ab upstream.
When HD-audio bus receives unsolicited events during its system
suspend/resume (S3 and S4) phase, the controller driver may still try
to process events although the codec chips are already (or yet)
powered down. This might screw up the codec communication, resulting
in CORB/RIRB errors. Such events should be rather skipped, as the
codec chip status such as the jack status will be fully refreshed at
the system resume time.
Since we're tracking the system suspend/resume state in codec
power.power_state field, let's add the check in the common unsol event
handler entry point to filter out such events.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org> # 183ab39eb0ea: ALSA: hda: Initialize power_state
Link: https://lore.kernel.org/r/20210310112809.9215-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 28e96c1693ec1cdc963807611f8b5ad400431e82 upstream.
The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on
AMD controller (1022:1457)") introduced a few workarounds for the
recent AMD HD-audio controller, and one of them is the forced BATCH
PCM mode so that PulseAudio avoids the timer-based scheduling. This
was thought to cover for some badly working applications, but this
actually worsens for more others. In total, this wasn't a good idea
to enforce it.
This is a partial revert of the commit above for dropping the PCM
BATCH enforcement part to recover from the regression again.
Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210308160726.22930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit eea46a0879bcca23e15071f9968c0f6e6596e470 upstream.
The per_pin->work might be still floating at the suspend, and this may
hit the access to the hardware at an unexpected timing. Cancel the
work properly at the suspend callback for avoiding the buggy access.
Note that the bug doesn't trigger easily in the recent kernels since
the work is queued only when the repoll count is set, and usually it's
only at the resume callback, but it's still possible to hit in
theory.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fc7c5c208eb7bc2df3a9f4234f14eca250001cb6 upstream.
The microphone in the Plantronics C320-M headset will randomly
fail to initialize properly, at least when using Microsoft Teams.
Introducing a 20ms delay on the control messages appears to
resolve the issue.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065
Tested-by: Andreas Kempe <kempe@lysator.liu.se>
Signed-off-by: John Ernberg <john.ernberg@actia.se>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303181405.39835-1-john.ernberg@actia.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ]
Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing,
thus the heaset functions are not working.
Add quirk for this model to select the correct input map, jack-detect
options and channel map to enable jack sensing and headset microphone.
This device uses IN1 for its internal MIC and JD2 for jack-detect.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26a9630c72ebac7c564db305a6aee54a8edde70e ]
Currently the mask operation on variable conf is just 3 bits so
the switch statement case value of 8 is unreachable dead code.
The function daio_mgr_dao_init can be passed a 4 bit value,
function dao_rsc_init calls it with conf set to:
conf = (desc->msr & 0x7) | (desc->passthru << 3);
so clearly when desc->passthru is set to 1 then conf can be
at least 8.
Fix this by changing the mask to 0xf.
Fixes: 8cc72361481f ("ALSA: SB X-Fi driver merge")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210227001527.1077484-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 26af17722a07597d3e556eda92c6fce8d528bc9f upstream.
There is another MSI board (1462:cc34) that has dual Realtek codecs,
and we need to apply the existing quirk for fixing the conflicts of
Master control.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303142346.28182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 48698c973e6b4dde94d87cd1ded56d9436e9c97d upstream.
This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This
fixes the issue of the device not recognizing a pluged in microphone.
The device has both, a microphone only jack, and a speaker + microphone
combo jack. The combo jack already works. The microphone-only jack does
not recognize when a device is pluged in without this patch.
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/0eee6545-5169-ef08-6cfa-5def8cd48c86@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ]
The Acer One S1002 tablet is using an analog mic on IN1 and has
its jack-detect connected to JD2_IN4N, instead of using the default
IN3 for its internal mic and JD1_IN4P for jack-detect.
Note it is also using AIF2 instead of AIF1 which is somewhat unusual,
this is correctly advertised in the ACPI CHAN package, so the speakers
do work without the quirk.
Add a quirk for the mic and jack-detect settings.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ]
The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using
SSP2 (AIF1) and it mostly works with the defaults. But instead of using
DMIC1 it is using an analog mic on IN1, add a quirk for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ]
The Estar Beauty HD MID 7316R tablet almost fully works with out default
settings. The only problem is that it has only 1 speaker so any sounds
only playing on the right channel get lost.
Add a quirk for this model using the default settings + MONO_SPEAKER.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4841b8e6318a7f0ae57c4e5ec09032ea057c97a8 upstream.
Modify 0x20 index 7 bit 5 to 1, make the 0x15 EAPD the same as 0x14.
Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/e62c5058957f48d8b8953e97135ff108@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit de5bfae2fd962a9da99f56382305ec7966a604b9 ]
The correct mask is 0x1f8 (Bit 3-8), but due to missing BIT() 0xf (Bit
0-3) was set instead. This means setting of CPCAP_BIT_MIC1_RX_TIMESLOT0
(Bit 3) still worked (part of both masks). On the other hand the code
does not properly clear the other MIC timeslot bits. I think this
is not a problem, since they are probably initialized to 0 and not
touched by the driver anywhere else. But the mask also contains some
wrong bits, that will be cleared. Bit 0 (CPCAP_BIT_SMB_CDC) should be
safe, since the driver enforces it to be 0 anyways.
Bit 1-2 are CPCAP_BIT_FS_INV and CPCAP_BIT_CLK_INV. This means enabling
audio recording forces the codec into SND_SOC_DAIFMT_NB_NF mode, which
is obviously bad.
The bug probably remained undetected, because there are not many use
cases for routing microphone to the CPU on platforms using cpcap and
user base is small. I do remember having some issues with bad sound
quality when testing voice recording back when I wrote the driver.
It probably was this bug.
Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Reviewed-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20210123172945.3958622-1-sre@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 856fe64da84c95a1d415564b981ae3908eea2a76 ]
There are two issues with this code. The first error path forgot to set
the error code and instead returns success. The second error path
doesn't clean up.
Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X9NE/9nK9/TuxuL+@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit fb3c293b82c31a9a68fbcf4e7a45fadd8a47ea2b upstream.
The commit f274baa49be6 ("ALSA: usb-audio: Allow non-vmalloc buffer
for PCM buffers") introduced the mode to allocate coherent pages for
PCM buffers, and it used bus->controller device as its DMA device.
It turned out, however, that bus->sysdev is a more appropriate device
to be used for DMA mapping in HCD code.
This patch corrects the device reference accordingly.
Note that, on most platforms, both point to the very same device,
hence this patch doesn't change anything practically. But on
platforms like xhcd-plat hcd, the change becomes effective.
Fixes: f274baa49be6 ("ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210205144559.29555-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4961167bf7482944ca09a6f71263b9e47f949851 upstream.
We've got another report indicating a similar problem wrt the
power-saving behavior with VIA codec on Clevo machines. Let's apply
the existing workaround generically to all Clevo devices with VIA
codecs to cover all in once.
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1181330
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210126165603.11683-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ]
haswell machine board is missing pm_ops what prevents it from undergoing
suspend-resume procedure successfully. Assign default snd_soc_pm_ops so
this is no longer the case.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 67ea698c3950d10925be33c21ca49ffb64e21842 upstream.
It turned out that VIA codecs also mute the sound in the lowest mixer
level. Turn on the dac_min_mute flag to indicate the mute-as-minimum
in TLV like already done in Conexant and IDT codecs.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 217bfbb8b0bfa24619b11ab75c135fec99b99b20 upstream.
snd_seq_oss_synth_make_info() didn't check the error code from
snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy()
with the uninitialized string as the source, which may lead to the
access over the limit.
Add the proper error check for avoiding the failure.
Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7c22eeaff8565d9a8374f320238c251ca31480b upstream.
As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L,
the second multiplication in
ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes ff->rx_bytes[port] <= 16777.
Fixes: 19174295788de77d ("ALSA: fireface: add transaction support")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9f65df9c589f249435255da37a5dd11f1bc86f4d upstream.
As snd_fw_async_midi_port.consume_bytes is unsigned int, and
NSEC_PER_SEC is 1000000000L, the second multiplication in
port->consume_bytes * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes port->consume_bytes <= 16777.
Fixes: 531f471834227d03 ("ALSA: firewire-lib/firewire-tascam: localize async midi port")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f373a811fd9a69fc8bafb9bcb41d2cfa36c62665 upstream.
Return -ETIMEDOUT if the dsp boot times out instead of returning
success.
Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/X9NEvCzuN+IObnTN@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 671ee4db952449acde126965bf76817a3159040d upstream.
When the axg-tdm-interface was introduced, the backend DAI was marked as an
endpoint when DPCM was walking the DAPM graph to find a its BE.
It is no longer the case since this
commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks")
Because of this, when DPCM finds a BE it does everything it needs on the
DAIs but it won't power up the widgets between the FE and the BE if there
is no actual endpoint after the BE.
On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE.
It is only linked to the dummy codec since there no actual HW after it.
>From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM
decoder, which is a widget between the FE and BE is not powered up.
>From the user perspective, everything seems fine but no data is produced.
Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem.
Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks")
Cc: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20201217150812.3247405-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5c6679b5cb120f07652418524ab186ac47680b49 upstream.
A widget's "dirty" list_head, much like its "list" list_head, eventually
chains back to a list_head on the snd_soc_card itself. This means that
the list can stick around even after the widget (or all widgets) have
been freed. Currently, however, widgets that are in the dirty list when
freed remain there, corrupting the entire list and leading to memory
errors and undefined behavior when the list is next accessed or
modified.
I encountered this issue when a component failed to probe relatively
late in snd_soc_bind_card(), causing it to bail out and call
soc_cleanup_card_resources(), which eventually called
snd_soc_dapm_free() with widgets that were still dirty from when they'd
been added.
Fixes: db432b414e20 ("ASoC: Do DAPM power checks only for widgets changed since last run")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/f8b5f031d50122bf1a9bfc9cae046badf4a7a31a.1607822410.git.tommyhebb@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f86de9b1c0663b0a3ca2dcddec9aa910ff0fbf2c upstream.
Cannot adjust speaker's volume on Lenovo C940.
Applying the alc298_fixup_speaker_volume function can fix the issue.
[ Additional note: C940 has I2S amp for the speaker and this needs the
same initialization as Dell machines.
The patch was slightly modified so that the quirk entry is moved
next to the corresponding Dell quirk entry. -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/ea25b4e5c468491aa2e9d6cb1f2fced3@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 744a11abc56405c5a106e63da30a941b6d27f737 upstream.
The current kernel does not support the cx11970 codec chip.
Add a codec configuration item to kernel.
[ Minor coding style fix by tiwai ]
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201229035226.62120-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4bfd6247fa9164c8e193a55ef9c0ea3ee22f82d8 upstream.
Clevo W35xSS_370SS with VIA codec has had the runtime PM problem that
looses the power state of some nodes after the runtime resume. This
was worked around by disabling the default runtime PM via a denylist
entry. Since 5.10.x made the runtime PM applied (casually) even
though it's disabled in the denylist, this problem was revisited. The
result was that disabling power_save_node feature suffices for the
runtime PM problem.
This patch implements the disablement of power_save_node feature in
VIA codec for the device. It also drops the former denylist entry,
too, as the runtime PM should work in the codec side properly now.
Fixes: b529ef2464ad ("ALSA: hda: Add Clevo W35xSS_370SS to the power_save blacklist")
Reported-by: Christian Labisch <clnetbox@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210104153046.19993-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c06ccf3ebb7503706ea49fd248e709287ef385a3 upstream.
The calculation of in_cables and out_cables bitmaps are done with the
bit shift by the value from the descriptor, which is an arbitrary
value, and can lead to UBSAN shift-out-of-bounds warnings.
Fix it by filtering the bad descriptor values with the check of the
upper bound 0x10 (the cable bitmaps are 16 bits).
Reported-by: syzbot+92e45ae45543f89e8c88@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201223174557.10249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 618de0f4ef11acd8cf26902e65493d46cc20cc89 ]
The PCM hw_params core function tries to clear up the PCM buffer
before actually using for avoiding the information leak from the
previous usages or the usage before a new allocation. It performs the
memset() with runtime->dma_bytes, but this might still leave some
remaining bytes untouched; namely, the PCM buffer size is aligned in
page size for mmap, hence runtime->dma_bytes doesn't necessarily cover
all PCM buffer pages, and the remaining bytes are exposed via mmap.
This patch changes the memory clearance to cover the all buffer pages
if the stream is supposed to be mmap-ready (that guarantees that the
buffer size is aligned in page size).
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201218145625.2045-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4ebd47037027c4beae99680bff3b20fdee5d7c1e upstream.
The snd_seq_queue struct contains various flags in the bit fields.
Those are categorized to two different use cases, both of which are
protected by different spinlocks. That implies that there are still
potential risks of the bad operations for bit fields by concurrent
accesses.
For addressing the problem, this patch rearranges those flags to be
a standard bool instead of a bit field.
Reported-by: syzbot+63cbe31877bb80ef58f5@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/20201206083456.21110-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9df28edce7c6ab38050235f6f8b43dd7ccd01b6d upstream.
Some buggy firmware don't give the current sample rate but leaves
zero. Handle this case more gracefully without warning but just skip
the current rate verification from the next time.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201218145858.2357-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6ca653e3f73a1af0f30dbf9c2c79d2897074989f upstream.
The Quanta NL3 laptop has both a headphone output jack and a headset
jack, on the right edge of the chassis.
The pin information suggests that both of these are at the Front.
The PulseAudio is confused to differentiate them so one of the jack
can neither get the jack sense working nor the audio output.
The ALC269_FIXUP_LIFEBOOK chained with ALC269_FIXUP_QUANTA_MUTE can
help to differentiate 2 jacks and get the 'Auto-Mute Mode' working
correctly.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201222150459.9545-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 11cb881bf075cea41092a20236ba708b18e1dbb2 upstream.
There are a few places that call round{up|down}_pow_of_two() with the
value zero, and this causes undefined behavior warnings. Avoid
calling those macros if such a nonsense value is passed; it's a minor
optimization as well, as we handle it as either an error or a value to
be skipped, instead.
Reported-by: syzbot+33ef0b6639a8d2d42b4c@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201218161730.26596-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2506318e382c4c7daa77bdc48f80a0ee82804588 upstream.
It seems that the HD-audio clear and reconfig sysfs don't work any
longer after the recent driver core change. There are multiple issues
around that: the linked list corruption and the dead device handling.
The former issue is fixed by another patch for the driver core itself,
while the latter patch needs to be addressed in HD-audio side.
This patch corresponds to the latter, it recovers those broken
functions by replacing the device detach and attach actions with the
standard core API functions, which are almost equivalent with unbind
and bind actions.
Fixes: 654888327e9f ("driver core: Avoid binding drivers to dead devices")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=209207
Link: https://lore.kernel.org/r/20201209150119.7705-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 85a7555575a0e48f9b73db310d0d762a08a46d63 ]
The error handling frees "ctl" but it's still on the "dsp->ctl_list"
list so that could result in a use after free. Remove it from the list
before returning.
Fixes: 2323736dca72 ("ASoC: wm_adsp: Add basic support for rev 1 firmware file format")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/X9B0keV/02wrx9Xs@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
