| Commit message (Collapse) | Author | Age | Files | Lines |
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G14(GA401) series with ALC289
This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.
Fixes: ff53664daff2 ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289")
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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with ALC289
This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140616.298892-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.
Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Notebook Pen S
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Enable Vref2 under long term using PLL2 to avoid clock unstable.
Signed-off-by: derek.fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1594721600-29994-1-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Clear BIT6 of INPPGA means not muted (Switch On).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657025-4903-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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<pierre-louis.bossart@linux.intel.com>:
This is hopefully the last set of fixes to avoid probe errors due to
stricter checks of DAI capabilities introduced late in the 5.8 cycle.
Daniel Baluta (1):
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
Pierre-Louis Bossart (2):
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
ASoC: Intel: bdw-rt5677: fix non BE conversion
include/sound/soc-dai.h | 1 +
sound/soc/generic/audio-graph-card.c | 4 +--
sound/soc/generic/simple-card.c | 4 +--
sound/soc/intel/boards/bdw-rt5677.c | 1 +
sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++
sound/soc/sof/imx/imx8.c | 8 ++++++
sound/soc/sof/imx/imx8m.c | 8 ++++++
7 files changed, 60 insertions(+), 4 deletions(-)
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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This is identical with change for Intel platforms done with
commit 8c05246c0b58 ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell")
and fixes a regression on i.MX8/i.MX8M:
[ 25.705750] esai-Codec: ASoC: no backend playback stream
[ 27.923378] esai-Codec: ASoC: no users playback at close - state
This is root-caused to the introduction of the DAI capability checks
with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a
requirement for all DAIs to report at least a non-zero min_channels
field.
Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When SOF is used, the normal links are converted into DPCM ones. This
generates an error
[ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd
Wake on Voice does not support playback
[ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on
Voice :-22
Fix by forcing the capture direction.
Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Curtis Malainey <curtis@malainey.com>
Link: https://lore.kernel.org/r/20200707210439.115300-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a helper to walk through all the DAIs and set dpcm_playback and
dpcm_capture flags based on the DAIs capabilities, and use this helper
to avoid setting these flags arbitrarily in generic cards.
The commit referenced in the Fixes tag did not introduce the
configuration issue but will prevent the card from probing when
detecting invalid configurations.
Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Bossart <pierre-louis.bossart@linux.intel.com>:
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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we need to free all allocated tlvs, not just the one allocated in
the loop before releasing kcontrols - other the tlvs references will
leak.
Fixes: 9f90af3a995298 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.
remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.
Fixes: 7df04ea7a31ea ('ASoC: topology: modify dapm route loading routine and add dapm route unloading')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Bossart <pierre-louis.bossart@linux.intel.com>:
V2 with a number of cleanups:
split between I2C and SoundWire modes, as done for rt5682, and updated
Kconfigs.
removed useless initializations common to both modes
removed idle_bias on
fixed register classified as volatile in error
fixed SPDX comments
Pierre-Louis Bossart (2):
ASoC: codecs: max98373: split I2C and common parts
ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies
Ryan Lee (2):
ASoC: codecs: max98373: Removed superfluous volume control from chip
default
ASoC: codecs: max98373: add SoundWire support
randerwang (2):
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: Intel: sdw_max98373: add card_late_probe support
sound/soc/codecs/Kconfig | 20 +-
sound/soc/codecs/Makefile | 4 +
sound/soc/codecs/max98373-i2c.c | 612 +++++++++++++++
sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++++
sound/soc/codecs/max98373-sdw.h | 72 ++
sound/soc/codecs/max98373.c | 619 +--------------
sound/soc/codecs/max98373.h | 17 +-
sound/soc/intel/boards/Kconfig | 7 +-
sound/soc/intel/boards/sof_sdw.c | 19 +-
sound/soc/intel/boards/sof_sdw_common.h | 6 +
sound/soc/intel/boards/sof_sdw_max98373.c | 12 +
11 files changed, 1668 insertions(+), 607 deletions(-)
create mode 100644 sound/soc/codecs/max98373-i2c.c
create mode 100644 sound/soc/codecs/max98373-sdw.c
create mode 100644 sound/soc/codecs/max98373-sdw.h
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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Idle_bias_on is used to decide bias on/off in standby state by dapm.
When Idle_bias_on is set to one, dapm will keep max98373 active at
idle time. Max98373 is doing nothing in this state, so remove
idle_bias_on setting to let max98373 get suspended when it is idle.
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ryan Lee <ryans.lee@maximintegrated.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Volume control in probe function is not necessary.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DSP should be notified for device removal only if the
probe was successful. Fixes the following KASAN bug:
BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707204027.114169-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ASoC devm_ functions that register a component
(devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will
clean their component by running snd_soc_unregister_component.
snd_soc_unregister_component will then remove all the components for the
device that was used to register the component in the first place.
However, some drivers register several components (such as a DAI and a
dmaengine PCM) on the same device, and if the dmaengine PCM is registered
first, then the DAI will be cleaned up first and
snd_dmaengine_pcm_unregister will be called next.
snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component
on the device, and if there's one unregister that component and release its
dmaengine channels. That doesn't happen in practice though since the first
call to snd_soc_unregister_component removed all the components, so we
never get the chance to release the dmaengine channels.
In order to fix this, instead of removing all the components for a given
device, we can simply remove the component that was registered in the first
place. We should have the same number of component registration than we
have components, so it should work just fine.
Signed-off-by: Maxime Ripard <maxime@cerno.tech>
Link: https://lore.kernel.org/r/20200707074237.287171-1-maxime@cerno.tech
Signed-off-by: Mark Brown <broonie@kernel.org>
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S8 and S24 formats does not work on this machine driver so force to use
S16_LE instead.
In addition, add constraint to limit the max value of rate because the
rate higher than 96000(172000, 192000) is not stable either.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Link: https://lore.kernel.org/r/20200630091615.4020059-1-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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When snd_pci_acp3x driver loads we see:
WARNING kernel:snd_pci_acp3x 0000:04:00.5: Unbalanced pm_runtime_enable!
at boot time.
same can be observed in /var/log/messages/.
Modifying pm runtime sequence for fixing unbalanced pm issue.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200630092242.7799-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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soundwire mode
Base on https://patchwork.kernel.org/patch/11237953/
Soundwire mode also should follow it.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200701071645.32061-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200628155231.71089-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Miix 2 10
The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock.
Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is
actually used to enable the amplifier for these speakers
(the IRQ to the CPU comes directly from the jack-detect switch).
Add a quirk for having an ext speaker-amplifier enable pin on GPIO1
and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong
GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the
new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external
speaker-amplifier as necessary.
Also update the ident field for the dmi_system_id table entry, the
Miix models are not Thinkpads.
Fixes: 67e03ff3f32f ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage,
so the correct mask is 0x7 not 0x3.
Because of this wrong mask we were programming the ldo bits
to a setting of binary 001 (0x05 & 0x03) instead of binary 101
when moving to SND_SOC_BIAS_PREPARE.
According to the datasheet 001 is a reserved value, so no idea
what it did, since the driver was working fine before I guess we
got lucky and it does something which is ok.
Fixes: 5e8351de740d ("ASoC: add RT5670 CODEC driver")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The default mode for SSP configuration is TDM 4 slot and so far we were
using this for the bus format on cht-bsw-rt56732 boards.
One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2.
The second piggy-backed, output-only codec is inside the keyboard-dock
(which has extra speakers). Unlike the main rt5672 codec, we cannot
configure this codec, it is hard coded to use 2 channel 24 bit I2S.
Using 4 channel TDM leads to the dock speakers codec (which listens in on
the data send from the SSP to the rt5672 codec) emiting horribly distorted
sound.
Since we only support 2 channels anyways, there is no need for TDM on any
cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere.
This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing
the bus format set in cht_codec_fixup() to I2S 2 channel.
This change has been tested on the following devices with a rt5672 codec:
Lenovo Miix 2 10
Lenovo Thinkpad 8
Lenovo Thinkpad 10 (gen 1)
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When building on allyesconfig kernel for a NO_DMA=y platform (e.g.
Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n,
leading to a link failure:
sound/soc/qcom/common.o: In function `qcom_snd_parse_of':
common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port'
While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845
don't, so the following warning is seen:
WARNING: unmet direct dependencies detected for SND_SOC_QDSP6
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n]
Selected by [y]:
- SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y]
- SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y]
Until recently, this warning was harmless (from a compile-testing
point-of-view), but the new user of q6afe_is_rx_port() turned this into
a hard failure.
As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on
QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is
safe to increase compile testing coverage. Hence fix the link failure
by dropping the HAS_DMA dependency of SND_SOC_QDSP6.
Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver")
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add headset_jack for the intel reference board support with
10ec:1230.
Signed-off-by: PeiSen Hou <pshou@realtek.com.tw>
Link: https://lore.kernel.org/r/20200716090134.9811-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS UX563 speaker can't output.
Add quirk to link suitable model will enable it.
This model also could enable headset Mic.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/96dee3ab01a04c28a7b44061e88009dd@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS UX533 and UX534 speaker still can't output.
End User feedback speaker didn't have output.
Add this COEF value will enable it.
Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/80334402a93b48e385f8f4841b59ae09@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200713060421.62435-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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with ALC289
This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200711110557.18681-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS platform couldn't need to use Headset Mode model.
It changes to the suitable model.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d05bcff170784ec7bb35023407148161@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING
state, so set that for partially draining streams in
snd_compr_drain_notify() and use a flag for partially draining streams
While at it, add locks for stream state change in
snd_compr_drain_notify() as well.
Fixes: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)")
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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USB Audio analyzer RTX6001 uses the same implicit feedback quirk
as other XMOS-based devices.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/822f0f20-1886-6884-a6b2-d11c685cbafa@ivitera.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These devices claim to be 96kHz mono, but actually are 48kHz stereo with
swapped channels and unaligned transfers.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200702071433.237843-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Veriton N4660G desktop's audio (1025:1248) with ALC269VC cannot
detect the headset microphone until ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE
quirk maps the NID 0x18 as the headset mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-3-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Aspire C20-820 AIO's audio (1025:1065) with ALC269VC can't
detect the headset microphone until ALC269VC_FIXUP_ACER_HEADSET_MIC
quirk maps the NID 0x18 as the headset mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-2-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of
headset, the line out and internal speaker until
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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1)
In snd_hda_pick_fixup(), quirks are first matched by PCI SSID and then, if
there is no match, by codec SSID. The Lenovo "ThinkPad X1 Carbon 7th" has
an audio chip with PCI SSID 0x2292 and codec SSID 0x2293[1]. Therefore, fix
the quirk meant for that device to match on .subdevice == 0x2292.
2)
The "Thinkpad X1 Yoga 7th" does not exist. The companion product to the
Carbon 7th is the Yoga 4th. That device has an audio chip with PCI SSID
0x2292 and codec SSID 0x2292[2]. Given the behavior of
snd_hda_pick_fixup(), it is not possible to have a separate quirk for the
Yoga based on SSID. Therefore, merge the quirks meant for the Carbon and
Yoga. This preserves the current behavior for the Yoga.
[1] This is the case on my own machine and can also be checked here
https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701
[2]
https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355
Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen")
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Kailang Yang <kailang@realtek.com>
Tested-by: Vincent Bernat <vincent@bernat.ch>
Tested-by: Even Brenden <evenbrenden@gmail.com>
Signed-off-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200703080005.8942-2-benjamin.poirier@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HDMI codec driver has two debug traces printed from different
functions but with identical message content:
"HDMI: hinfo 000000006a6b84d9 not registered"
Fix this duplication and also add a bit more context in addition to raw
object pointer, to help analysis of kernel logs.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When HDMI PCM devices are opened in a specific order, with at least one
HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the
connected monitor, on recent Intel platforms (ICL/JSL and newer). While
this is not a typical sequence, at least Pulseaudio does this every time
when it is started, to discover the available PCMs.
The rootcause is an invalid assumption in hdmi_add_pin(), where the
total number of converters is assumed to be known at the time the
function is called. On older Intel platforms this held true, but after
ICL/JSL, the order how pins and converters are in the subnode list as
returned by snd_hda_get_sub_nodes(), was changed. As a result,
information for some converters was not stored to per_pin->mux_nids.
And this means some pins cannot be connected to all converters, and
application instead gets -EBUSY instead at open.
The assumption that converters are always before pins in the subnode
list, is not really a valid one. Fix the problem in hdmi_parse_codec()
by introducing separate loops for discovering converters and pins.
BugLink: https://github.com/thesofproject/linux/issues/1978
BugLink: https://github.com/thesofproject/linux/issues/2216
BugLink: https://github.com/thesofproject/linux/issues/2217
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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