For PCM format the supported format is mono/stereo, S16_LE/S32_LE,
and 8KHz-192kHz frequency.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
commit 8edac489e7 upstream.
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: 9f90af3a99 ('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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6f0307df83 upstream.
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: 7df04ea7a3 ('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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b6aa06de77 upstream.
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: a212008925 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 85ca6b17e2 upstream.
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: 67e03ff3f3 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit b3df80ab6d ]
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5cacc6f576 upstream.
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: 5e8351de74 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 60379ba085 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 568e4e8212 upstream.
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f52 (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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 56275036d8 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
In SPDIF full duplex mode in order to get both RX and TX working we
need to start RX first, and TX second, fix this by porting SPDIF RX
configuration code from XCVR FW into Kernel driver and avoid using
M0+ core for SPDIF configuration.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 158ae2f5e6 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 473fbe13fd upstream.
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: 4e05110673 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f50a121d2f upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ff53664daf upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ef9ddb9dc4 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9b7e5208a9 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 68359a1ad8 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6e8a914ad6 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 73094608b8 ]
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Christoffer Nielsen <cn@obviux.dk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6f4ea2074d ]
Force it to use asynchronous playback.
Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f0b ("ALSA: usb-audio: Add quirk for Focusrite
Scarlett Solo").
This also seems to prevent regular clicks when playing at 44100Hz
on Scarlett 2i2 (2nd gen). I did not notice any side effects.
Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested.
Signed-off-by: Gregor Pintar <grpintar@gmail.com>
Reviewed-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fd60e0683e ]
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:16d8) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com>
Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d8695bc5b1 ]
A slight refactoring of the registration quirk code. Now it uses the
table lookup for easy additions in future. Also the return type was
changed to bool, and got a few more comments.
Link: https://lore.kernel.org/r/20200325103322.2508-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 55f7326170 ]
Create a quirk that allows special processing and/or
skipping the call to snd_card_register.
For HyperX AMP, which uses two interfaces, but only has
a capture stream in the second, this allows the capture
stream to merge with the first PCM.
Signed-off-by: Chris Wulff <crwulff@gmail.com>
Link: https://lore.kernel.org/r/20200314165449.4086-3-crwulff@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2edb84e304 ]
MicroBook IIc operates in UAC2 mode by default. This patch addresses
several issues with it:
- MicroBook II and IIc shares the same USB ID. We can distinguish them
by interface class.
- MaxPacketsOnly attribute is erroneously set in endpoint descriptors.
As a result this card produces noise with all sample rates other than
96 KHz. This also causes issues like IOMMU page faults and other
problems with host controller.
- Sample rate changes takes more than 2 seconds for this device. Clock
validity request returns false during that period, so the clock validity
quirk is required.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200229151815.14199-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The Channel Status bits must be reversed.
Fixes: 86401d592b ("MLK-23725-1: ASoC: fsl_xcvr: reimplement CS read function")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Add fixed sai mclk for PDM software decimation improve
libimxswpdm conversion algorithm performance for
voice applications by using exact division ratios for
clock configurations mclk, bitclk.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
There is no need for S24_LE support on TX since IEC958 ALSA
plugin is now configured to cover this function.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 781c90c034 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6e15d1261d upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8eae7e9b39 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9774dc218b upstream.
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/ThinkPadhttps://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701
[2]
https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPadhttps://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355
Fixes: d2cd795c4e ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Fixes: 54a6a7dc10 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e337bf19f6 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6a6ca7881b upstream.
We have a Dell AIO, there is neither internal speaker nor internal
mic, only a multi-function audio jack on it.
Users reported that after freshly installing the OS and plug
a headset to the audio jack, the headset can't output sound. I
reproduced this bug, at that moment, the Input Source is as below:
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Headphone Mic' 'Headset Mic'
Item0: 'Headphone Mic'
That is because the patch_realtek will set this audio jack as mic_in
mode if Input Source's value is hp_mic.
If it is not fresh installing, this issue will not happen since the
systemd will run alsactl restore -f /var/lib/alsa/asound.state, this
will set the 'Input Source' according to history value.
If there is internal speaker or internal mic, this issue will not
happen since there is valid sink/source in the pulseaudio, the PA will
set the 'Input Source' according to active_port.
To fix this issue, change the parser function to let the hs_mic be
stored ahead of hp_mic.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200625083833.11264-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ad155712bb upstream.
The stack object “info” in snd_opl3_ioctl() has a leaking problem.
It has 2 padding bytes which are not initialized and leaked via
“copy_to_user”.
Signed-off-by: xidongwang <wangxidong_97@163.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1594006058-30362-1-git-send-email-wangxidong_97@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit f79a732a83 ]
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: f44f2a5417 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add dsp mu interrupt into wakeup source when LPA mode.
So the interrupt will wakeup A core when DSP consumed
all audio bitstream data.
Signed-off-by: Bing Song <bing.song@nxp.com>
The buffer is share between A core and DSP. Enlarge it, so A core will
suspend longer when DSP is playbacking audio.
Signed-off-by: Bing Song <bing.song@nxp.com>
One big buffer shared between A core and DSP to buffer audio
bitstram. So A core can sleep more time when DSP is playing
buffered audio bit stream.
Signed-off-by: Bing Song <bing.song@nxp.com>
IMX8MP_CLK_AUDIO_ROOT and IMX8MP_CLK_AUDIO_AXI_DIV will gated when
suspend even if dsp driver ignored suspend when low power audio offload
playback. The root cause is the two clocks is controlled when audiomix
power on/off. Add the two clocks in dsp driver, so driver will add
one ref count when useing dsp, so the two clocks don't be gated when
A core suspend during low power audio playback.
Signed-off-by: Bing Song <bing.song@nxp.com>
This reverts commit aba41867dd which is
commit f0bd62b640 upstream.
It causes a number of reported issues and a fix for it has not hit
Linus's tree yet. Revert this to resolve those problems.
Cc: Alexander Tsoy <alexander@tsoy.me>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Sasha Levin <sashal@kernel.org>
Cc: Hans de Goede <jwrdegoede@fedoraproject.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Depending on sample rate the PDM must be clocked at either 24.576MHz or
22.5792MHz. CLK_DIV is later calculated as function of PDM root clock
frequency and FS. Setting PDM root clock = FS * 1024 is wrong because
for low sample rate such as 8000 the PDM root clock will be 8000 * 1024
= 8192000 Hz so PDM will be underclocked for this sample rate.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
commit b2c22910fe upstream.
There are two more HP systems control mute LED from HDA codec and need
to expose micmute led class so SoF can control micmute LED.
Add quirks to support them.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit adb36a8203 upstream.
These IDs are for upcoming NVIDIA chips with audio functions that are largely
similar to the existing ones.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit f141a42215 ]
Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.
Fixes: fc05a5b222 ("ASoC: rockchip: add support for pdm controller")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Reviewed-by: Heiko Stuebner <heiko@sntech.de>
Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ed1220df6e ]
For mono channel, SSI will switch to Normal mode.
In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).
The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.
So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.
Fixes: b0a7043d5c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a212008925 ]
Currently both FE and BE dai-links are configured bi-directional,
However the DSP BE dais are only single directional,
so set the directions as supported by the BE dais.
Fixes: c25e295cd7 (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: John Stultz <john.stultz@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a95737440 ]
This patch adds support to q6afe_is_rx_port() to get direction
of DSP BE dai port, this is useful for setting dailink
directions correctly.
Fixes: c25e295cd7 (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6476b60f32 ]
Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.
This patch fixes EOS handling!
Fixes: 68fd8480bb ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200611124159.20742-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 220345e98f upstream.
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a32a1fc998 upstream.
We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.
Signed-off-by: Chihhao Chen <chihhao.chen@mediatek.com>
Signed-off-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7585db1b0 upstream.
This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit ff58bbc7b9 ]
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Fixes: 10ce77e481 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 79d4f823a0 ]
The Asus T101HA uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like the Asus T100HA. Note unlike the T100HA its jack-detect is not
inverted.
Add a DMI quirk with the correct settings for this model.
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/20200608204634.93407-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 199a5e8fda ]
The Toshiba Encore WT10-A tablet almost fully works with the default
settings for Bay Trail CR devices. The only issue is that it uses a
digital mic. connected the the DMIC1 input instead of an analog mic.
Add a quirk for this model using the default settings with the input-map
replaced with BYT_RT5640_DMIC1_MAP.
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/20200608204634.93407-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ba4e5abc6c ]
With additional checks on dailinks, we see errors such as
[ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd
NoCodec-6 does not support playback
It's not clear why we set the dpcm_playback and dpcm_capture flags
unconditionally, add a check on number of channels for each direction
to avoid invalid configurations.
Fixes: 8017b8fd37 ('ASoC: SOF: Add Nocodec machine driver support')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 607fa205a7 ]
Additional checks for valid DAIs expose a corner case, where existing
BE dailinks get modified, e.g. HDMI links are tagged with
dpcm_capture=1 even if the DAIs are for playback.
This patch makes those changes conditional and flags configuration
issues when a BE dailink is has no_pcm=0 but dpcm_playback or
dpcm_capture=1 (which makes no sense).
As discussed on the alsa-devel mailing list, there are redundant flags
for dpcm_playback, dpcm_capture, playback_only, capture_only. This
will have to be cleaned-up in a future update. For now only correct
and flag problematic configurations.
Fixes: 218fe9b7ec ("ASoC: soc-core: Set dpcm_playback / dpcm_capture")
Suggested-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 36124fb19f ]
fsl_asrc_dma_hw_params() invokes dma_request_channel() or
fsl_asrc_get_dma_channel(), which returns a reference of the specified
dma_chan object to "pair->dma_chan[dir]" with increased refcnt.
The reference counting issue happens in one exception handling path of
fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End,
the function forgets to decrease the refcnt increased by
dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt
leak.
Fix this issue by calling dma_release_channel() when config DMA channel
failed.
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 25bf943e4e ]
Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()"
if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding
"pm_runtime_put()" into this error path.
Fixes: f65bb92ca1 ("ASoC: img-i2s-in: Add runtime PM")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0e0e10fde0 ]
The Toshiba Encore WT8-A tablet almost fully works with the default
settings for non-CR Bay Trail devices. The only problem is that its
jack-detect switch is not inverted (it is active high instead of
the normal active low).
Add a quirk for this model using the default settings +
BYT_RT5640_JD_NOT_INV.
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/20200518072416.5348-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 10ce77e481 ]
For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.
This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")
Fixes: c249177944 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series")
Signed-off-by: Erwin Burema <e.burema@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751
Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 03990fd58d ]
If an error occurs after the call to 'omap_mcbsp_init()', the reference to
'mcbsp->fclk' must be decremented, as already done in the remove function.
This can be achieved easily by using the devm_ variant of 'clk_get()'
when the reference is taken in 'omap_mcbsp_init()'
This fixes the leak in the probe and has the side effect to simplify both
the error handling path of 'omap_mcbsp_init()' and the remove function.
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Peter Ujfalusi <peter.ujflausi@ti.com>
Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4e8748fcae ]
There are 2 issues here:
- if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()'
is a no-op because the 'mop500_dai_links' structure has not been
initialized yet, so the referenced are not decremented
- The reference stored in 'mop500_dai_links[i].codecs' is refcounted
only once in the probe and must be decremented only once.
Fixes: 39013bd60e ("ASoC: Ux500: Dispose of device nodes correctly")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 66c705d07d ]
SSI BUSIF buffer is possible to overflow or underflow, especially in a
hypervisor environment. If there is no interrupt support, it will eventually
lead to errors in pcm data.
This patch adds overflow and underflow interrupt support for SSI BUSIF buffer.
Reported-by: Chen Li <licheng0822@thundersoft.com>
Signed-off-by: Yongbo Zhang <giraffesnn123@gmail.com>
Tested-by: Chen Li <licheng0822@thundersoft.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7d8785bc7a ]
Fix to return negative error code -ENOMEM from the IPC init error
handling case instead of 0, as done elsewhere in this function.
Fixes: c16211d622 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8304cf77c9 ]
Although the value of FDF is used just for outgoing stream, the assignment
to union member is done for both directions of stream. At present this
causes no issue because the value of same position is reassigned later for
opposite stream. However, it's better to add if statement.
Fixes: d3d10a4a1b ("ALSA: firewire-lib: use union for directional parameters")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a446873d7 ]
During S3 test, the following error was observed:
[ 726.174237] i2c_designware i2c_designware.0: platform_pm_resume+0x0/0x3d returned 0 after 0 usecs
[ 726.184187] max98373 i2c-MX98373:00: calling max98373_resume+0x0/0x30 [snd_soc_max98373] @ 12698, parent: i2c-11
[ 726.195589] max98373 i2c-MX98373:00: Reset command failed. (ret:-16)
When calling regmap_update_bits(), since map->reg_update_bits is NULL,
_regmap_read() is entered with the following logic:
if (!map->cache_bypass) {
ret = regcache_read(map, reg, val);
if (ret == 0)
return 0;
}
if (map->cache_only)
return -EBUSY;
regcache_read() hits -EINVAL because MAX98373_R2000_SW_RESET is volatile,
as map->cache_only is set by codec suspend, thus -EBUSY is returned.
Fix by moving max98373_reset() after cache_only set to false in max98373_resume().
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Link: https://lore.kernel.org/r/1588376661-29799-1-git-send-email-yong.zhi@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b6cc38f3f ]
The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.
Do delete the link at the right place inside the spinlock.
Fixes: 8fdff6a319 ("ALSA: snd-usb: implement new endpoint streaming model")
Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f0bd62b640 ]
For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.
But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).
This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3b8a299a58 ]
free_irq() is missing in case of error, fix that.
Signed-off-by: Pavel Machek (CIP) <pavel@denx.de>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200606153103.GA17905@amd
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c26fde3b15 ]
This provides a better separation between runtime and PM sleep
callbacks.
Only do nothing if given runtime flag is set and calback is not set.
With the current implementation, if PM sleep callback is set but runtime
callback is not set then at runtime resume we reload the firmware even
if we do not support runtime resume callback.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-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/20200515135958.17511-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7f0d5053c5 ]
The "header->number" comes from the ioctl and it needs to be clamped to
prevent out of bounds writes.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dbd1317978 ]
Currently mute LED and micmute LED share the same GPIO polarity.
So split the polarity for mute and micmute, in case they have different
polarities.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a697ae6ea5 ]
davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a
reference of the specified dma_chan object to "chan" with increased
refcnt.
When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes
invalid, so the refcount should be decreased to keep refcount balanced.
The reference counting issue happens in one exception handling path of
davinci_mcasp_get_dma_type(). When chan device is NULL, the function
forgets to decrease the refcnt increased by dma_request_chan(), causing
a refcnt leak.
Fix this issue by calling dma_release_channel() when chan device is
NULL.
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1fecbb71fe ]
Disable exception interrupt before scheduling tasklet, otherwise if
the tasklet isn't handled immediately, there will be endless xrun
interrupt.
Fixes: 7ccafa2b38 ("ASoC: fsl_esai: recover the channel swap after xrun")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.1587968824.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fe17e6cdc0 ]
when do randconfig like this:
CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y
CONFIG_SND_SOC_SOF_IMX8=y
CONFIG_SND_SOC_SOF_OF=y
CONFIG_IMX_DSP=m
CONFIG_IMX_SCU=y
there is a link error:
sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg':
imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell'
Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this
Fixes: f9ad754684 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3ef9d5073b ]
The microphone-jack state needs to be masked in a case of a 4-pin jack
when microphone and ground pins are shorted. Presence of nvidia,headset
tells that WM8903 CODEC driver should mask microphone's status if short
circuit is detected, i.e headphones are inserted.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Resume common flow (System PM / Runtime PM) is like this:
sof_resume
-> specific device resume
-> snd_sof_load_firmware
-> snd_sof_dsp_reset (1)
-> load_modules()
-> snd_sof_run_firmware (2)
We need to implement dsp_reset callback (1) that will actually reset the
DSP but keep it stalled.
In order to implement this we do the following:
-> put DSP into reset (assert CoreReset bit from PWRCTL)
-> stall the DSP using RunStall bit from AudioDSP_REG2 Audiomix
-> take DSP out of reset (de-assert CoreReset bit from PWRCTL)
At this moment the DSP is taken out of reset and Stalled! This means
that we can load the firmware and then start the DSP (2).
Until now we resetted the DSP by turning down the Audiomix PD. This
doesn't work for Runtime PM if another IP is keeping Audiomix PD up.
By introducing dsp_reset() we no longer rely on turning off the audiomix
to reset the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
This function is used to stall the DSP and it will be used
together with PWRCTL register in order to properly reset the
DSP to handle the Power Management use case.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Use a meaningful named macro to represent RunStall bit from
AudioDSP_REG2 register.
This register is used to Run/Stall the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 0c5086f569 upstream.
The HP Thunderbolt Dock has two separate USB devices, one is for speaker
and one is for headset. Add names for them so userspace can apply UCM
settings.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200608062630.10806-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 862b2509d1 upstream.
When a USB-audio interface gets runtime-suspended via auto-pm feature,
the driver suspends all functionality and increment
chip->num_suspended_intf. Later on, when the system gets suspended to
S3, the driver increments chip->num_suspended_intf again, skips the
device changes, and sets the card power state to
SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from
S3, the resume callback decrements chip->num_suspended_intf. Since
this refcount is still not zero (it's been runtime-suspended), the
whole resume is skipped. But there is a small pitfall here.
The problem is that the driver doesn't restore the card power state
after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So,
even after the system resume finishes, the card instance still appears
as if it were system-suspended, and this confuses many ioctl accesses
that are blocked unexpectedly.
In details, we have two issues behind the scene: one is that the card
power state is changed only when the refcount becomes zero, and
another is that the prior auto-suspend check is kept in a boolean
flag. Although the latter problem is almost negligible since the
auto-pm feature is imposed only on the primary interface, but this can
be a potential problem on the devices with multiple interfaces.
This patch addresses those issues by the following:
- Replace chip->autosuspended boolean flag with chip->system_suspend
counter
- At the first system-suspend, chip->num_suspended_intf is recorded to
chip->system_suspend
- At system-resume, the card power state is restored when the
chip->num_suspended_intf refcount reaches to chip->system_suspend,
i.e. the state returns to the auto-suspended
Also, the patch fixes yet another hidden problem by the code
refactoring along with the fixes above: namely, when some resume
procedure failed, the driver left chip->num_suspended_intf that was
already decreased, and it might lead to the refcount unbalance.
In the new code, the refcount decrement is done after the whole resume
procedure, and the problem is avoided as well.
Fixes: 0662292aec ("ALSA: usb-audio: Handle normal and auto-suspend equally")
Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e18035cf5c upstream.
Add and use snd_pcm_stream_lock_nested() in snd_pcm_link/unlink
implementation. The code is fine, but generates a lockdep complaint:
============================================
WARNING: possible recursive locking detected
5.7.1mq+ #381 Tainted: G O
--------------------------------------------
pulseaudio/4180 is trying to acquire lock:
ffff888402d6f508 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xda8/0xee0 [snd_pcm]
but task is already holding lock:
ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
other info that might help us debug this:
Possible unsafe locking scenario:
CPU0
----
lock(&group->lock);
lock(&group->lock);
*** DEADLOCK ***
May be due to missing lock nesting notation
2 locks held by pulseaudio/4180:
#0: ffffffffa1a05190 (snd_pcm_link_rwsem){++++}-{3:3}, at: snd_pcm_common_ioctl+0xca0/0xee0 [snd_pcm]
#1: ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
[...]
Cc: stable@vger.kernel.org
Fixes: f57f3df03a ("ALSA: pcm: More fine-grained PCM link locking")
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/37252c65941e58473b1219ca9fab03d48f47e3e3.1591610330.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit 573fcbfd31 upstream.
A couple of Lenovo ThinkCentre machines all have 2 front mics and they
use the same codec alc623 and have the same pin config, so add a
pintbl entry for those machines to apply the fixup
ALC283_FIXUP_HEADSET_MIC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200608115541.9531-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bbd6aac3ae upstream.
128000 and 192000 are congruence modulo 32000, thus it's wrong to
distinguish them as multiple of 32000 and 48000 by modulo 32000 at
first.
Additionally, used condition statement to detect quadruple speed can
cause missing bit flag.
Furthermore, counter to ensure the configuration is wrong and it
causes false positive.
This commit fixes the above three bugs.
Cc: <stable@vger.kernel.org>
Fixes: 60aec494b3 ("ALSA: fireface: support allocate_resources operation in latter protocol")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8ba4dc3cff upstream.
The xmax values for Master Playback Volume and Mic Boost
Capture Volume are specified incorrectly (one greater)
which results in the wrong dB gain being shown to the user
in the case of Master Playback Volume.
Signed-off-by: Pavel Dobias <dobias@2n.cz>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200515120757.24669-1-dobias@2n.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Similar with commit a4c2646bc92 ("MLK-24263-5 ASoC: SOF: Add support for system / runtime PM")
this adds the system/runtime PM handlers.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SOF boots the DSP at probe and keeps it up all the time.
With this change, after booting if no one is using the DSP
the SOF core will turn off the DSP to save power.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SOF core will not call device specific System suspend
if the DSP is not booted (this is mostly because the way
PCI handles PM, which before system suspend it activates
the devices calling runtime resume).
So, in case we use OF we need to make sure that a simple
system suspend/resume cycle where the DSP is already off
will not turn on the DSP.
This part was completely re-written in the upstream kernel,
and this change is just a summary of the changes in upstream.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We make use of common imx8m_suspend / imx8m_resume functions
for both system PM and runtime PM.
imx8m_suspend:
- frees the MU channels
- disables the clocks
imx8m_resume
- enables the clocks
- requests the MU channels
On i.MX8MP there is no dedicated functionality to put the DSP in reset.
The only way of doing this is to POWER DOWN the Audiomix domain.
We are able to do this because turning off the clocks and freeing the
channels makes the Audiomix to have no users thus PM kernel core turns
it down.
Now, care must be taken when handling system sleep. For this we
introduce a global variable to hold if the system was suspend or not via
system PM handlers.
SOF core will not call system PM suspend handler if the DSP is already down,
but at resume it will call the system PM resume. So, we need to keep track of
the state via the suspend variable.
Few insights on how SOF core handles the PM:
- SOF core uses PM runtime autosuspend (with a timeout of 2 secs)
- at probe, SOF core boots the DSP and lets the PM runtime suspend to
turn it off, if there is no activity
- when someone opens the ALSA sound card (aplay/arecord, etc) SOF core
calls PM runtime resume to turn on the DSP
- when the ALSA sound card is closed SOF core make use of PM subsystem
to call PM runtime suspend and thus turning off the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
This provides a better separation between runtime and PM sleep
callbacks.
Only do nothing if given runtime flag is set and calback is not set.
With the current implementation, if PM sleep callback is set but runtime
callback is not set then at runtime resume we reload the firmware even
if we do not support runtime resume callback.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SPDIF TX fails to play after resume because PHY is put in reset on suspend,
and is not re-initialized on resume. Move SPDIF TX configuration in "prepare"
callback since "prepare" is called after resume.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in, suspend/resume process, etc.
SD device plug in:
mmc1: new high speed SDHC card at address aaaa
mmcblk1: mmc1:aaaa SA16G 14.8 GiB
imx-cs42888 sound-cs42888: failed to find codec platform device
USB device plug in:
usb 2-1: USB disconnect, device number 2
usb 3-1: new SuperSpeed Gen 1 USB device number 2 using cdns-usb3
usb-storage 3-1:1.0: USB Mass Storage device detected
scsi host0: usb-storage 3-1:1.0
imx-cs42888 sound-cs42888: failed to find codec platform device
scsi 0:0:0:0: Direct-Access SanDisk Ultra 1.00 PQ: 0 ANSI: 6
sd 0:0:0:0: [sda] 60062500 512-byte logical blocks: (30.8 GB/28.6 GiB)
sd 0:0:0:0: [sda] Write Protect is off
sd 0:0:0:0: [sda] Write cache: disabled, read cache: enabled, doesn't support DPO or FUA
sda: sda1
sd 0:0:0:0: [sda] Attached SCSI removable disk
imx-cs42888 sound-cs42888: failed to find codec platform device
Suspend/resume process:
OOM killer disabled.
Freezing remaining freezable tasks ... (elapsed 0.083 seconds) done.
printk: Suspending console(s) (use no_console_suspend to debug)
fec 5b040000.ethernet eth0: Link is Down
PM: suspend devices took 0.684 seconds
Disabling non-boot CPUs ...
CPU1: shutdown
psci: CPU1 killed (polled 4 ms)
CPU2: shutdown
psci: CPU2 killed (polled 0 ms)
CPU3: shutdown
psci: CPU3 killed (polled 0 ms)
Enabling non-boot CPUs ...
Detected VIPT I-cache on CPU1
GICv3: CPU1: found redistributor 1 region 0:0x0000000051b20000
CPU1: Booted secondary processor 0x0000000001 [0x410fd042]
CPU1 is up
Detected VIPT I-cache on CPU2
GICv3: CPU2: found redistributor 2 region 0:0x0000000051b40000
CPU2: Booted secondary processor 0x0000000002 [0x410fd042]
CPU2 is up
Detected VIPT I-cache on CPU3
GICv3: CPU3: found redistributor 3 region 0:0x0000000051b60000
CPU3: Booted secondary processor 0x0000000003 [0x410fd042]
CPU3 is up
imx-cs42888 sound-cs42888: failed to find codec platform device
PM: resume devices took 0.240 seconds
OOM killer enabled.
Restarting tasks ...
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
[ Upstream commit d745cc1ab6 ]
Those strings are exposed to the user space as the
card name thus used in the GUIs. The common
standard is to avoid '_' here. The worst case
is 'sof-skl_hda_card' string.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191028164624.14334-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Compute mclk to allow low power and performance settings
based on sample_rate:
[ 8000 - 11025] low power mode
[16000 - 64000] performance mode
Update Copyright year.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
Add DSD_U8, DSD_U16_LE and DSD_U32_LE format constrains
in order to use SAI DSD settings for handling multi
channel capture streams. One channel per dataline.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
Add dummy codec dai DSD_U8, DSD_U16_LE and DSD_U32_LE
format support to allow machine ASoC drivers that
support DSD formats to use snd-soc-dummy-dai
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
[ Upstream commit 7f5ad9c900 ]
Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4020d1ccbe ]
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 399c01aa49 ]
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 54947cd64c ("ALSA: hda/realtek - Fix
speaker output regression on Thinkpad T570"). Essentially it fixes
the DAC / pin pairing by a static table. It was confirmed and merged
to stable kernel later.
Now, interestingly, we got another regression report for the very same
model (T570) about the similar problem, and the commit above was the
culprit. That is, by some reason, there are devices that prefer the
DAC1, and another device DAC2!
Unfortunately those have the same ID and we have no idea what can
differentiate, in this patch, a new fixup model "tpt470-dock-fix" is
provided, so that users with such a machine can apply it manually.
When model=tpt470-dock-fix option is passed to snd-hda-intel module,
it avoids the fixed DAC pairing and the DAC1 is assigned to the
speaker like the earlier versions.
Fixes: 54947cd64c ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570")
BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fb8cd6481f ]
The "info.index" variable can be 31 in "1 << info.index".
This might trigger an undefined behavior since 1 is signed.
Fix this by casting 1 to 1u just to be sure "1u << 31" is defined.
Signed-off-by: Changming Liu <liu.changm@northeastern.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4548.namprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit da7a8f1a8f ]
This is another TRX40 based motherboard with ALC1220-VB USB-audio
that requires a static mapping table.
This motherboard also has a PCI device which advertises no codecs. The
PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD
vendor ID, don't blacklist for now in case other boards have a working
audio device with the same ssid.
alsa-info.sh report for this board:
http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c
Signed-off-by: Andrew Oakley <andrew@adoakley.name>
Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 259eb82475 upstream.
A few known Clevo machines (PC50, PC70, X170) with ALC1220 codec need
the existing quirk for pins for PB51 and co.
Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200519065012.13119-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d9e8fe0cff upstream.
The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec
requires a similar workaround for Clevo laptops to enforce the
DAC/mixer connection path. Set up a quirk entry for that.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7513c5786 upstream.
There is a corner case that ALSA keeps increasing the hw_ptr but DMA
already stop working/updating the position for a long time.
In following log we can see the position returned from DMA driver does
not move at all but the hw_ptr got increased at some point of time so
snd_pcm_avail() will return a large number which seems to be a buffer
underrun event from user space program point of view. The program
thinks there is space in the buffer and fill more data.
[ 418.510086] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368
[ 418.510149] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6910 avail 9554
...
[ 418.681052] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15102 avail 1362
[ 418.681130] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
[ 418.726515] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 16464 avail 16368
This is because the hw_base will be increased by runtime->buffer_size
frames unconditionally if the hw_ptr is not updated for over half of
buffer time. As the hw_base increases, so does the hw_ptr increased
by the same number.
The avail value returned from snd_pcm_avail() could exceed the limit
(buffer_size) easily becase the hw_ptr itself got increased by same
buffer_size samples when the corner case happens. In following log,
the buffer_size is 16368 samples but the avail is 21810 samples so
CRAS server complains about it.
[ 418.851755] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 27390 avail 5442
[ 418.926491] sound pcmC0D5p: pos 96 hw_ptr 32832 appl_ptr 27390 avail 21810
cras_server[1907]: pcm_avail returned frames larger than buf_size:
sof-glkda7219max: :0,5: 21810 > 16368
By updating runtime->hw_ptr_jiffies each time the HWSYNC is called,
the hw_base will keep the same when buffer stall happens at long as
the interval between each HWSYNC call is shorter than half of buffer
time.
Following is a log captured by a patched kernel. The hw_base/hw_ptr
value is fixed in this corner case and user space program should be
aware of the buffer stall and handle it.
[ 293.525543] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368
[ 293.525606] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6880 avail 9584
[ 293.525975] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 10976 avail 5488
[ 293.611178] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15072 avail 1392
[ 293.696429] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
...
[ 381.139517] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1589776238-23877-1-git-send-email-brent.lu@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b0cb099062 upstream.
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit
e8a91ae18b ("ALSA: ice1712: Add support for STAudio ADCIII") we
enabled the ADCIII ports using the model=staudio option but forgot
this part to ensure the STDSP24 card is initialized properly.
Fixes: e8a91ae18b ("ALSA: ice1712: Add support for STAudio ADCIII")
Signed-off-by: Scott Bahling <sbahling@suse.com>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934
Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9e43342b46 ]
The ASUS laptop GL503VM with ALC295 can't detect the headset microphone.
The headset microphone does not work until pin 0x19 is enabled for it.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Link: https://lore.kernel.org/r/20200512061525.133985-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 14425f1f52 ]
Some models of the Samsung Notebook 9 have very quiet and distorted
headphone output. This quirk changes the VREF value of the ALC298
codec NID 0x1a from default HIZ to new 100.
[ adjusted to 5.7-base and rearranged in SSID order -- tiwai ]
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Link: https://lore.kernel.org/r/20200510032838.1989130-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 24164f434d ]
HP new platform has new mute led feature.
COEF index 0x34 bit 5 to control playback mute led.
COEF index 0x35 bit 2 and bit 3 to control Mic mute led.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 431e76c3ed ]
HP Note Book supported new mute Led.
Hardware PIN was not enough to meet old LED rule.
JD2 to control playback mute led.
GPO3 to control capture mute led.
(ALC285 didn't control GPO3 via verb command)
This two PIN just could control by COEF registers.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a462be52f ]
In the commit 8e85def572 ("ALSA: hda: enable regmap internal
locking"), we re-enabled the regmap lock due to the reported
regression that showed the possible concurrent accesses. It was a
temporary workaround, and there are still a few opened races even
after the revert. In this patch, we cover those still opened windows
with a proper mutex lock and disable the regmap internal lock again.
First off, the patch introduces a new snd_hdac_device.regmap_lock
mutex that is applied for each snd_hdac_regmap_*() call, including
read, write and update helpers. The mutex is applied carefully so
that it won't block the self-power-up procedure in the helper
function. Also, this assures the protection for the accesses without
regmap, too.
The snd_hdac_regmap_update_raw() is refactored to use the standard
regmap_update_bits_check() function instead of the open-code. The
non-regmap case is still open-coded but it's an easy part. The all
read and write operations are in the single mutex protection, so it's
now race-free.
In addition, a couple of new helper functions are added:
snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both
are called from HD-audio legacy driver. The former is to initialize
the given verb bits but only once when it's not initialized yet. Due
to this condition, the function invokes regcache_cache_only(), and
it's now performed inside the regmap_lock (formerly it was racy) too.
The latter function is for simply invoking regcache_sync() inside the
regmap_lock, which is called from the codec resume call path.
Along with that, the HD-audio codec driver code is slightly modified /
simplified to adapt those new functions.
And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are
rewritten with the helper macro. It's just for simplification because
the code logic is identical among all those functions.
Tested-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200109090104.26073-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8a71821f12 ]
GCC reports the following warning with W=1
sound/pci/hda/patch_realtek.c: In function ‘alc269_suspend’:
sound/pci/hda/patch_realtek.c:3616:29: warning: suggest braces around
empty body in an ‘if’ statement [-Wempty-body]
3616 | alc5505_dsp_suspend(codec);
| ^
sound/pci/hda/patch_realtek.c: In function ‘alc269_resume’:
sound/pci/hda/patch_realtek.c:3651:28: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
3651 | alc5505_dsp_resume(codec);
| ^
This is a classic macro problem and can indeed lead to bad program
flows.
Fix by using the usual "do { } while (0)" pattern
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The original patch updates REG_MICFIL_STAT in fsl_micfil_reset
method, but not in configure_hwvad_interrupts, fix this.
Fixes: e12b0060cd ("MLK-22598: ASoC: fsl_micfil: Reset channel output
data Flag")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
During probe SOF driver loads the topology, which will send messages
to DSP to create the Audio components.
SOF DAI component at creation time calls probe() and touches DAI
registers thus we need to make sure DAI clocks are turned on during
probe.
Using pm_runtime_get_sync() we force resume to be called which will
enable all necessary clocks.
Later we call the pm_runtime_put_sync() and mark the device as
autosuspend. This will cause suspend to be called which will disable
all clocks to save power.
Later, when userspace uses the sound card (via aplay/arecord) or any
alsa-lib calls the normal runtime resume/suspend flow will happen.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
in_filled_len need to be reset on suspend, for suspend, every state
will go back to initilization state.
Fixes: 3e87108614d1 ("MLK-24044: ASoC: fsl_easrc_m2m: Fix output length not accurate")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
initilize the m2m->complete in open() to avoid the NULL pointer in suspend
because the suspend can be called before initilizing m2m->complete in convert
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
commit c1f6e3c818 upstream.
The rawmidi core allows user to resize the runtime buffer via ioctl,
and this may lead to UAF when performed during concurrent reads or
writes: the read/write functions unlock the runtime lock temporarily
during copying form/to user-space, and that's the race window.
This patch fixes the hole by introducing a reference counter for the
runtime buffer read/write access and returns -EBUSY error when the
resize is performed concurrently against read/write.
Note that the ref count field is a simple integer instead of
refcount_t here, since the all contexts accessing the buffer is
basically protected with a spinlock, hence we need no expensive atomic
ops. Also, note that this busy check is needed only against read /
write functions, and not in receive/transmit callbacks; the race can
happen only at the spinlock hole mentioned in the above, while the
whole function is protected for receive / transmit callbacks.
Reported-by: butt3rflyh4ck <butterflyhuangxx@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAFcO6XMWpUVK_yzzCpp8_XP7+=oUpQvuBeCbMffEDkpe8jWrfg@mail.gmail.com
Link: https://lore.kernel.org/r/s5heerw3r5z.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b94e59d30 upstream.
ASUS ZenBook UX431DA requires an additional COEF setup when booted
from the recent Windows 10, otherwise it produces the noisy output.
The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly
due to the pop noise reduction.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207553
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200512073203.14091-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b590b38ca3 upstream.
Lenovo Thinkpad T530 seems to have a sensitive internal mic capture
that needs to limit the mic boost like a few other Thinkpad models.
Although we may change the quirk for ALC269_FIXUP_LENOVO_DOCK, this
hits way too many other laptop models, so let's add a new fixup model
that limits the internal mic boost on top of the existing quirk and
apply to only T530.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1171293
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200514160533.10337-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 1034872123 ]
The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current
printk format for the event includes 'sizeof(u8)' macro expected to be
extended in compilation time. However, this is not done. As a result,
perf tools cannot parse the event for printing:
$ mount -l -t debugfs
debugfs on /sys/kernel/debug type debugfs (rw,nosuid,nodev,noexec,relatime)
$ cat /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/format
...
print fmt: "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s",
REC->second, REC->cycle, REC->src, REC->dest, REC->channel,
REC->payload_quadlets, REC->data_blocks, REC->data_block_counter,
REC->packet_index, REC->irq, REC->index,
__print_array(__get_dynamic_array(cip_header),
__get_dynamic_array_len(cip_header),
sizeof(u8))
$ sudo perf record -e snd_firewire_lib:amdtp_packet
[snd_firewire_lib:amdtp_packet] function sizeof not defined
Error: expected type 5 but read 0
This commit fixes it by obsoleting the macro with actual size.
Cc: <stable@vger.kernel.org>
Fixes: bde2bbdb30 ("ALSA: firewire-lib: use dynamic array for CIP header of tracing events")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200503045718.86337-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 52e4e36807 ]
Commit 317d931392 ("ALSA: hda/realtek - Set default power save node to
0") makes the ALC225 have pop noise on S3 resume and cold boot.
The previous fix enable power save node universally for ALC225, however
it makes some ALC225 systems unable to produce any sound.
So let's only enable power save node for the affected Dell Wyse
platform.
Fixes: 317d931392 ("ALSA: hda/realtek - Set default power save node to 0")
BugLink: https://bugs.launchpad.net/bugs/1866357
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ca76282b6f ]
A race exists between build_pcms() and build_controls() phases of codec
setup. Build_pcms() sets up notifier for jack events. If a monitor event
is received before build_controls() is run, the initial jack state is
lost and never reported via mixer controls.
The problem can be hit at least with SOF as the controller driver. SOF
calls snd_hda_codec_build_controls() in its workqueue-based probe and
this can be delayed enough to hit the race condition.
Fix the issue by invalidating the per-pin ELD information when
build_controls() is called. The existing call to hdmi_present_sense()
will update the ELD contents. This ensures initial monitor state is
correctly reflected via mixer controls.
BugLink: https://github.com/thesofproject/linux/issues/1687
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add fsl,imx-audio-ak5552 to imx-ak5558 machine ASoC
driver, limit actual number of channels for ak5552
ADC support
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
We need to accumulate the input data length for calculating the
output length because there are data need to be prefilled in
the beginning. If the fraction part beyond the integrity of
sample is not considered in calculating, which may cause the
output task timeout.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
At SPDIF TX interrupt the firmware running on M0+ core
overwrites TX DPATH BYPASS FEM bit which turns the
expected PHY DMAC TX PLL clock 2*bitclock instead of
expected 10*bitclock if SAI PLL is used as PHY DMAC TX
PLL.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
The existing "shutdown" callback does not consider
full duplex mode, fix this.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Update drivers constrains, limit sample rates to
[8Khz - 64Khz] and up to 8 channels.
Use SND_SOC_DAIFMT_PDM format type and remove
tdm slot settings to allow capture from multiple
SAI_RxD pins.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
The criteria used to drop eARC mode into ARC mode
makes eARC function to fail. Drop it for now.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
With route "Playback" -> "Playback", dapm will enable
the "Playback" widget to be power on, there is this same
widget for codec component, then codec will enter
the SND_SOC_BIAS_ON level, then mclk is enabled
When the mclk is bound with a power domain, the power
domain will be enabled always.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
On imx8mp, there is no dedicated reset controller for DSP and
there is no dedicated power domain for DSP.
The power of DSP is bound with audiomix, so we need to check
the DSP status for judging the audiomix is reset or not.
We use the PID register for status check, after reset, it will
be set to zero, when DSP is used we set it to 1.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Fix the error handler in probe. otherwise the resource
can't be release when error happen.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
commit 977dfef40c upstream.
The commit 3c6fd1f07e ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
Since the empty codec problem appear on the certain AMD platform (PCI
ID 1022:1487), this patch changes the blacklist matching to both PCI
ID and SSID using pci_match_id(). Also, the entry that was removed by
the previous fix for ASUS ROG Zenigh II is re-added.
Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 54cb622168 ]
Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of
a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2
- the same logic as in rsnd_ssi_start. The attempt to disable these SSIs
was harmless, but caused a "status check failed" message to be printed
for every SSI in the multi-SSI setup.
The disabling of interrupts is still performed, as they are enabled for
all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the
EN bit for an SSI where it was not set by rsnd_ssi_start.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0c258657dd ]
The master SSI of a multi-SSI setup was attached both to the
RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream.
This is not correct wrt. the meaning of being "parent" in the rest of
the SSI code, where it seems to indicate an SSI that provides clock and
word sync but is not transmitting/receiving audio data.
Not treating the multi-SSI master as parent allows removal of various
special cases to the rsnd_ssi_is_parent conditions introduced in commit
a09fb3f28a ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode").
It also fixes the issue that operations performed via rsnd_dai_call()
were performed twice for the master SSI. This caused some "status check
failed" spam when stopping a multi-SSI stream as the driver attempted to
stop the master SSI twice.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 326b509238 ]
If we don't find any pcm, pcm will point at address at an offset from
the the list head and not a meaningful structure. Fix this by returning
correct pcm if found and NULL if not. Found with coccinelle.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b94e164759 ]
The HDMI?_SEL register maps up to four stereo SSI data lanes onto the
sdata[0..3] inputs of the HDMI output block. The upper half of the
register contains four blocks of 4 bits, with the most significant
controlling the sdata3 line and the least significant the sdata0 line.
The shift calculation has an off-by-one error, causing the parent SSI to
be mapped to sdata3, the first multi-SSI child to sdata0 and so forth.
As the parent SSI transmits the stereo L/R channels, and the HDMI core
expects it on the sdata0 line, this causes no audio to be output when
playing stereo audio on a multichannel capable HDMI out, and
multichannel audio has permutated channels.
Fix the shift calculation to map the parent SSI to sdata0, the first
child to sdata1 etc.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a09fb3f28a ]
The parent SSI of a multi-SSI setup must be fully setup, started and
stopped since it is also part of the playback/capture setup. So only
skip the SSI (as per commit 203cdf51f2 ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 597b046f0d ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26d8788159 ]
As done in already existing cases, we should use le32_to_cpu macro while
accessing hdr->magic. Found with sparse.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit aa7812737f ]
As mentioned slightly out of patch context in the code, there
is no reset routine for the chip. On boards where the chip is
supplied by a fixed regulator, it might not even be resetted
during (e.g. watchdog) reboot and can be in any state.
If the device is probed with VAG enabled, the driver's probe
routine will generate a loud pop sound when ANA_POWER is
being programmed. Avoid this by properly disabling just the
VAG bit and waiting the required power down time.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Fabio Estevam <festivem@gmail.com>
Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b3677fc3d6 ]
Function pcm_new_ver can fail, so we should check it's return value and
handle possible error.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6856e887ea ]
Function soc_tplg_add_route can propagate error code from callback, we
should check its return value and handle fail in correct way.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2ae548f30d ]
Functions soc_tplg_denum_create, soc_tplg_dmixer_create,
soc_tplg_dbytes_create can fail, so their return values should be
checked and error should be propagated.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 482db55ae8 ]
Function soc_tplg_create_tlv can fail, so we should check if it succeded
or not and proceed appropriately.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Need to use pm_runtime_get_sync & pm_runtime_put_sync in probe
to bind the attached power domain with device.
Fixes: ee1a47b87f ("MLK-23618-6: ASoC: fsl_dsp: refine handling of multi power domain")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
The call flow:
devm_regmap_init_mmio_clk
- clk_prepare()
- clk_pm_runtime_get()
Cause the power domain of ipg clock always be enabled.
which impact the power consumption.
so we can't bind clock with regmap, but explicitly enable
clock when using.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
The call flow:
devm_regmap_init_mmio_clk
- clk_prepare()
- clk_pm_runtime_get()
Cause the power domain of mem clock always be enabled.
which impact the power consumption.
so we can't bind clock with regmap, but explicitly enable
clock when using.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
With imx-pcm-dma, the dma channel is created in probe, when
there is power domain attached with dma device, the power
domain will be enabled when channel is created, then the
power of the domain will be always enabled from beginning.
So switch to imx-pcm-dma-v2, then the dma channel will be
created when playback or capture really started.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
The call flow:
devm_regmap_init_mmio_clk
- clk_prepare()
- clk_pm_runtime_get()
Cause the power domain of bus clock always be enabled.
which impact the power consumption.
So we can't bind clock with regmap, then explicitly enable
clock when using.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
commit 5ce00760a8 upstream.
gcc-10 points out a few instances of suspicious integer arithmetic
leading to value truncation:
sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure':
sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask'
351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c: In function 'snd_miro_configure':
sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask'
1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~
These are all harmless here as only the low 8 bit are passed down
anyway. Change the macros to inline functions to make the code
more readable and also avoid the warning.
Strictly speaking those functions also need locking to make the
read/write pair atomic, but it seems unlikely that anyone would
still run into that issue.
Fixes: 1841f613fd ("[ALSA] Add snd-miro driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4285de0725 upstream.
The checks of the plugin buffer overflow in the previous fix by commit
f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
are put in the wrong places mistakenly, which leads to the expected
(repeated) sound when the rate plugin is involved. Fix in the right
places.
Also, at those right places, the zero check is needed for the
termination node, so added there as well, and let's get it done,
finally.
Fixes: f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200424193350.19678-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a2f6472409 upstream.
Fix the following coccicheck warning:
sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846
After add sanity check to pass klockwork check,
The spdif_mutex should be unlock before return true
in check_non_pcm_per_cvt().
Fixes: 960a581e22 ("ALSA: hda: fix some klockwork scan warnings")
Signed-off-by: Wu Bo <wubo40@huawei.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 547d2c9cf4 upstream.
The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0.
Fixes: f656891c66 ("ALSA: usb-audio: add more quirks for DSD interfaces")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ef0b3203c7 upstream.
This new Lenovo ThinkCenter has two front mics which can't be handled
by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change
the location for one of the mics.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
This reverts commit 005aa9f0af which is
commit 1164284270 upstream.
It should not have been backported, I only looked at the "Fixes:" tag,
not the changelog text itself, my fault.
Reported-by: Guillaume Tucker <guillaume.tucker@collabora.com>
Cc: Jerome Brunet <jbrunet@baylibre.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Sasha Levin <sashal@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a22ae72b86 upstream.
v5.4 changes in soc-core tightened the checks on soc_dapm_add_routes,
which results in the ASoC card probe failing.
Introduce a flag to be set in machine drivers to prevent the probe
from stopping in case of incomplete topologies or missing routes. This
flag is for backwards compatibility only and shall not be used for
newer machine drivers.
Example with an HDaudio card with a bad topology:
[ 236.177898] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Failed to
add route iDisp1_out -> direct -> iDisp1 Tx
[ 236.177902] skl_hda_dsp_generic skl_hda_dsp_generic:
snd_soc_bind_card: snd_soc_dapm_add_routes failed: -19
with the disable_route_checks set:
[ 64.031657] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Failed to
add route iDisp1_out -> direct -> iDisp1 Tx
[ 64.031661] skl_hda_dsp_generic skl_hda_dsp_generic:
snd_soc_bind_card: disable_route_checks set, ignoring errors on
add_routes
Fixes: daa480bde6 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200309192744.18380-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9a6418487b ]
Before the pci_driver->probe() is called, the pci subsystem calls
runtime_forbid() and runtime_get_sync() on this pci dev, so only call
runtime_put_autosuspend() is not enough to enable the runtime_pm on
this device.
For controllers with vgaswitcheroo feature, the pci/quirks.c will call
runtime_allow() for this dev, then the controllers could enter
rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like
Intel hda controllers, the runtime_pm is not enabled because the
runtime_allow() is not called.
Since it is no harm calling runtime_allow() twice, here let hda
driver call runtime_allow() for all controllers. Then the runtime_pm
is enabled on all controllers after the put_autosuspend() is called.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9479e75fca ]
Currently, when the HD-audio controller driver doesn't detect any
codecs, it tries to abort the probe. But this abort happens at the
delayed probe, i.e. the primary probe call already returned success,
hence the driver is never unbound until user does so explicitly.
As a result, it may leave the HD-audio device in the running state
without the runtime PM. More badly, if the device is a HD-audio bus
that is tied with a GPU, GPU cannot reach to the full power down and
consumes unnecessarily much power.
This patch changes the logic after no-codec situation; it continues
probing without the further codec initialization but keep the
controller driver running normally.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Tested-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2393e7555b ]
snd-hda-intel driver handles the most of its probe task in the delayed
work (either via workqueue or via firmware loader). When an error
happens in the later delayed probe, we can't deregister the device
itself because the probe callback already returned success and the
device was bound. So, for now, we set hda->init_failed flag and make
the rest untouched until the device gets really unbound.
However, this leaves the device up running, keeping the resources
without any use that prevents other operations.
In this patch, we release the resources at first when a probe error
happens in the delayed probe stage, but keeps the top-level object, so
that the PM and other ops can still refer to the object itself.
Also for simplicity, snd_hda_intel object is allocated via devm, so
that we can get rid of the explicit kfree calls.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 1e060a453c upstream.
After suspend & resume, wm8960_hw_params may be called when
bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking
is not called. But if sample rate is changed at that time, then
the output clock rate will be not correct.
So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params
is not necessary and it causes above issue.
Fixes: 3176bf2d7c ("ASoC: wm8960: update pll and clock setting function")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1587468525-27514-1-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 1164284270 upstream.
Since the addition of commit 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow
playback/capture if supported"), meson-axg cards which have codec-to-codec
links fail to init and Oops:
Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
Internal error: Oops: 96000044 [#1] PREEMPT SMP
CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
pc : invalidate_paths_ep+0x30/0xe0
lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
Call trace:
invalidate_paths_ep+0x30/0xe0
snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
dpcm_path_get+0x38/0xd0
dpcm_fe_dai_open+0x70/0x920
snd_pcm_open_substream+0x564/0x840
snd_pcm_open+0xfc/0x228
snd_pcm_capture_open+0x4c/0x78
snd_open+0xac/0x1a8
...
While initiliazing the links, ASoC treats the codec-to-codec links of this
card type as a DPCM backend. This error eventually leads to the Oops.
Most of the card driver code is shared between DPCM backends and
codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on
codec-to-codec links, leading to this problem. This commit fixes that.
Fixes: 0a8f1117a6 ("ASoC: meson: axg-card: add basic codec-to-codec link support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9df8ba7c63 upstream.
If probe fails after enabling the regulators regulator_put is called for
each supply without having them disabled before. This produces some
warnings like
WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c
[<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14)
[<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4)
[<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4)
[<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c)
[<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38)
[<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38)
[<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c)
[<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c)
[<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c)
[<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60)
[<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc)
[<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0)
[<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0)
[<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108)
[<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88)
[<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250)
[<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244)
[<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540)
[<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8)
[<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20)
Fixes: 3fd6e7d9a1 (ASoC: tas571x: New driver for TI TAS571x power amplifiers)
Signed-off-by: Philipp Puschmann <p.puschmann@pironex.de>
Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0c824ec094 upstream.
For some reason, the MI2S DAIs do not have channels_min/max defined.
This means that snd_soc_dai_stream_valid() returns false,
i.e. the DAIs have neither valid playback nor capture stream.
It's quite surprising that this ever worked correctly,
but in 5.7-rc1 this is now failing badly: :)
Commit 0e9cf4c452 ("ASoC: pcm: check if cpu-dai supports a given stream")
introduced a check for snd_soc_dai_stream_valid() before calling
hw_params(), which means that the q6i2s_hw_params() function
was never called, eventually resulting in:
qcom-q6afe aprsvc:q6afe:4:4: no line is assigned
... even though "qcom,sd-lines" is set in the device tree.
Commit 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
now even avoids creating PCM devices if the stream is not supported,
which means that it is failing even earlier with e.g.:
Primary MI2S: ASoC: no backend playback stream
Avoid all that trouble by adding channels_min/max for the MI2S DAIs.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
A power domain associated with a device may be disabled in
a separate thread by "genpd_power_off_work_fn" function in
case the device has no PM runtime enabled at that moment.
This will stop the parent clock of "bus" clk and hang
the probe in regmap read/write operation. In order to avoid
this PM runtime must be enabled before any regmap read/write
ops. Aside of this replace clk bus clocks with
pm_runtime_get/put_sync calls.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Fixes: c2641e1974 ("MLK-23618-9: ASoC: fsl_sai: Don't bind clock with regmap")
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Set regmap to use regcache only in probe in order to avoid
issue on cat /sys/kernel/debug/regmap/30cc0000.xcvr/registers
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit ebf1474745 upstream.
snd_soc_dapm_kcontrol widget which is created by autodisable control
should contain correct on_val, mask and shift because it is set when the
widget is powered and changed value is applied on registers by following
code in dapm_seq_run_coalesced().
mask |= w->mask << w->shift;
if (w->power)
value |= w->on_val << w->shift;
else
value |= w->off_val << w->shift;
Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent
double shift.
And, on_val in dapm_kcontrol_set_value() is modified to get correct
value in the dapm_seq_run_coalesced().
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1c82679258 upstream.
Many Focusrite devices supports a limited set of sample rates per
altsetting. These includes audio interfaces with ADAT ports:
- Scarlett 18i6, 18i8 1st gen, 18i20 1st gen;
- Scarlett 18i8 2nd gen, 18i20 2nd gen;
- Scarlett 18i8 3rd gen, 18i20 3rd gen;
- Clarett 2Pre USB, 4Pre USB, 8Pre USB.
Maximum rate is exposed in the last 4 bytes of Format Type descriptor
which has a non-standard bLength = 10.
Tested-by: Alexey Skobkin <skobkin-ru@ya.ru>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 59e1947ca0 upstream.
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".
When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.
The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.
Fix this issue by jumping to "end" label when those error scenarios
occur.
Fixes: 447d6275f0 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b392350ec3 upstream.
As the recent regression showed, we want sometimes to turn off the
audio component binding just for debugging. This patch adds the
module option to control it easily without compilation.
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200415162523.27499-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 67791202c5 upstream.
The commit 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping
spec->init_amp detection") changed the way to assign spec->init_amp
field that specifies the way to initialize the amp. Along with the
change, the commit also replaced a few fixups that set spec->init_amp
in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather
aligning to the other fixups, and not supposed to change the actual
behavior.
However, this change turned out to cause a regression on FSC S7020,
which hit exactly the above. The reason was that there is still one
place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE
call, namely in alc_ssid_check().
This patch fixes the regression by adding the proper spec->init_amp
override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED.
Fixes: 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping spec->init_amp detection")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329
Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7686e34852 upstream.
The error handling code in usX2Y_rate_set() may hit a potential NULL
dereference when an error occurs before allocating all us->urb[].
Add a proper NULL check for fixing the corner case.
Reported-by: Lin Yi <teroincn@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit fef66ae73a ]
It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a017 ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.
My first thought was that this could be easily deduced from the
associated terminals, but some of them have even no associate terminal
ID, hence it's not too trivial to figure out.
Since the number of such terminals are small and limited, this patch
implements another quirk table for the simple mapping of the
connectors. It's not really scalable, but let's hope that there will
be not many such funky devices in future.
Fixes: 7dc3c5a017 ("ALSA: usb-audio: Don't create jack controls for PCM terminals")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a43c1c41bc ]
TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need
yet more quirks for the proper control names.
This patch provides the mapping table for those boards, correcting the
FU names for volume and mute controls as well as the terminal names
for jack controls. It also improves build_connector_control() not to
add the directional suffix blindly if the string is given from the
mapping table.
With this patch applied, the new UCM profiles will be effective.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a8cf44f085 ]
The commit 3c6fd1f07e ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
This patch reverts the corresponding entry as a temporary solution.
Although Zenith II and co will see get the empty HD-audio bus again,
it'd be merely resource wastes and won't affect the functionality,
so it's no end of the world. We'll need to address this later,
e.g. by either switching to DMI string matching or using PCI ID &
SSID pairs.
Fixes: 3c6fd1f07e ("ALSA: hda: Add driver blacklist")
Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e6110114d1 ]
Check if DMA pages were successfully allocated in initialization
before calling free. For many types of memory (like sgbufs)
the extra free is harmless, but not all backends track allocation
state, so add an explicit check.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200124213625.30186-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 73d8c94084 ]
Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card.
The MIDI controller part is standard but the PCM part is "vendor specific".
Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE.
Input is not working.
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c8b78f24c1 ]
The MPMAN MPWIN895CL 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/20200405133726.24154-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 81630dc042 ]
sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some
places it is called with the drv->lock mutex already held.
So it must always be called with the mutex locked. This commit adds missing
locking in the sst_set_be_modules() code-path.
Fixes: 24c8d14192 ("ASoC: Intel: mrfld: add DSP core controls")
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/20200402185359.3424-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The issue is GPU will crash when do dsp suspend test, and only
meet this issue when load GPU as module.
The reason is that dsp framework's global data set size little
than actul size. So dsp will touch the memory that not owned
dsp, then caused GPU crash.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
We need to check priv->pd_dev after allocating memory. Issue
detected by Coverity (CID9607920).
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Unnecessary header files may slow down build performance.
Issue reported by coverity (CID9607920).
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
In order to be able to access SAI3/SDMA3 registers
we need to enable SAI3_IPG and SDMA3_ROOT clocks.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Currently the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP are defined
in the SOF PCM core, which doesn't scale. To account for platform
variations, these flags need to be set in DSP ops.
This patch only moves the definitions and does not change any
functionality.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We're passing "&posn" instead of "posn" so it ends up corrupting
memory instead of doing something useful.
Fixes: 53e0c72d98 ("ASoC: SOF: Add support for IPC IO between DSP and Host")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200303101858.ytehbrivocyp3cnf@kili.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
In tx_wait_done the ipc payload is copied before the DSP transaction
error code is checked. This might lead to corrupted data in kernel side
even though the error would be handled later. It is also pointless to
copy the data in case of error. So change the order of error check and
copy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Link: https://lore.kernel.org/r/20200228231850.9226-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
M0+ core must be released prior raising any interrupt
to M0+ core, otherwise interrupt will be lost.
Fixes: b164260078 ("MLK-23567-2 ASoC: fsl_xcvr: enable SPDIF TX")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Before the parameters getting from dsp is not right, now correct it.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
XCVR driver working mode is set by "fsl,xcvr-mode" DTS
property. Being static - it does not allow changing
XCVR driver mode at runtime. Rework the driver by replacing
static DTS property with amixer control.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 25faa4bd37 upstream.
At the error path of the firmware loading error, the driver tries to
release the card object and set NULL to drvdata. This may be referred
badly at the possible PM action, as the driver itself is still bound
and the PM callbacks read the card object.
Instead, we continue the probing as if it were no option set. This is
often a better choice than the forced abort, too.
Fixes: 5cb543dba9 ("ALSA: hda - Deferred probing with request_firmware_nowait()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 10db5bccc3 upstream.
freeze_noirq and thaw_noirq need to check the PM availability like
other PM ops. There are cases where the device got disabled due to
the error, and the PM operation should be ignored for that.
Fixes: 3e6db33aaf ("ALSA: hda - Set SKL+ hda controller power at freeze() and thaw()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 934b96594e upstream.
Add the mapping check to build_connector_control() so that the device
specific quirk can provide the node to skip for the badly behaving
connector controls. As an example, ALC1220-VB-based codec implements
the skip entry for the broken SPDIF connector detection.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7dc3c5a017 upstream.
Some funky firmwares set the connector flag even on PCM terminals
although it doesn't make sense (and even actually the firmware doesn't
react properly!). Let's skip creation of jack controls in such a
case.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3507245b82 upstream.
The mapping table may contain also ignore_ctl_error flag for devices
that are known to behave wild. Since this flag always writes the
card's own ignore_ctl_error flag, it overrides the value already set
by the module option, so it doesn't follow user's expectation.
Let's fix the code not to clear the flag that has been set by user.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 48cc429735 upstream.
The ignore_ctl_error option should filter the error at kctl accesses,
but there was an overlook: mixer_ctl_connector_get() returns an error
from the request.
This patch covers the forgotten code path and apply filter_error()
properly. The locking error is still returned since this is a fatal
error that has to be reported even with ignore_ctl_error option.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4963d66b8a upstream.
On Asus FX505DT with Realtek ALC233, the headset mic is connected
to pin 0x19, with default 0x411111f0.
Enable headset mic by reconfiguring the pin to an external mic
associated with the headphone on 0x21. Mic jack detection was also
found to be working.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131
Signed-off-by: Adam Barber <barberadam995@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3025571edd upstream.
Currently function sst_platform_get_resources always returns zero and
error return codes set by the function are never returned. Fix this
by returning the error return code in variable ret rather than the
hard coded zero.
Addresses-Coverity: ("Unused value")
Fixes: f533a035e4 ("ASoC: Intel: mrfld - create separate module for pci part")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200208220720.36657-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f5e056e1e4 upstream.
The check on p->sink looks bogus, I believe it should be p->source
since the following code blocks are related to p->source. Fix
this by replacing p->sink with p->source.
Fixes: 24c8d14192 ("ASoC: Intel: mrfld: add DSP core controls")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Addresses-Coverity: ("Copy-paste error")
Link: https://lore.kernel.org/r/20191119113640.166940-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
For IEC958 kcontrol type amixer tool returns just first 4 bytes,
add a byte array control so that we'll be able to set all 24 bytes
of TX CS by using amixer.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
[ Upstream commit 7506baeed8 ]
The commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
converts some function calls to their non-devm equivalents. The
appropriate cleanup code was added to the remove function, but not
to the probe function. Add a call to snd_dmaengine_pcm_unregister
to compensate for the call to snd_dmaengine_pcm_register in case
of subsequent failure.
Fixes: commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 1d3aa4a551 upstream.
MSI GL63 laptop requires the similar quirk like other MSI models,
ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID
for the device, hence we need to take the codec SSID (1462:1275)
instead.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ca707b3f00 upstream.
The audio setup on the Lenovo Carbon X1 8th gen is the same as that on
the Lenovo Carbon X1 7th gen, as such it needs the same
ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk.
This fixes volume control of the speaker not working among other things.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196
Cc: stable@vger.kernel.org
Suggested-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f36938aa74 upstream.
patch_realtek.c has historically failed to properly configure the PC
Beep Hidden Register for the ALC256 codec (among others). Depending on
your kernel version, symptoms of this misconfiguration can range from
chassis noise, picked up by a poorly-shielded PCBEEP trace, getting
amplified and played on your internal speaker and/or headphones to loud
feedback, which responds to the "Headphone Mic Boost" ALSA control,
getting played through your headphones. For details of the problem, see
the patch in this series titled "ALSA: hda/realtek - Set principled PC
Beep configuration for ALC256", which fixes the configuration.
These symptoms have been most noticed on the Dell XPS 13 9350 and 9360,
popular laptops that use the ALC256. As a result, several model-specific
fixups have been introduced to try and fix the problem, the most
egregious of which locks the "Headphone Mic Boost" control as a hack to
minimize noise from a feedback loop that shouldn't have been there in
the first place.
Now that the underlying issue has been fixed, remove all these fixups.
Remaining fixups needed by the XPS 13 are all picked up by existing pin
quirks.
This change should, for the XPS 13 9350/9360
- Significantly increase volume and audio quality on headphones
- Eliminate headphone popping on suspend/resume
- Allow "Headphone Mic Boost" to be set again, making the headphone
jack fully usable as a microphone jack too.
Fixes: 8c69729b44 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd78561 ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c447374494 upstream.
The Realtek PC Beep Hidden Register[1] is currently set by
patch_realtek.c in two different places:
In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to
non-beep input on 1Ah and no 1Ah loopback to either headphones or
speakers. (Although, curiously, the loopback amp is still enabled.) This
write was added fairly recently by commit e3743f4311 ("ALSA:
hda/realtek - Dell headphone has noise on unmute for ALC236") and is a
safe default. However, it happens in the wrong place:
alc_fill_eapd_coef() runs on module load and cold boot but not on S3
resume, meaning the register loses its value after suspend.
Conversely, in alc256_init(), the register is updated to unset bit 13
(disable speaker loopback) and set bit 5 (set non-beep input on 1Ah).
Although this write does run on S3 resume, it's not quite enough to fix
up the register's default value of 0x3717. What's missing is a set of
bit 14 to disable headphone loopback. Without that, we end up with a
feedback loop where the headphone jack is being driven by amplified
samples of itself[2].
This change eliminates the update in alc256_init() and replaces it with
the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is
supposed to be the codec's default value, so using it will make
debugging easier for Realtek.
Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs.
[1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst
[2] Setting the "Headphone Mic Boost" control from userspace changes
this feedback loop and has been a widely-shared workaround for headphone
noise on laptops like the Dell XPS 13 9350. This commit eliminates the
feedback loop and makes the workaround unnecessary.
Fixes: e1e8c1fdce ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 476c02e0b4 upstream.
On the Lenovo X1C7 machines, after we plug the headset, the rt_resume()
and rt_suspend() of the codec driver will be called periodically, the
driver can't stay in the rt_suspend state even users doen't use the
sound card.
Through debugging, I found when running rt_suspend(), it will call
alc225_shutup(), in this function, it will change 3k pull down control
by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a
fake key event and that event will resume the codec, when codec
suspend agin, it will trigger the fake key event one more time, this
process will repeat.
If disable the key event before changing the pull down control, it
will not trigger fake key event. It also needs to restore the pull
down control and re-enable the key event, otherwise the system can't
get key event when codec is in rt_suspend state.
Also move some functions ahead of alc225_shutup(), this can save the
function declaration.
Fixes: 76f7dec08f (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1)
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200329082018.20486-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f5a88b0acc upstream.
The system in question uses ALC285, and it uses GPIO 0x04 to control its
mute LED.
The mic mute LED can be controlled by GPIO 0x01, however the system uses
DMIC so we should use that to control mic mute LED.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200327044626.29582-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ae769d3556 upstream.
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications. The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.
This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.
Fixes: f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Tested-and-reported-by: Jari Ruusu <jari.ruusu@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200403072515.25539-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c47914c00b upstream.
The access to Analog Capture Source control value implemented in
prodigy_hifi.c is wrong, as caught by the recently introduced sanity
check; it should be accessing value.enumerated.item[] instead of
value.integer.value[]. This patch corrects the wrong access pattern.
Fixes: 6b8d6e5518 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0ad3f0b384 upstream.
The beep control helper function blindly stores the values in two
stereo channels no matter whether the actual control is mono or
stereo. This is practically harmless, but it annoys the recently
introduced sanity check, resulting in an error when the checker is
enabled.
This patch corrects the behavior to store only on the defined array
member.
Fixes: 0401e8548e ("ALSA: hda - Move beep helper functions to hda_beep.c")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Zhang Peng