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commit d2c63b7dfd06788a466d5ec8a850491f084c5fc2 upstream.
It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen
Laptop) is fixed by the same workaround applied to other AMD chips.
Update the driver_data entry for Raven (1022:15e3) to use the newly
introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains
AZX_DCAPS_PM_RUNTIME, we can drop that bit, too.
Reported-and-tested-by: Dennis Padiernos <depadiernos@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190920073040.31764-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 7b485d175631be676424aedb8cd2f66d0c93da78 upstream.
The same fixup to enable laptop imic is needed for ASUS M9V with AD1986A
codec like another HP machine.
Signed-off-by: Shih-Yuan Lee (FourDollars) <fourdollars@debian.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190920134052.GA8035@localhost
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3a9236e97207f2469254b4098995159b80174d95 upstream.
At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data
channels per data block in CIP.
Both iO14 and iO26 have the same contents in their configuration ROM.
For this reason, ALSA Dice driver attempts to distinguish them according
to the value of TX0_AUDIO register at probe callback. Although the way is
valid at lower and middle sampling rate, it's lastly invalid at higher
sampling rate because because the two models returns the same value for
read transaction to the register.
In the most cases, users just plug-in the device and ALSA dice driver
detects it. In the case, the device runs at lower sampling rate and
the driver detects expectedly. For this reason, this commit leaves the
way to detect as is.
Fixes: 28b208f600a3 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Cc: <stable@vger.kernel.org> # v4.18+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190916101851.30409-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f41f900568d9ffd896cc941db7021eb14bd55910 upstream.
EVGA NU Audio is actually a USB audio device on a PCIexpress card,
with it's own USB controller. It supports both PCM and DSD.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190924071143.30911-1-jussi@sonarnerd.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 029d2c0fd61eac74700fb4ffff36fc63bfff7e5e upstream.
This patch adds quirk VID ID for Hiby portable players family with
native DSD playback support.
Signed-off-by: Ilya Pshonkin <sudokamikaze@protonmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190917074937.157802-1-ilya.pshonkin@netforce.ua
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2a36c16efab254dd6017efeb35ad88ecc96f2328 upstream.
This ThinkCentre machine has a new realtek codec alc222, it is not
in the support list, we add it in the realtek.c then this machine
can apply FIXUPs for the realtek codec.
And this machine has two front mics which can't be handled
by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as
a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't
work on this machine. Fortunately another FIXUP
ALC283_FIXUP_HEADSET_MIC also can change the location for one of the
two mics on this machine.
Link: https://lore.kernel.org/r/20190904055327.9883-1-hui.wang@canonical.com
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 60083f9e94b2f28047d71ed778adf89357c1a8fb upstream.
Original pin node values of ASUS UX431FL with ALC294:
0x12 0xb7a60140
0x13 0x40000000
0x14 0x90170110
0x15 0x411111f0
0x16 0x411111f0
0x17 0x90170111
0x18 0x411111f0
0x19 0x411111f0
0x1a 0x411111f0
0x1b 0x411111f0
0x1d 0x4066852d
0x1e 0x411111f0
0x1f 0x411111f0
0x21 0x04211020
1. Has duplicated internal speakers (0x14 & 0x17) which makes the output
route become confused. So, the output volume cannot be changed by
setting.
2. Misses the headset mic pin node.
This patch disables the confusing speaker (NID 0x14) and enables the
headset mic (NID 0x19).
Link: https://lore.kernel.org/r/20190902100054.6941-1-jian-hong@endlessm.com
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d33cd42d86671bed870827aa399aeb9f1da74119 upstream.
HP Pavilion 15 (AMD Ryzen-based model) with 103c:84e7 needs the same
quirk like HP Envy/Spectre x360 for enabling the mute LED over Mic3 pin.
[ rearranged in the SSID number order by tiwai ]
Signed-off-by: Sam Bazley <sambazley@fastmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 89781d0806c2c4f29072d3f00cb2dd4274aabc3d upstream.
The recent change to shuffle the codec initialization procedure for
Realtek via commit 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on
later") caused the silent output on some machines. This change was
supposed to be safe, but it isn't actually; some devices have quirk
setups to override the EAPD via COEF or BTL in the additional verb
table, which is applied at the beginning of snd_hda_gen_init(). And
this EAPD setup is again overridden in alc_auto_init_amp().
For recovering from the regression, tell snd_hda_gen_init() not to
apply the verbs there by a new flag, then apply the verbs in
alc_init().
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204727
Fixes: 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 333f31436d3db19f4286f8862a00ea1d8d8420a1 upstream.
Since the chained quirks via chained_before flag is applied before the
depth check, it may lead to the endless recursive calls, when the
chain were set up incorrectly. Fix it by moving the depth check at
the beginning of the loop.
Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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This reverts commit ab4f4d331a982f52a964f7f4d1ec452280f122e7 which is
commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b upstream.
Mark Brown writes:
I nacked this patch when Sasha posted it - it only improves
diagnostics and might make systems that worked by accident break
since it turns things into a hard failure, it won't make
anything that didn't work previously work.
Reported-by: Mark Brown <broonie@kernel.org>
Cc: Ricard Wanderlof <ricardw@axis.com>
Cc: Sasha Levin <sashal@kernel.org>
Link: https://lore.kernel.org/lkml/20190904181027.GG4348@sirena.co.uk
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1a15718b41df026cffd0e42cfdc38a1384ce19f9 upstream.
Behringer UFX1604 requires the similar quirk to apply implicit fb like
another Behringer model UFX1204 in order to fix the noisy playback.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 6de3c9e3f6b3eaf66859e1379b3f35dda781416b upstream.
The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code. It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.
This patch fixes the check before the conversion.
While we're at it, correct a typo in the comment in the function,
too.
Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 75545304eba6a3d282f923b96a466dc25a81e359 upstream.
The input pool of a client might be deleted via the resize ioctl, the
the access to it should be covered by the proper locks. Currently the
only missing place is the call in snd_seq_ioctl_get_client_pool(), and
this patch papers over it.
Reported-by: syzbot+4a75454b9ca2777f35c7@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f9ef724d4896763479f3921afd1ee61552fc9836 upstream.
"enabled" parameter historically referred to the device input or
output, not to the led indicator. After the changes added with the led
helper functions the mic mute led logic refers to the led and not to
the mic input which caused led indicator to be negated.
Fixing logic in cxt_update_gpio_led and updated
cxt_fixup_gpio_mute_hook
Also updated debug messages to ease further debugging if necessary.
Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1bc8d18c75fef3b478dbdfef722aae09e2a9fde7 upstream.
I forgot to release the allocated object at the early error path in
line6_init_pcm(). For addressing it, slightly shuffle the code so
that the PCM destructor (pcm->private_free) is assigned properly
before all error paths.
Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2ca371d847511f97ef991ef612a2ce805489840e upstream.
This patch adds a new PCI subsys ID for the SBZ, as found and tested by
me and some reddit users.
Link: https://lore.kernel.org/lkml/20190819204008.14426-1-p.rekowski@gmail.com
Signed-off-by: Paweł Rekowski <p.rekowski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f9f0e9ed350e15d51ad07364b4cf910de50c472a upstream.
The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a
variable size depending on both input and output pins. Its size is to
fit with input * output bits. The problem is that the input size
can't be determined simply from the unit descriptor itself but it
needs to parse the whole connected sources. Although the
uac_mixer_unit_get_channels() tries to check some possible overflow of
this bitmap, it's incomplete due to the lack of the evaluation of
input pins.
For covering possible overflows, this patch adds the bitmap overflow
check in the loop of input pins in parse_audio_mixer_unit().
Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 88639051017fb61a414b636dd0fc490da2b62b64 ]
AMD platform device acp3x_rv_i2s created by parent PCI device
driver. Pass struct device of the parent to
snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use
correct dma_ops. Otherwise, it will use default dma_ops which
is nommu_dma_ops on x86_64 even when IOMMU is enabled and
set to non passthrough mode.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 1e112c35e3c96db7c8ca6ddaa96574f00c06e7db ]
The slot_width is a property for the bus while the constraint for
SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format.
Applying slot_width constraint to sample_bits works most of the time, but
it will blacklist valid formats in some cases.
With slot_width 24 we can support S24_3LE and S24_LE formats as they both
look the same on the bus, but a a 24 constraint on sample_bits would not
allow S24_LE as it is stored in 32bits in memory.
Implement a simple hw_rule function to allow all formats which require less
or equal number of bits on the bus as slot_width (if configured).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 789e162a6255325325bd321ab0cd51dc7e285054 ]
This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0.
Revert "ASoC: rockchip: i2s: Support mono capture"
Previous discussion in
https://patchwork.kernel.org/patch/10147153/
explains the issue of the patch.
While device is configured as 1-ch, hardware is still
generating a 2-ch stream.
When user space reads the data and assumes it is a 1-ch stream,
the rate will be slower by 2x.
Revert the change so 1-ch is not supported.
User space can selectively take one channel data out of two channel
if 1-ch is preferred.
Currently, both channels record identical data.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 34a2a80ff30b5d2330abfa8980c7f0cc15a8158a ]
When running McASP as master capture alone will not record any audio unless
a parallel playback stream is running. As soon as the playback stops the
captured data is going to be silent again.
In McASP master mode we need to set the PDIR for the clock pins and fix
the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above
AMUTE.
This went unnoticed as most of the boards uses McASP as slave and neither
of these issues are visible (audible) in those setups.
Fixes: ca3d9433349e ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b ]
If the DAI format setup fails, there is no valid communication format
between CPU and CODEC, so fail card instantiation, rather than continue
with a card that will most likely not function properly.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 45004d66f2a28d78f543fb2ffbc133e31dc2d162 ]
In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in
dapm_cnew_widget() to hold the new dapm widget. Then, different actions are
taken according to the id of the widget, i.e., 'w->id'. If any failure
occurs during this process, snd_soc_dapm_new_control_unlocked() should be
terminated by going to the 'request_failed' label. However, the allocated
kernel buffer is not freed on this code path, leading to a memory leak bug.
To fix the above issue, free the buffer before returning from
snd_soc_dapm_new_control_unlocked() through the 'request_failed' label.
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 8dd26dff00c0636b1d8621acaeef3f6f3a39dd77 ]
DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a
list of the widgets connected to a specific front end DAI so it
can search through this list for available back end DAIs. The
custom_stop_condition was added to is_connected_ep to facilitate this
list not containing more widgets than is necessary. Doing so both
speeds up the DPCM handling as less widgets need to be searched and
avoids issues with CODEC to CODEC links as these would be confused
with back end DAIs if they appeared in the list of available widgets.
custom_stop_condition was implemented by aborting the graph walk
when the condition is triggered, however there is an issue with this
approach. Whilst walking the graph is_connected_ep should update the
endpoints cache on each widget, if the walk is aborted the number
of attached end points is unknown for that sub-graph. When the stop
condition triggered, the original patch ignored the triggering widget
and returned zero connected end points; a later patch updated this
to set the triggering widget's cache to 1 and return that. Both of
these approaches result in inaccurate values being stored in various
end point caches as the values propagate back through the graph,
which can result in later issues with widgets powering/not powering
unexpectedly.
As the original goal was to reduce the size of the widget list passed
to the DPCM code, the simplest solution is to limit the functionality
of the custom_stop_condition to the widget list. This means the rest
of the graph will still be processed resulting in correct end point
caches, but only widgets up to the stop condition will be added to the
returned widget list.
Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets")
Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links")
Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit aa2ba991c4206d5b778dcaa7b4997396e79f8e90 ]
The Irbis NB41 netbook has its internal mic on IN2, inverted jack-detect
and stereo speakers, add a quirk for this.
Cc: russianneuromancer@ya.ru
Reported-and-tested-by: russianneuromancer@ya.ru
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20190712112708.25327-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 2abee12c0ab1924a69993d2c063a39a952e7d836 ]
The cpu_dai variable is still being used after the of_node_put() call,
which may result in double-free:
of_node_put(cpu_dai); ---> released here
ret = devm_snd_soc_register_card(dev, card);
if (ret < 0) {
...
goto err_put_clk_i2s; --> jump to err_put_clk_i2s
...
err_put_clk_i2s:
clk_put(priv->clk_i2s_bus);
err_put_sclk:
clk_put(priv->sclk_i2s);
err_put_cpu_dai:
of_node_put(cpu_dai); --> double-free here
Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance")
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Krzysztof Kozlowski <krzk@kernel.org>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Sylwester Nawrocki <s.nawrocki@samsung.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 9b6d104a6b150bd4d3e5b039340e1f6b20c2e3c1 ]
The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.
Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI")
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Krzysztof Kozlowski <krzk@kernel.org>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Sylwester Nawrocki <s.nawrocki@samsung.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit ec3042ad39d4e2ddbc3a3344f90bb10d8feb53bc ]
commit c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in
graph_get_dai_id()") fixups use-after-free issue,
but, it need to use "const" for reg. This patch adds it.
We will have below without this patch
LINUX/sound/soc/generic/audio-graph-card.c: In function 'graph_get_dai_id':
LINUX/sound/soc/generic/audio-graph-card.c:87:7: warning: assignment discards\
'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
reg = of_get_property(node, "reg", NULL);
Fixes: c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Wen Yang <wen.yang99@zte.com.cn>
Link: https://lore.kernel.org/r/87sgrd43ja.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit c152f8491a8d9a4b25afd65a86eb5e55e2a8c380 ]
After calling of_node_put() on the node variable, it is still being
used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.
Fixes: a0c426fe1433 ("ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()")
Link: https://lore.kernel.org/r/1562743509-30496-5-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit aa2e362cb6b3f5ca88093ada01e1a0ace8a517b2 ]
After calling of_node_put() on the ports, port, and node variables,
they are still being used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.
Fixes: dd98fbc558a0 ("ASoC: audio-graph-card: cleanup DAI link loop method - step1")
Link: https://lore.kernel.org/r/1562743509-30496-4-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 27862d5a3325bc531ec15e3c607e44aa0fd57f6f ]
The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.
Fixes: d947cdfd4be2 ("ASoC: simple-card: cleanup DAI link loop method - step1")
Link: https://lore.kernel.org/r/1562743509-30496-3-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 724808ad556c15e9473418d082f8aae81dd267f6 ]
The node variable is still being used after the of_node_put() call,
which may result in use-after-free.
Fixes: cfc652a73331 ("ASoC: simple-card: tidyup prefix for snd_soc_codec_conf")
Link: https://lore.kernel.org/r/1562743509-30496-2-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 37151a41df800493cfcbbef4f7208ffe04feb959 ]
lost wakeup can occur after enabling irq, therefore put task
into interruptible before enabling interrupts,
without this change, task can be put to sleep and snd_pcm_drain
will delay
Fixes: f2b3614cefb6 ("ALSA: PCM - Don't check DMA time-out too shortly")
Signed-off-by: Yuki Tsunashima <ytsunashima@jp.adit-jv.com>
Signed-off-by: Suresh Udipi <sudipi@jp.adit-jv.com>
[ported from 4.9]
Signed-off-by: Adam Miartus <amiartus@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 401714d9534aad8c24196b32600da683116bbe09 upstream.
We have 3 new lenovo laptops which have conexant codec 0x14f11f86,
these 3 laptops also have the noise issue when rebooting, after
letting the codec enter D3 before rebooting or poweroff, the noise
disappers.
Instead of adding a new ID again in the reboot_notify(), let us make
this function apply to all conexant codec. In theory make codec enter
D3 before rebooting or poweroff is harmless, and I tested this change
on a couple of other Lenovo laptops which have different conexant
codecs, there is no side effect so far.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 871b9066027702e6e6589da0e1edd3b7dede7205 upstream.
Make codec enter D3 before rebooting or poweroff can fix the noise
issue on some laptops. And in theory it is harmless for all codecs
to enter D3 before rebooting or poweroff, let us add a generic
reboot_notify, then realtek and conexant drivers can call this
function.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cfef67f016e4c00a2f423256fc678a6967a9fc09 upstream.
In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc().
Then, the pin widgets in 'codec' are parsed. However, if the parsing
process fails, 'spec' is not deallocated, leading to a memory leak.
To fix the above issue, free 'spec' before returning the error.
Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit de768ce45466f3009809719eb7b1f6f5277d9373 upstream.
MSI MPG X570 board is with another AMD HD-audio controller (PCI ID
1022:1487) and it requires the same workaround applied for X370, etc
(PCI ID 1022:1457).
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit daac07156b330b18eb5071aec4b3ddca1c377f2c upstream.
The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.
```
struct uac_mixer_unit_descriptor {
__u8 bLength;
__u8 bDescriptorType;
__u8 bDescriptorSubtype;
__u8 bUnitID;
__u8 bNrInPins;
__u8 baSourceID[];
}
```
This patch fixes the bug by add a sanity check on the length of
the descriptor.
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 19bce474c45be69a284ecee660aa12d8f1e88f18 upstream.
`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.
This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 190d03814eb3b49d4f87ff38fef26d36f3568a60 upstream.
HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same
quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db upstream.
A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:
- Set up the proper driver caps for this controller, similar as the
other AMD controller.
- Correct the DMA position reporting with the fixed FIFO size, which
is similar like as workaround used for VIA chip set.
- Even after the position correction, PulseAudio still shows
mysterious stalls of playback streams when a capture is triggered in
timer-scheduled mode. Since we have no clear way to eliminate the
stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
a temporary workaround.
This patch implements the workarounds. For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.
Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c1c6c877b0c79fd7e05c931435aa42211eaeebaf upstream.
The commit bfcba288b97f ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object. This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.
Fix the bug by setting the PCM hw info flag locally.
Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3d92aa45fbfd7319e3a19f4ec59fd32b3862b723 upstream.
In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.
To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.
Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1be3c1fae6c1e1f5bb982b255d2034034454527a upstream.
In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.
To fix the above issue, free 'b->packets' before returning the error code.
Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v2.6.39+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a67060201b746a308b1674f66bf289c9faef6d09 upstream.
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 3b8179944cb0dd53e5223996966746cdc8a60657 ]
Draining makes little sense in the situation of hardware overrun, as the
hardware will have consumed all its available samples. Additionally,
draining whilst the stream is paused would presumably get stuck as no
data is being consumed on the DSP side.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit a70ab8a8645083f3700814e757f2940a88b7ef88 ]
Partial drain and next track are intended for gapless playback and
don't really have an obvious interpretation for a capture stream, so
makes sense to not allow those operations on capture streams.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 26c3f1542f5064310ad26794c09321780d00c57d ]
Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call
snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which
allow a transition to SNDRV_PCM_STATE_SETUP. The stream should
only be able to move to the setup state once it has received a
SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing
those ioctls whilst in the open state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 4475f8c4ab7b248991a60d9c02808dbb813d6be8 ]
A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.
To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.
Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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