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>
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>
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>
[ 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>
commit 3c6fd1f07e upstream.
The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.
This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry. As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2a48218f8e upstream.
Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio. The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.
As a workaround, simply ignore this node by adding a usbmix_name_map
table entry. The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit abca9e4a04 upstream.
Current topology doesn't add prefix of component to new kcontrol.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 21fca8bdbb upstream.
soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3bbbb7728f upstream.
Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0ab070917a upstream.
If regwshift is 32 and the selected architecture compiles '<<' operator
for signed int literal into rotating shift, '1<<regwshift' became 1 and
it makes regwmask to 0x0.
The literal is set to unsigned long to get intended regwmask.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/001001d60665$db7af3e0$9270dba0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9401d5aa32 upstream.
The 4-bit divider value was written at offset 8, while the jz4740
programming manual locates it at offset 0.
Fixes: 26b0aad80a ("ASoC: jz4740: Add dynamic sampling rate support to jz4740-i2s")
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200306222931.39664-2-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e9097e47e3 upstream.
I have a system which has an EVGA X99 Classified motherboard. The pin
assignments for the HD Audio controller are not correct under Linux.
Windows 10 works fine and informs me that it's using the Recon3Di
driver, and on Linux, `cat
/sys/class/sound/card0/device/subsystem_{vendor,device}` yields
0x3842
0x1038
This patch adds a corresponding entry to the quirk list.
Signed-off-by: Geoffrey Allott <geoffrey@allott.email>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/a6cd56b678c00ce2db3685e4278919f2584f8244.camel@allott.email
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3b36b13d5e upstream.
Commit 317d931392 ("ALSA: hda/realtek - Set default power save node to
0") makes the ALC225 have pop noise on S3 resume and cold boot.
So partially revert this commit for ALC225 to fix the regression.
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/20200311061328.17614-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5461e0530c upstream.
The return value checks in snd_pcm_plug_alloc() are covered with
snd_BUG_ON() macro that may trigger a kernel WARNING depending on the
kconfig. But since the error condition can be triggered by a weird
user space parameter passed to OSS layer, we shouldn't give the kernel
stack trace just for that. As it's a normal error condition, let's
remove snd_BUG_ON() macro usage there.
Reported-by: syzbot+2a59ee7a9831b264f45e@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f2ecf903ef upstream.
Each OSS PCM plugins allocate its internal buffer per pre-calculation
of the max buffer size through the chain of plugins (calling
src_frames and dst_frames callbacks). This works for most plugins,
but the rate plugin might behave incorrectly. The calculation in the
rate plugin involves with the fractional position, i.e. it may vary
depending on the input position. Since the buffer size
pre-calculation is always done with the offset zero, it may return a
shorter size than it might be; this may result in the out-of-bound
access as spotted by fuzzer.
This patch addresses those possible buffer overflow accesses by simply
setting the upper limit per the given buffer size for each plugin
before src_frames() and after dst_frames() calls.
Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com
Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4384f167ce upstream.
The virmidi driver handles sysex event exceptionally in a short-cut
snd_seq_dump_var_event() call, but this missed the reset of the
running status. As a result, it may lead to an incomplete command
right after the sysex when an event with the same running status was
queued.
Fix it by clearing the running status properly via alling
snd_midi_event_reset_decode() for that code path.
Reported-by: Andreas Steinmetz <ast@domdv.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/3b4a4e0f232b7afbaf0a843f63d0e538e3029bfd.camel@domdv.de
Link: https://lore.kernel.org/r/20200316090506.23966-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a124458a12 upstream.
A headset on the desktop like Acer N50-600 does not work, until quirk
ALC662_FIXUP_ACER_NITRO_HEADSET_MODE is applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200317082806.73194-3-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d683469b3c upstream.
The MIDI input event parser of the LINE6 driver may enter into an
endless loop when the unexpected data sequence is given, as it tries
to continue the secondary bytes without termination. Also, when the
input data is too short, the parser returns a negative error, while
the caller doesn't handle it properly. This would lead to the
unexpected behavior as well.
This patch addresses those issues by checking the return value
correctly and handling the one-byte event in the parser properly.
The bug was reported by syzkaller.
Reported-by: syzbot+cce32521ee0a824c21f7@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com
Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 0d6defc7e0 ]
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional. The following problems have been seen
with STM32 SAI driver.
1) DMA channel is not requested:
With the sound card rebind the simplified call sequence is:
stm32_sai_sub_probe
snd_soc_register_component
snd_soc_try_rebind_card
snd_soc_instantiate_card
devm_snd_dmaengine_pcm_register
The problem occurs because the pcm must be registered,
before snd_soc_instantiate_card() is called.
Modify SAI driver, to change the call sequence as follows:
stm32_sai_sub_probe
devm_snd_dmaengine_pcm_register
snd_soc_register_component
snd_soc_try_rebind_card
2) DMA channel is not released:
dma_release_channel() is not called when
devm_dmaengine_pcm_release() is executed.
This occurs because SND_DMAENGINE_PCM_DRV_NAME component,
has already been released through devm_component_release().
devm_dmaengine_pcm_release() should be called before
devm_component_release() to avoid this problem.
Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from SAI driver, to have the right sequence.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Message-Id: <20200304102406.8093-1-olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit f2adbae0cb upstream.
HP want to keep BIOS verb table for release platform.
So, it need to add 0x19 pin for quirk.
Fixes: 5af29028fd ("ALSA: hda/realtek - Add Headset Mic supported for HP cPC")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/74636ccb700a4cbda24c58a99dc430ce@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6b0f95c49d upstream.
Apply const prefix to each coef table array.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9b3193089e upstream.
commit c2caa4da46 ("ASoC: Fix widget powerdown on shutdown") added a
set of the power state during snd_soc_dapm_shutdown to ensure the
widgets powered off. However, when commit 39eb5fd13d
("ASoC: dapm: Delay w->power update until the changes are written")
added the new_power member of the widget structure, to differentiate
between the current power state and the target power state, it did not
update the shutdown to use the new_power member.
As new_power has not updated it will be left in the state set by the
last DAPM sequence, ie. 1 for active widgets. So as the DAPM sequence
for the shutdown proceeds it will turn the widgets on (despite them
already being on) rather than turning them off.
Fixes: 39eb5fd13d ("ASoC: dapm: Delay w->power update until the changes are written")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200228153145.21013-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8308a09e87 upstream.
Incrementation of avail_clk_cnt was incorrectly moved to error path. Put
it back to success path.
Fixes: 6ee927f2f0 ('ASoC: Intel: Skylake: Fix NULL ptr dereference when unloading clk dev')
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200224125202.13784-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ac0a689979 upstream.
When we get a clock error during probe we have to call
regulator_bulk_disable before bailing out, otherwise we trigger
a warning in regulator_put.
Fix this by using "goto err" like in the error cases above.
Fixes: 5a3af12931 ("ASoC: pcm512x: Add PCM512x driver")
Signed-off-by: Matthias Reichl <hias@horus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200220202956.29233-1-hias@horus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6c89ffea60 upstream.
dpcm_show_state() invokes multiple snprintf() calls to concatenate
formatted strings on the fixed size buffer. The usage of snprintf()
is supposed for avoiding the buffer overflow, but it doesn't work as
expected because snprintf() doesn't return the actual output size but
the size to be written.
Fix this bug by replacing all snprintf() calls with scnprintf()
calls.
Fixes: f86dcef87b ("ASoC: dpcm: Add debugFS support for DPCM")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200218111737.14193-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 549cd0ba04 upstream.
The debugfs output of intel skl driver writes strings with multiple
snprintf() calls with the fixed size. This was supposed to avoid the
buffer overflow but actually it still would, because snprintf()
returns the expected size to be output, not the actual output size.
Fix it by replacing snprintf() calls with scnprintf().
Fixes: d14700a01f ("ASoC: Intel: Skylake: Debugfs facility to dump module config")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200218111737.14193-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 64bbacc5f0 upstream.
skl_print_pins() loops over all given pins but it overwrites the text
at the very same position while increasing the returned length.
Fix this to show the all pin contents properly.
Fixes: d14700a01f ("ASoC: Intel: Skylake: Debugfs facility to dump module config")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200218111737.14193-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 613cea5935 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 242c46c023 upstream.
In case of ABI version mismatch, _manifest needs to be freed as
it is just a copy of the original topology manifest. However, if
a driver manifest handler is defined, that would get executed and
the cleanup is never reached. Fix that by getting the return status
of manifest() instead of returning directly.
Fixes: 583958fa2e ("ASoC: topology: Make manifest backward compatible from ABI v4")
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Link: https://lore.kernel.org/r/20200207185325.22320-3-dragos_tarcatu@mentor.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2b2d5c4db7 upstream.
If soc_tplg_link_config() fails, _link needs to be freed in case of
topology ABI version mismatch. However the current code is returning
directly and ends up leaking memory in this case.
This patch fixes that.
Fixes: 593d9e52f9 ("ASoC: topology: Add support to configure existing physical DAI links")
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Link: https://lore.kernel.org/r/20200207185325.22320-2-dragos_tarcatu@mentor.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8b33a134a9 upstream.
A headset on the laptop like ASUS B9450FA does not work, until quirk
ALC294_FIXUP_ASUS_HPE is applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200225072920.109199-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0d45e86d22 upstream.
The Gigabyte X570 Aorus Master 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/20200223092416.15016-2-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 43bcb1c050 ]
snd_hdac_ext_bus_link_get() does not work correctly in case
there are multiple codecs on the bus. It unconditionally
resets the bus->codec_mask value. As per documentation in
hdaudio.h and existing use in client code, this field should
be used to store bit flag of detected codecs on the bus.
By overwriting value of the codec_mask, information on all
detected codecs is lost. No current user of hdac is impacted,
but use of bus->codec_mask is planned in future patches
for SOF.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c37c0ab029 ]
Need to chain the THINKPAD_ACPI, otherwise the mute led will not
work.
Fixes: d2cd795c4e ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200219052306.24935-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e68d669657 upstream.
TGL supports more than three iDisp DAI's.
Add support for iDisp4 CPU DAI.
Without this patch, we saw the below error on our TGL DUT:
sof_rt5682 tgl_max98357a_rt5682: ASoC: CPU DAI iDisp4 Pin not registered
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191220171037.10689-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit dc7497795e upstream.
snd_seq_check_queue() passes the current tick and time of the given
queue as a pointer to snd_seq_prioq_cell_out(), but those might be
updated concurrently by the seq timer update.
Fix it by retrieving the current tick and time via the proper helper
functions at first, and pass those values to snd_seq_prioq_cell_out()
later in the loops.
snd_seq_timer_get_cur_time() takes a new argument and adjusts with the
current system time only when it's requested so; this update isn't
needed for snd_seq_check_queue(), as it's called either from the
interrupt handler or right after queuing.
Also, snd_seq_timer_get_cur_tick() is changed to read the value in the
spinlock for the concurrency, too.
Reported-by: syzbot+fd5e0eaa1a32999173b2@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/20200214111316.26939-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bb51e669fa upstream.
The queue flags are represented in bit fields and the concurrent
access may result in unexpected results. Although the current code
should be mostly OK as it's only reading a field while writing other
fields as KCSAN reported, it's safer to cover both with a proper
spinlock protection.
This patch fixes the possible concurrent read by protecting with
q->owner_lock. Also the queue owner field is protected as well since
it's the field to be protected by the lock itself.
Reported-by: syzbot+65c6c92d04304d0a8efc@syzkaller.appspotmail.com
Reported-by: syzbot+e60ddfa48717579799dd@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/20200214111316.26939-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d152088978 upstream.
If the imx-sdma driver is built as a module, the fsl-sai device doesn't
disable on probing failure, which causes the warning in the next probing:
==================================================================
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
fsl-sai 308a0000.sai: Unbalanced pm_runtime_enable!
==================================================================
Disabling the device properly fixes the issue.
Fixes: 812ad463e0 ("ASoC: fsl_sai: Add support for runtime pm")
Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Link: https://lore.kernel.org/r/20200205160436.3813642-1-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9437bfda00 upstream.
The ssc audio driver can call into both pdc and dma backends. With the
latest rework, the logic to do this in a safe way avoiding link errors
was removed, bringing back link errors that were fixed long ago in commit
061981ff8c ("ASoC: atmel: properly select dma driver state") such as
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register'
Fix it this time using Makefile hacks and a comment to prevent this
from accidentally getting removed again rather than Kconfig hacks.
Fixes: 1829141055 ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200130130545.31148-1-codrin.ciubotariu@microchip.com
Reviewed-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 96781fd941 upstream.
Use the correct mask for this two-bit field. This fixes setting the DAI
data format to RIGHT_J or DSP_A.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Cc: stable@kernel.org
Link: https://lore.kernel.org/r/20200217064250.15516-7-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b6570fdb96 upstream.
The PCM runtime was freed during PMU in the case that the event hook
encountered an error. However, it is also unconditionally freed during
PMD. Avoid a double-free by dropping the call to kfree in the PMU hook.
Fixes: a72706ed82 ("ASoC: codec2codec: remove ephemeral variables")
Cc: stable@vger.kernel.org
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200213061147.29386-2-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 44eeb081b8 upstream.
Some code in HD-audio driver calls snprintf() in a loop and still
expects that the return value were actually written size, while
snprintf() returns the expected would-be length instead. When the
given buffer limit were small, this leads to a buffer overflow.
Use scnprintf() for addressing those issues. It returns the actually
written size unlike snprintf().
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200218091409.27162-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit b81cbf7abf ]
With firmware 2.82 Line6 changed the usb id of some of the Helix
devices but the quirks is still needed.
Add it to the quirk list for line6 helix family of devices.
Thanks to Jens for pointing out the missing ids.
Signed-off-by: Nicola Lunghi <nick83ola@gmail.com>
Link: https://lore.kernel.org/r/20200125150917.5040-1-nick83ola@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ef7d84caa5 ]
Lenovo Thinkpad T420s uses the same codec as T420, so apply the
same quirk to enable audio output on a docking station.
Signed-off-by: Peter Große <pegro@friiks.de>
Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2928fa0a97 ]
The initial snd_hda_get_sub_node() can fail on certain
devices (e.g. some Chromebook models using Intel GLK).
The failure rate is very low, but as this is is part of
the probe process, end-user impact is high.
In observed cases, related hardware status registers have
expected values, but the node query still fails. Retrying
the node query does seem to help, so fix the problem by
adding retry logic to the query. This does not impact
non-Intel platforms.
BugLink: https://github.com/thesofproject/linux/issues/1642
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a6947c9d86 ]
With fourth pin added for iDisp for skl_dai, update SOF_SKL_DAI_NUM to
account for the change. Without this, dais from the bottom of the list
are skipped. In current state that's the case for 'Alt Analog CPU DAI'.
Fixes: ac42b142cd76 ("ASoC: SOF: Intel: hda: Add iDisp4 DAI")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113114054.9716-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c249177944 ]
This fixes crackling sound during playback.
Further note: MOTU is known for reusing Product IDs for different
devices or different generations of the device (e.g. MicroBook
I/II/IIc shares a single Product ID). This patch was only tested with
M4 audio interface, but the same Product ID is also used by M2. Hope
it will work for M2 as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a3afa29942 ]
We need to unlock before we returning on this error path.
Fixes: 73ac9f5e5b ("ALSA: usb-audio: Add boot quirk for MOTU M Series")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200115174604.rhanfgy4j3uc65cx@kili.mountain
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d873997192 ]
Fix GCC warning with W=1, previous cleanup did not remove unnecessary
variable.
sound/soc/sof/intel/hda-dai.c: In function ‘hda_link_pcm_prepare’:
sound/soc/sof/intel/hda-dai.c:265:31: warning: variable ‘hda_stream’
set but not used [-Wunused-but-set-variable]
265 | struct sof_intel_hda_stream *hda_stream;
| ^~~~~~~~~~
Fixes: a3ebccb52e ("ASoC: SOF: Intel: hda: reset link DMA state in prepare")
Cc: 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/20200113205620.27285-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8fea78029f ]
If CONFIG_SND_ATMEL_SOC_DMA=m, build error:
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
(.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register'
Function atmel_pcm_dma_platform_register is defined under
CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in
CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Chen Zhou <chenzhou10@huawei.com>
Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 73ac9f5e5b ]
Add delay to make sure that audio urbs are not sent too early.
Otherwise the device hangs. Windows driver makes ~2s delay, so use
about the same time delay value.
snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which
is an overkill. Thus a quirk that is called only once is implemented.
Also send two vendor-specific control messages before and after
the delay. This behaviour is blindly copied from the Windows driver.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d4b74e218a ]
Some members of the Google_Hatch family include a rt5682 jack codec, but
no speaker amplifier. This uses the same driver (sof_rt5682) as a
combination of rt5682 jack codec and max98357a speaker amplifier. Within
the sof_rt5682 driver, these cases are not currently distinguishable,
relying on a DMI quirk to decide the configuration. This causes an
incorrect configuration when only the rt5682 is present on a
Google_Hatch device.
For CML, the jack codec is used as the primary key when matching,
with a possible speaker amplifier described in quirk_data. The two cases
of interest are the second and third 10EC5682 entries in
snd_soc_acpi_intel_cml_machines[]. The second entry matches the
combination of rt5682 and max98357a, resulting in the quirk_data field
in the snd_soc_acpi_mach being non-null, pointing at
max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker
amplifier. The third entry matches just the rt5682, resulting in a null
quirk_data.
The sof_rt5682 driver's DMI data matching identifies that a speaker
amplifier is present for all Google_Hatch family devices. Detect cases
where there is no speaker amplifier by checking for a null quirk_data in
the snd_soc_acpi_mach and remove the speaker amplifier bit in that case.
Signed-off-by: Sam McNally <sammc@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200103124921.v3.1.Ib87c4a7fbb3fc818ea12198e291b87dc2d5bc8c2@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f1dd4795b1 ]
A long-standing compile warning was seen during build test:
sound/sh/aica.c: In function 'load_aica_firmware':
sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
Fixes: 198de43d75 ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device")
Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5fab582967 ]
Dell E7xx laptops have also mic mute LED that is driven by the
dell-laptop platform driver. Bind it with the capture control as
already done for other models.
A caveat is that the fixup hook for the mic mute LED has to be applied
at last, otherwise it results in the invalid override of the callback.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205529
Link: https://lore.kernel.org/r/20200105081119.21396-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5da116f164 ]
Remove unused variables that are left over after the conversion of new
PCM ops:
sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime'
sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime'
sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime'
Fixes: 1cc2f8ba0b ("ALSA: sh: Convert to the new PCM ops")
Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit df4654bd6e ]
Clang warns:
../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation;
statement is not part of the previous 'if' [-Wmisleading-indentation]
info->version = USX2Y_DRIVER_VERSION;
^
../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here
if (us428->chip_status & USX2Y_STAT_CHIP_INIT)
^
1 warning generated.
This warning occurs because there is a space before the tab on this
line. Remove it so that the indentation is consistent with the Linux
kernel coding style and clang no longer warns.
This was introduced before the beginning of git history so no fixes tag.
Link: https://github.com/ClangBuiltLinux/linux/issues/831
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6605f0ca3a ]
This patch adds the driver data and updates quirk info
for tgl with max98357a speaker amp and ALC5682 headset codec.
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191126143205.21987-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c68e07970e ]
TGL supports one more HDMI DAI than previous models.
So add quirk support for number of HDMI DAI's.
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191126143205.21987-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d61fe22c2a ]
A design of ALSA control core allows applications to execute three
operations for TLV feature; read, write and command. Furthermore, it
allows driver developers to process the operations by two ways; allocated
array or callback function. In the former, read operation is just allowed,
thus developers uses the latter when device driver supports variety of
models or the target model is expected to dynamically change information
stored in TLV container.
The core also allows applications to lock any element so that the other
applications can't perform write operation to the element for element
value and TLV information. When the element is locked, write and command
operation for TLV information are prohibited as well as element value.
Any read operation should be allowed in the case.
At present, when an element has callback function for TLV information,
TLV read operation returns EPERM if the element is locked. On the
other hand, the read operation is success when an element has allocated
array for TLV information. In both cases, read operation is success for
element value expectedly.
This commit fixes the bug. This change can be backported to v4.14
kernel or later.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 9f35a31283 upstream.
It should be safe to ignore clock validity check result if the following
conditions are met:
- only one single sample rate is supported;
- the terminal is directly connected to the clock source;
- the clock type is internal.
This is to deal with some Denon DJ controllers that always reports that
clock is invalid.
Tested-by: Tobias Oszlanyi <toszlanyi@yahoo.de>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200212235450.697348-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 93f9d1a4ac upstream.
The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate,
but it returns the rate in byte-reversed order.
When setting sampling rate, the driver produces these warning messages:
[168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100
[168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000
[168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000
As can be seen from the hexadecimal conversion, the current rate read
back is byte-reversed from the rate that was set.
44100 == 0x00ac44, 4500480 == 0x44ac00
48000 == 0x00bb80, 8436480 == 0x80bb00
96000 == 0x017700, 30465 == 0x007701
Rather than implementing a new quirk to reverse the order, just skip
checking the rate to avoid spamming the log.
Signed-off-by: Arvind Sankar <nivedita@alum.mit.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200211162235.1639889-1-nivedita@alum.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d75a170fd8 upstream.
We've got a regression report about M-Audio Fast Track C400 device,
and the git bisection resulted in the commit e0ccdef926 ("ALSA:
usb-audio: Clean up check_input_term()"). This commit was about the
rewrite of the input terminal parser, and it's not too obvious from
the change what really broke. The answer is: it's the interpretation
of UAC2/3 effect units.
In the original code, UAC2 effect unit is as if through UAC1
processing unit because both UAC1 PU and UAC2/3 EU share the same
number (0x07). The old code went through a complex switch-case
fallthrough, finally bailing out in the middle:
if (protocol == UAC_VERSION_2 &&
hdr[2] == UAC2_EFFECT_UNIT) {
/* UAC2/UAC1 unit IDs overlap here in an
* uncompatible way. Ignore this unit for now.
*/
return 0;
}
... and this special handling was missing in the new code; the new
code treats UAC2/3 effect unit as if it were equivalent with the
processing unit.
Actually, the old code was too confusing. The effect unit has an
incompatible unit description with the processing unit, so we
shouldn't have dealt with EU in the same way.
This patch addresses the regression by changing the effect unit
handling to the own parser function. The own parser function makes
the clear distinct with PU, so it improves the readability, too.
The EU parser just sets the type and the id like the old kernels.
Once when the proper effect unit support is added, we can revisit this
parser function, but for now, let's keep this simple setup as is.
Fixes: e0ccdef926 ("ALSA: usb-audio: Clean up check_input_term()")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147
Link: https://lore.kernel.org/r/20200211160521.31990-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit acbf27746e ]
Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST
determine the order in which FE DAI and BE DAI are triggered.
In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is
triggered before the BE DAI and in the case of
SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before
the FE DAI. And this order remains the same irrespective of the
trigger command.
In the case of the SOF driver, during playback, the FW
expects the BE DAI to be triggered before the FE DAI during
the START trigger. The BE DAI trigger handles the starting of
Link DMA and so it must be started before the FE DAI is started
to prevent xruns during pause/release. This can be addressed
by setting the trigger order for the FE dai link to
SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger,
the FW expects the FE DAI to be triggered before the BE DAI.
Retaining the same order during the START and STOP commands,
results in FW error as the DAI component in the FW is still
active.
The issue can be fixed by mirroring the trigger order of
FE and BE DAI's during the START and STOP trigger. So, with the
trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be
trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME
and the BE DAI will be triggered first during the
STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order
set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered
first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands
and the FE DAI will be triggered first during the
SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191104224812.3393-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e19ecbf105 upstream.
Comparing the voltage of VDDA and VDDIO to determine whether or not to
enable VDDC manual override is insufficient. This is a problem in case
the VDDA is supplied from different regulator than VDDIO, while both
report the same voltage to the regulator framework. In that case where
VDDA and VDDIO is supplied by different regulators, the VDDC manual
override must not be applied.
Fixes: b6319b061b ("ASoC: sgtl5000: Fix charge pump source assignment")
Signed-off-by: Marek Vasut <marex@denx.de>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Igor Opaniuk <igor.opaniuk@toradex.com>
Cc: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Link: https://lore.kernel.org/r/20191220164450.1395038-2-marex@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 864cee90d4 upstream.
On TODDR sm1, the fifo threshold register field is slightly different
compared to the other SoCs. This leads to the fifo A being flushed to
memory every 8kB. If the period is smaller than that, several periods
are pushed to memory and notified at once. This is not ideal.
Fix the register field update. With this, the fifos are flushed every
128B. We could still do better, like adapt the threshold depending on
the period size, but at least it consistent across the different
SoC/fifos
Fixes: 5ac825c3d8 ("ASoC: meson: axg-toddr: add sm1 support")
Reported-by: Alden DSouza <aldend@google.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20191218172420.1199117-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 410e5e55c9 ]
The initial intent of releasing resources in the .remove does not work
well with HDaudio codecs. If the probe_continue() fails in a work
queue, e.g. due to missing firmware or authentication issues, we don't
release any resources, and as a result the kernel oopses during
suspend operations.
The suggested fix is to release all resources during errors in
probe_continue(), and use fw_state to track resource allocation
state, so that .remove does not attempt to release the same
hardware resources twice. PM operations are also modified so that
no action is done if DSP resources have been freed due to
an error at probe.
Reported-by: Takashi Iwai <tiwai@suse.de>
Co-developed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Bugzilla: http://bugzilla.suse.com/show_bug.cgi?id=1161246
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200124213625.30186-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6ca5cecbd1 ]
Add a state machine for FW boot to track the
different stages of FW boot and replace the boot_complete
field with fw_state field in struct snd_sof_dev.
This will be used to determine the actions to be performed
during system suspend.
One of the main motivations for adding this change is the
fact that errors during the top-level SOF device probe cannot
be propagated and therefore suspending the SOF device normally
during system suspend could potentially run into errors.
For example, with the current flow, if the FW boot failed
for some reason and the system suspends, the SOF device
suspend could fail because the CTX_SAVE IPC would be attempted
even though the FW never really booted successfully causing it
to time out. Another scenario that the state machine fixes
is when the runtime suspend for the SOF device fails and
the DSP is powered down nevertheless, the CTX_SAVE IPC during
system suspend would timeout because the DSP is already
powered down.
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191218002616.7652-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4d024fe8f8 upstream.
It turned out that the recent simplification of HD-audio bus access
helpers caused a regression on the virtual HD-audio device on QEMU
with ARM platforms. The driver got a CORB/RIRB timeout and couldn't
probe any codecs.
The essential difference that caused a problem was the enforced
aligned MMIO accesses by simplification. Since snd-hda-tegra driver
is enabled on ARM, it enables CONFIG_SND_HDA_ALIGNED_MMIO, which makes
the all HD-audio drivers using the aligned MMIO accesses. While this
is mandatory for snd-hda-tegra, it seems that snd-hda-intel on ARM
gets broken by this access pattern.
For addressing the regression, this patch introduces a new flag,
aligned_mmio, to hdac_bus object, and applies the aligned MMIO only
when this flag is set. This change affects only platforms with
CONFIG_SND_HDA_ALIGNED_MMIO set, i.e. mostly only for ARM platforms.
Unfortunately the patch became a big bigger than it should be, just
because the former calls didn't take hdac_bus object in the argument,
hence we had to extend the call patterns.
Fixes: 19abfefd4c ("ALSA: hda: Direct MMIO accesses")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1161152
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200120104127.28985-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2acf25f13e upstream.
The loop termination for iterating over all formats should contain
SNDRV_PCM_FORMAT_LAST, not less than it.
Fixes: 9b151fec13 ("ALSA: dummy - Add debug proc file")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200201080530.22390-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d8f489355c upstream.
The Scarlett gen2 mixer quirk code defines a few record types to
communicate via USB hub, and those must be all little-endian.
This patch changes the field types to LE to annotate endianess
properly. It also fixes the incorrect usage of leXX_to_cpu() in a
couple of places, which was caught by sparse after this change.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200201080530.22390-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f8e5f90b3a upstream.
I overlooked that some fields are words and need the converts from
LE in the recently added USB descriptor validation code.
This patch fixes those with the proper macro usages.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200201080530.22390-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8ce1cbd6ce ]
The code which checks the return value for snd_soc_add_dai_link() call
in soc_tplg_fe_link_create() moved the snd_soc_add_dai_link() call before
link->dobj members initialization.
While it does not affect the latest kernels, the old soc-core.c code
in the stable kernels is affected. The snd_soc_add_dai_link() function uses
the link->dobj.type member to check, if the link structure is valid.
Reorder the link->dobj initialization to make things work again.
It's harmless for the recent code (and the structure should be properly
initialized before other calls anyway).
The problem is in stable linux-5.4.y since version 5.4.11 when the
upstream commit 76d2703649 was applied.
Fixes: 76d2703649 ("ASoC: topology: Check return value for snd_soc_add_dai_link()")
Cc: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200122190752.3081016-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2c63bea714 ]
In case system has multiple HDA controllers, it can happen that
same HDA codec driver is used for codecs of multiple controllers.
In this case, SOF may fail to probe the HDA driver and SOF
initialization fails.
SOF HDA code currently relies that a call to request_module() will
also run device matching logic to attach driver to the codec instance.
However if driver for another HDA controller was already loaded and it
already loaded the HDA codec driver, this breaks current logic in SOF.
In this case the request_module() SOF does becomes a no-op and HDA
Codec driver is not attached to the codec instance sitting on the HDA
bus SOF is controlling. Typical scenario would be a system with both
external and internal GPUs, with driver of the external GPU loaded
first.
Fix this by adding similar logic as is used in legacy HDA driver
where an explicit device_attach() call is done after request_module().
Also add logic to propagate errors reported by device_attach() back
to caller. This also works in the case where drivers are not built
as modules.
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/20200110235751.3404-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 89b71b3f02 ]
The rt5640->jack is NULL if jack is already disabled at the time of
driver's module unloading.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20200106014707.11378-1-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d64ebdbfd4 ]
We have a new Dell machine which needs to apply the quirk
ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table
to fix it this time. And we could remove all pintbls of alc236
for applying DELL1_MIC_NO_PRESENCE on Dell machines.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20191121022644.8078-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit adebb11139 ]
As of commit 648e921888 ("clk: x86: Stop marking clocks as
CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock
it's using for the codec's mclk. It does this from commit 7735bce05a
("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling
pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for
the codec mclk, resulting in white noise with some digital microphones.
Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0
instead - mirroring the changes made to cht_bsw_max98090_ti in
commit a182ecd380 ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for
boards using pmc_plt_clk_0") and making use of the existing
dmi_check_system() call and related infrastructure added in
commit 22af29114e ("ASoC: Intel: cht-bsw-rt5645: add quirks for
SSP0/AIF1/AIF2 routing").
Signed-off-by: Sam McNally <sammc@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190917054933.209335-1-sammc@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dd836ddf4e ]
remove_link() is currently calling snd_soc_remove_dai_link() after
it has already freed the memory for the link name. But this is later
read from snd_soc_get_pcm_runtime() causing a KASAN use-after-free
warning. Reorder the cleanups to fix this issue.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20191204210447.11701-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 77fffa7422 ]
The driver forgets to call pm_runtime_disable in probe failure
and remove.
Add the missed calls to fix it.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20191203111303.12933-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 921162c81a ]
When the PCM_PARAM IPC fails while configuring the FE, the kernel
oopses in the HDaudio link DMA .hw_free operation. The root cause is a
NULL dma_data since the BE .hw_params was never called by the SOC
core.
This error can also happen if the HDaudio link DMA configuration IPC
fails in the BE .hw_params.
This patches makes sure the dma_data is properly saved in .hw_params,
and tested before being use in hw_free.
GitHub issue: https://github.com/thesofproject/linux/issues/1417
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-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/20191218000518.5830-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b06e46427f ]
If sof_machine_check() fails during driver probe, the IPC
state is not initialized and this will lead to a NULL
dereference at driver unload. Example log is as follows:
[ 1535.980630] sof-audio-pci 0000:00:1f.3: error: no matching ASoC machine driver found - aborting probe
[ 1535.980631] sof-audio-pci 0000:00:1f.3: error: failed to get machine info -19
[ 1535.980632] sof-audio-pci 0000:00:1f.3: error: sof_probe_work failed err: -19
[ 1550.798373] BUG: kernel NULL pointer dereference, address: 0000000000000008
...
[ 1550.798393] Call Trace:
[ 1550.798397] snd_sof_ipc_free+0x15/0x30 [snd_sof]
[ 1550.798399] snd_sof_device_remove+0x29/0xa0 [snd_sof]
[ 1550.798400] sof_pci_remove+0x10/0x30 [snd_sof_pci]
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/20191218000518.5830-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5d1b71226d upstream.
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae5 ("ALSA: snd-usb: add support for implicit feedback")
Fixes: ca10a7ebdf ("ALSA: usb-audio: FT C400 sync playback EP to capture EP")
Fixes: 5e35dc0338 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204")
Fixes: 17f08b0d9a ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II")
Fixes: 103e962564 ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk")
Cc: stable <stable@vger.kernel.org> # 3.5
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 747d1f076d upstream.
ALSA firewire-tascam driver can bring corruption due to spin lock without
restoration of IRQ flag in SoftIRQ context. This commit fixes the bug.
Cc: Scott Bahling <sbahling@suse.com>
Cc: <stable@vger.kernel.org> # v4.21
Fixes: d716742243 ("ALSA: firewire-tascam: queue events for change of control surface")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113085719.26788-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 60adcfde92 upstream.
snd_seq_info_timer_read() reads the information of the timer assigned
for each queue, but it's done in a racy way which may lead to UAF as
spotted by syzkaller.
This patch applies the missing q->timer_mutex lock while accessing the
timer object as well as a slight code change to adapt the standard
coding style.
Reported-by: syzbot+2b2ef983f973e5c40943@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200115203733.26530-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 869bced7a0 upstream.
When a quirk for the Irbis NB41 netbook was added, to override the defaults
for this device, I forgot to add/keep the BYT_CHT_ES8316_SSP0 part of the
defaults, completely breaking audio on this netbook.
This commit adds the BYT_CHT_ES8316_SSP0 flag to the Irbis NB41 netbook
quirk, making audio work again.
Cc: stable@vger.kernel.org
Cc: russianneuromancer@ya.ru
Fixes: aa2ba991c4 ("ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook")
Reported-and-tested-by: russianneuromancer@ya.ru
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/20200106113903.279394-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 057efcf9fa upstream.
MIC BIAS Internal1 is broken at the moment because we always
enable the internal rbias resistor to the TX2 line (connected to
the headset microphone), rather than enabling the resistor connected
to TX1.
Move the RBIAS code to pm8916_wcd_analog_enable_micbias_int1/2()
to fix this.
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200111164006.43074-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e0beec8839 upstream.
MIC BIAS External1 sets pm8916_wcd_analog_enable_micbias_ext1()
as event handler, which ends up in pm8916_wcd_analog_enable_micbias_ext().
But pm8916_wcd_analog_enable_micbias_ext() only handles the POST_PMU
event, which is not specified in the event flags for MIC BIAS External1.
This means that the code in the event handler is never actually run.
Set SND_SOC_DAPM_POST_PMU as the only event for the handler to fix this.
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200111164006.43074-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Takashi Iwai