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[ Upstream commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b ]
If the DAI format setup fails, there is no valid communication format
between CPU and CODEC, so fail card instantiation, rather than continue
with a card that will most likely not function properly.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 8dd26dff00c0636b1d8621acaeef3f6f3a39dd77 ]
DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a
list of the widgets connected to a specific front end DAI so it
can search through this list for available back end DAIs. The
custom_stop_condition was added to is_connected_ep to facilitate this
list not containing more widgets than is necessary. Doing so both
speeds up the DPCM handling as less widgets need to be searched and
avoids issues with CODEC to CODEC links as these would be confused
with back end DAIs if they appeared in the list of available widgets.
custom_stop_condition was implemented by aborting the graph walk
when the condition is triggered, however there is an issue with this
approach. Whilst walking the graph is_connected_ep should update the
endpoints cache on each widget, if the walk is aborted the number
of attached end points is unknown for that sub-graph. When the stop
condition triggered, the original patch ignored the triggering widget
and returned zero connected end points; a later patch updated this
to set the triggering widget's cache to 1 and return that. Both of
these approaches result in inaccurate values being stored in various
end point caches as the values propagate back through the graph,
which can result in later issues with widgets powering/not powering
unexpectedly.
As the original goal was to reduce the size of the widget list passed
to the DPCM code, the simplest solution is to limit the functionality
of the custom_stop_condition to the widget list. This means the rest
of the graph will still be processed resulting in correct end point
caches, but only widgets up to the stop condition will be added to the
returned widget list.
Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets")
Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links")
Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 401714d9534aad8c24196b32600da683116bbe09 upstream.
We have 3 new lenovo laptops which have conexant codec 0x14f11f86,
these 3 laptops also have the noise issue when rebooting, after
letting the codec enter D3 before rebooting or poweroff, the noise
disappers.
Instead of adding a new ID again in the reboot_notify(), let us make
this function apply to all conexant codec. In theory make codec enter
D3 before rebooting or poweroff is harmless, and I tested this change
on a couple of other Lenovo laptops which have different conexant
codecs, there is no side effect so far.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 871b9066027702e6e6589da0e1edd3b7dede7205 upstream.
Make codec enter D3 before rebooting or poweroff can fix the noise
issue on some laptops. And in theory it is harmless for all codecs
to enter D3 before rebooting or poweroff, let us add a generic
reboot_notify, then realtek and conexant drivers can call this
function.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cfef67f016e4c00a2f423256fc678a6967a9fc09 upstream.
In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc().
Then, the pin widgets in 'codec' are parsed. However, if the parsing
process fails, 'spec' is not deallocated, leading to a memory leak.
To fix the above issue, free 'spec' before returning the error.
Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit de768ce45466f3009809719eb7b1f6f5277d9373 upstream.
MSI MPG X570 board is with another AMD HD-audio controller (PCI ID
1022:1487) and it requires the same workaround applied for X370, etc
(PCI ID 1022:1457).
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db upstream.
A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:
- Set up the proper driver caps for this controller, similar as the
other AMD controller.
- Correct the DMA position reporting with the fixed FIFO size, which
is similar like as workaround used for VIA chip set.
- Even after the position correction, PulseAudio still shows
mysterious stalls of playback streams when a capture is triggered in
timer-scheduled mode. Since we have no clear way to eliminate the
stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
a temporary workaround.
This patch implements the workarounds. For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.
Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c1c6c877b0c79fd7e05c931435aa42211eaeebaf upstream.
The commit bfcba288b97f ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object. This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.
Fix the bug by setting the PCM hw info flag locally.
Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1be3c1fae6c1e1f5bb982b255d2034034454527a upstream.
In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.
To fix the above issue, free 'b->packets' before returning the error code.
Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v2.6.39+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 3b8179944cb0dd53e5223996966746cdc8a60657 ]
Draining makes little sense in the situation of hardware overrun, as the
hardware will have consumed all its available samples. Additionally,
draining whilst the stream is paused would presumably get stuck as no
data is being consumed on the DSP side.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit a70ab8a8645083f3700814e757f2940a88b7ef88 ]
Partial drain and next track are intended for gapless playback and
don't really have an obvious interpretation for a capture stream, so
makes sense to not allow those operations on capture streams.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 26c3f1542f5064310ad26794c09321780d00c57d ]
Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call
snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which
allow a transition to SNDRV_PCM_STATE_SETUP. The stream should
only be able to move to the setup state once it has received a
SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing
those ioctls whilst in the open state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 4475f8c4ab7b248991a60d9c02808dbb813d6be8 ]
A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.
To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.
Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit c7cd7c748a3250ca33509f9235efab9c803aca09 upstream.
In sound_insert_unit(), the controlling structure 's' is allocated through
kmalloc(). Then it is added to the sound driver list by invoking
__sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is
removed from the list through __sound_remove_unit(). If 'index' is not less
than 0, -EBUSY is returned to indicate the error. However, 's' is not
deallocated on this execution path, leading to a memory leak bug.
To fix the above issue, free 's' before -EBUSY is returned.
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3f8809499bf02ef7874254c5e23fc764a47a21a0 upstream.
This conexant codec isn't in the supported codec list yet, the hda
generic driver can drive this codec well, but on a Lenovo machine
with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI
to make the leds work. After adding this codec to the list, the
driver patch_conexant.c will apply THINKPAD_ACPI to this machine.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 70256b42caaf3e13c2932c2be7903a73fbe8bb8b upstream.
Commit 7b9584fa1c0b ("staging: line6: Move altsetting to properties")
set a wrong altsetting for LINE6_PODHD500_1 during refactoring.
Set the correct altsetting number to fix the issue.
BugLink: https://bugs.launchpad.net/bugs/1790595
Fixes: 7b9584fa1c0b ("staging: line6: Move altsetting to properties")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 4b4e0e32e4b09274dbc9d173016c1a026f44608c upstream.
Without this patch, the headset-mic and headphone-mic don't work.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ede34f397ddb063b145b9e7d79c6026f819ded13 upstream.
The fix for the racy writes and ioctls to sequencer widened the
application of client->ioctl_mutex to the whole write loop. Although
it does unlock/relock for the lengthy operation like the event dup,
the loop keeps the ioctl_mutex for the whole time in other
situations. This may take quite long time if the user-space would
give a huge buffer, and this is a likely cause of some weird behavior
spotted by syzcaller fuzzer.
This patch puts a simple workaround, just adding a mutex break in the
loop when a large number of events have been processed. This
shouldn't hit any performance drop because the threshold is set high
enough for usual operations.
Fixes: 7bd800915677 ("ALSA: seq: More protection for concurrent write and ioctl races")
Reported-by: syzbot+97aae04ce27e39cbfca9@syzkaller.appspotmail.com
Reported-by: syzbot+4c595632b98bb8ffcc66@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ceaea851b9ea75f9ea2bbefb53ff0d4b27cd5a6e upstream.
Back in ff9fb72bc07705c (debugfs: return error values, not NULL) the
debugfs APIs were changed to return error pointers rather than NULL
pointers on error, breaking the error checking in ASoC. Update the
code to use IS_ERR() and log the codes that are returned as part of
the error messages.
Fixes: ff9fb72bc07705c (debugfs: return error values, not NULL)
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d07a9a4f66e944fcc900812cbc2f6817bde6a43d upstream.
Dell headset mode platform with ALC236.
It doesn't recording after system resume from S3.
S3 mode was deep. s2idle was not has this issue.
S3 deep will cut of codec power. So, the register will back to default
after resume back.
This patch will solve this issue.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit bef33e19203dde434bcdf21c449e3fb4f06c2618 upstream.
On M710q Lenovo ThinkCentre machine, there are two front mics,
we change the location for one of them to avoid conflicts.
Signed-off-by: Dennis Wassenberg <dennis.wassenberg@secunet.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2acf5a3e6e9371e63c9e4ff54d84d08f630467a0 upstream.
There are a couple of left shifts of unsigned 8 bit values that
first get promoted to signed ints and hence get sign extended
on the shift if the top bit of the 8 bit values are set. Fix
this by casting the 8 bit values to unsigned ints to stop the
unintentional sign extension.
Addresses-Coverity: ("Unintended sign extension")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3450121997ce872eb7f1248417225827ea249710 upstream.
LINE6 drivers allocate the buffers based on the value returned from
usb_maxpacket() calls. The manipulated device may return zero for
this, and this results in the kmalloc() with zero size (and it may
succeed) while the other part of the driver code writes the packet
data with the fixed size -- which eventually overwrites.
This patch adds a simple sanity check for the invalid buffer size for
avoiding that problem.
Reported-by: syzbot+219f00fb49874dcaea17@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 7fbd1753b64eafe21cf842348a40a691d0dee440 upstream.
In IEC 61883-6, 8 MIDI data streams are multiplexed into single
MIDI conformant data channel. The index of stream is calculated by
modulo 8 of the value of data block counter.
In fireworks, the value of data block counter in CIP header has a quirk
with firmware version v5.0.0, v5.7.3 and v5.8.0. This brings ALSA
IEC 61883-1/6 packet streaming engine to miss detection of MIDI
messages.
This commit fixes the miss detection to modify the value of data block
counter for the modulo calculation.
For maintainers, this bug exists since a commit 18f5ed365d3f ("ALSA:
fireworks/firewire-lib: add support for recent firmware quirk") in Linux
kernel v4.2. There're many changes since the commit. This fix can be
backported to Linux kernel v4.4 or later. I tagged a base commit to the
backport for your convenience.
Besides, my work for Linux kernel v5.3 brings heavy code refactoring and
some structure members are renamed in 'sound/firewire/amdtp-stream.h'.
The content of this patch brings conflict when merging -rc tree with
this patch and the latest tree. I request maintainers to solve the
conflict to replace 'tx_first_dbc' with 'ctx_data.tx.first_dbc'.
Fixes: df075feefbd3 ("ALSA: firewire-lib: complete AM824 data block processing layer")
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c3ea60c231446663afd6ea1054da6b7f830855ca upstream.
There are two occurrances of a call to snd_seq_oss_fill_addr where
the dest_client and dest_port arguments are in the wrong order. Fix
this by swapping them around.
Addresses-Coverity: ("Arguments in wrong order")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 04268bf2757a125616b6c2140e6250f43b7b737a ]
When we call snd_soc_component_set_jack(component, NULL, NULL) we should
set rt274->jack to passed jack, so when interrupt is triggered it calls
snd_soc_jack_report(rt274->jack, ...) with proper value.
This fixes problem in machine where in register, we call
snd_soc_register(component, &headset, NULL), which just calls
rt274_mic_detect via callback.
Now when machine driver is removed "headset" will be gone, so we
need to tell codec driver that it's gone with:
snd_soc_register(component, NULL, NULL), but we also need to be able
to handle NULL jack argument here gracefully.
If we don't set it to NULL, next time the rt274_irq runs it will call
snd_soc_jack_report with first argument being invalid pointer and there
will be Oops.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit f9927000cb35f250051f0f1878db12ee2626eea1 ]
Whilst testing the capture functionality of the i2s on the newer
SoCs it was noticed that the recording was somewhat distorted.
This was due to the offset not being set correctly on the receiver
side.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 7e46169a5f35762f335898a75d1b8a242f2ae0f5 ]
Although not causing any noticeable issues, the mask for the
channel offset is covering too many bits.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 5628c8979642a076f91ee86c3bae5ad251639af0 ]
The supported formats are S16_LE and S24_LE now. However, by datasheet
of max98090, S24_LE is only supported when it is in the right justified
mode. We should remove 24-bit format if it is not in that mode to avoid
triggering error.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 5087a8f17df868601cd7568299e91c28086d2b45 ]
If playback/capture is paused and system enters S3, after system returns
from suspend, BE dai needs to call prepare() callback when playback/capture
is released from pause if RESUME_INFO flag is not set.
Currently, the dpcm_be_dai_prepare() function will block calling prepare()
if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the
following test case fail if the pcm uses BE:
playback -> pause -> S3 suspend -> S3 resume -> pause release
The playback may exit abnormally when pause is released because the BE dai
prepare() is not called.
This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in
SND_SOC_DPCM_STATE_PAUSED state.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit f3df05c805983427319eddc2411a2105ee1757cf ]
The cs4265_readable_register function stopped short of the maximum
register.
An example bug is taken from :
https://github.com/Audio-Injector/Ultra/issues/25
Where alsactl store fails with :
Cannot read control '2,0,0,C Data Buffer,0': Input/output error
This patch fixes the bug by setting the cs4265 to have readable
registers up to the maximum hardware register CS4265_MAX_REGISTER.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit fa763f1b2858752e6150ffff46886a1b7faffc82 ]
We observed the same issue as reported by commit a8d7bde23e7130686b7662
("ALSA: hda - Force polling mode on CFL for fixing codec communication")
We don't have a better solution. So apply the same workaround to CNL.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit f0654ba94e33699b295ce4f3dc73094db6209035 ]
This reverts commit feb689025fbb6f0aa6297d3ddf97de945ea4ad32.
The fix attempt was incorrect, leading to the mutex deadlock through
the close of OSS sequencer client. The proper fix needs more
consideration, so let's revert it now.
Fixes: feb689025fbb ("ALSA: seq: Protect in-kernel ioctl calls with mutex")
Reported-by: syzbot+47ded6c0f23016cde310@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 2eabc5ec8ab4d4748a82050dfcb994119b983750 ]
The snd_seq_ioctl_get_subscription() retrieves the port subscriber
information as a pointer, while the object isn't protected, hence it
may be deleted before the actual reference. This race was spotted by
syzkaller and may lead to a UAF.
The fix is simply copying the data in the lookup function that
performs in the rwsem to protect against the deletion.
Reported-by: syzbot+9437020c82413d00222d@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit feb689025fbb6f0aa6297d3ddf97de945ea4ad32 ]
ALSA OSS sequencer calls the ioctl function indirectly via
snd_seq_kernel_client_ctl(). While we already applied the protection
against races between the normal ioctls and writes via the client's
ioctl_mutex, this code path was left untouched. And this seems to be
the cause of still remaining some rare UAF as spontaneously triggered
by syzkaller.
For the sake of robustness, wrap the ioctl_mutex also for the call via
snd_seq_kernel_client_ctl(), too.
Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit b06c58c2a1eed571ea2a6640fdb85b7b00196b1e upstream.
When the output sample rate is [8kHz, 30kHz], the limitation
of the supported ratio range is [1/24, 8]. In the driver
we use (8kHz, 30kHz) instead of [8kHz, 30kHz].
So this patch is to fix this issue and the potential rounding
issue with divider.
Fixes: fff6e03c7b65 ("ASoC: fsl_asrc: add support for 8-30kHz
output sample rate")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ad6eecbfc01c987e0253371f274c3872042e4350 upstream.
Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver
support for CS42448/CS42888")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0e3fb6995bfabb23c172e8b883bf5ac57102678e upstream.
The data for isochronous resources is not destroyed in expected place.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 9b2bb4f2f4a2 ("ALSA: firewire-motu: add stream management functionality")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 717f43d81afc1250300479075952a0e36d74ded3 upstream.
ALC255 and ALC256 were some difference for hidden register.
This update was suitable for ALC256.
Fixes: e69e7e03ed22 ("ALSA: hda/realtek - ALC256 speaker noise issue")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d8fa87c368f5b4096c4746894fdcc195da285df1 upstream.
Stanton SCS.1m can transfer isochronous packet with Multi Bit Linear
Audio data channels, therefore it allows software to capture PCM
substream. However, ALSA oxfw driver doesn't.
This commit changes the driver to add one PCM substream for capture
direction.
Fixes: de5126cc3c0b ("ALSA: oxfw: add stream format quirk for SCS.1 models")
Cc: <stable@vger.kernel.org> # v4.5+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 7c32ae35fbf9cffb7aa3736f44dec10c944ca18e upstream.
The call of unsubscribe_port() which manages the group count and
module refcount from delete_and_unsubscribe_port() looks racy; it's
not covered by the group list lock, and it's likely a cause of the
reported unbalance at port deletion. Let's move the call inside the
group list_mutex to plug the hole.
Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit f495222e28275222ab6fd93813bd3d462e16d340 ]
Currently the IRQ handler in HD-audio controller driver is registered
before the chip initialization. That is, we have some window opened
between the azx_acquire_irq() call and the CORB/RIRB setup. If an
interrupt is triggered in this small window, the IRQ handler may
access to the uninitialized RIRB buffer, which leads to a NULL
dereference Oops.
This is usually no big problem since most of Intel chips do register
the IRQ via MSI, and we've already fixed the order of the IRQ
enablement and the CORB/RIRB setup in the former commit b61749a89f82
("sound: enable interrupt after dma buffer initialization"), hence the
IRQ won't be triggered in that room. However, some platforms use a
shared IRQ, and this may allow the IRQ trigger by another source.
Another possibility is the kdump environment: a stale interrupt might
be present in there, the IRQ handler can be falsely triggered as well.
For covering this small race, let's move the azx_acquire_irq() call
after hda_intel_init_chip() call. Although this is a bit radical
change, it can cover more widely than checking the CORB/RIRB setup
locally in the callee side.
Reported-by: Liwei Song <liwei.song@windriver.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 317d9313925cd8388304286c0d3c8dda7f060a2d upstream.
I measured power consumption between power_save_node=1 and power_save_node=0.
It's almost the same.
Codec will enter to runtime suspend and suspend.
That pin also will enter to D3. Don't need to enter to D3 by single pin.
So, Disable power_save_node as default. It will avoid more issues.
Windows Driver also has not this option at runtime PM.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 8ca5104715cfd14254ea5aecc390ae583b707607 ]
Building with clang shows a variable that is only used by the
suspend/resume functions but defined outside of their #ifdef block:
sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted
We commonly fix these by marking the PM functions as __maybe_unused,
but here that would grow the davinci_mcasp structure, so instead
add another #ifdef here.
Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ]
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ]
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit ddb351145a967ee791a0fb0156852ec2fcb746ba ]
is_slave_mode defaults to false because sai structure
that contains it is kzalloc'ed.
Anyhow, if we decide to set the following configuration
SAI slave -> SAI master, is_slave_mode will remain set on true
although SAI being master it should be set to false.
Fix this by updating is_slave_mode for each call of
fsl_sai_set_dai_fmt.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit ea751227c813ab833609afecfeedaf0aa26f327e ]
During randconfig builds, I occasionally run into an invalid configuration
of the freescale FIQ sound support:
WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ
Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]
Selected by [y]:
- SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y])
sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove':
imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit'
sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe':
imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init'
The Kconfig warning is a result of the symbol being defined inside of
the "if SND_IMX_SOC" block, and is otherwise harmless. The link error
is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not
imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same
time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency,
which then causes a link failure from imx-ssi.
The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever
one of its potential users is built-in.
Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 30180e8436046344b12813dc954b2e01dfdcd22d ]
If the hdmi codec startup fails, it should clear the current_substream
pointer to free the device. This is properly done for the audio_startup()
callback but for snd_pcm_hw_constraint_eld().
Make sure the pointer cleared if an error is reported.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 56df90b631fc027fe28b70d41352d820797239bb upstream.
Add patch for realtek codec in Lenovo B50-70 that fixes inverted
internal microphone channel.
Device IdeaPad Y410P has the same PCI SSID as Lenovo B50-70,
but first one is about fix the noise and it didn't seem help in a
later kernel version.
So I replaced IdeaPad Y410P device description with B50-70 and apply
inverted microphone fix.
Bugzilla: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215
Signed-off-by: Michał Wadowski <wadosm@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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