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This reverts commit 714a8438fc8ae88aa22c25065e241bce0260db13 which is
commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b upstream.
Mark Brown writes:
I nacked this patch when Sasha posted it - it only improves
diagnostics and might make systems that worked by accident break
since it turns things into a hard failure, it won't make
anything that didn't work previously work.
Reported-by: Mark Brown <broonie@kernel.org>
Cc: Ricard Wanderlof <ricardw@axis.com>
Cc: Sasha Levin <sashal@kernel.org>
Link: https://lore.kernel.org/lkml/20190904181027.GG4348@sirena.co.uk
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1a15718b41df026cffd0e42cfdc38a1384ce19f9 upstream.
Behringer UFX1604 requires the similar quirk to apply implicit fb like
another Behringer model UFX1204 in order to fix the noisy playback.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 6de3c9e3f6b3eaf66859e1379b3f35dda781416b upstream.
The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code. It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.
This patch fixes the check before the conversion.
While we're at it, correct a typo in the comment in the function,
too.
Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 75545304eba6a3d282f923b96a466dc25a81e359 upstream.
The input pool of a client might be deleted via the resize ioctl, the
the access to it should be covered by the proper locks. Currently the
only missing place is the call in snd_seq_ioctl_get_client_pool(), and
this patch papers over it.
Reported-by: syzbot+4a75454b9ca2777f35c7@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f9ef724d4896763479f3921afd1ee61552fc9836 upstream.
"enabled" parameter historically referred to the device input or
output, not to the led indicator. After the changes added with the led
helper functions the mic mute led logic refers to the led and not to
the mic input which caused led indicator to be negated.
Fixing logic in cxt_update_gpio_led and updated
cxt_fixup_gpio_mute_hook
Also updated debug messages to ease further debugging if necessary.
Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1bc8d18c75fef3b478dbdfef722aae09e2a9fde7 upstream.
I forgot to release the allocated object at the early error path in
line6_init_pcm(). For addressing it, slightly shuffle the code so
that the PCM destructor (pcm->private_free) is assigned properly
before all error paths.
Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f9f0e9ed350e15d51ad07364b4cf910de50c472a upstream.
The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a
variable size depending on both input and output pins. Its size is to
fit with input * output bits. The problem is that the input size
can't be determined simply from the unit descriptor itself but it
needs to parse the whole connected sources. Although the
uac_mixer_unit_get_channels() tries to check some possible overflow of
this bitmap, it's incomplete due to the lack of the evaluation of
input pins.
For covering possible overflows, this patch adds the bitmap overflow
check in the loop of input pins in parse_audio_mixer_unit().
Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 1e112c35e3c96db7c8ca6ddaa96574f00c06e7db ]
The slot_width is a property for the bus while the constraint for
SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format.
Applying slot_width constraint to sample_bits works most of the time, but
it will blacklist valid formats in some cases.
With slot_width 24 we can support S24_3LE and S24_LE formats as they both
look the same on the bus, but a a 24 constraint on sample_bits would not
allow S24_LE as it is stored in 32bits in memory.
Implement a simple hw_rule function to allow all formats which require less
or equal number of bits on the bus as slot_width (if configured).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 789e162a6255325325bd321ab0cd51dc7e285054 ]
This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0.
Revert "ASoC: rockchip: i2s: Support mono capture"
Previous discussion in
https://patchwork.kernel.org/patch/10147153/
explains the issue of the patch.
While device is configured as 1-ch, hardware is still
generating a 2-ch stream.
When user space reads the data and assumes it is a 1-ch stream,
the rate will be slower by 2x.
Revert the change so 1-ch is not supported.
User space can selectively take one channel data out of two channel
if 1-ch is preferred.
Currently, both channels record identical data.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 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 daac07156b330b18eb5071aec4b3ddca1c377f2c upstream.
The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.
```
struct uac_mixer_unit_descriptor {
__u8 bLength;
__u8 bDescriptorType;
__u8 bDescriptorSubtype;
__u8 bUnitID;
__u8 bNrInPins;
__u8 baSourceID[];
}
```
This patch fixes the bug by add a sanity check on the length of
the descriptor.
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 19bce474c45be69a284ecee660aa12d8f1e88f18 upstream.
`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.
This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 190d03814eb3b49d4f87ff38fef26d36f3568a60 upstream.
HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same
quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db upstream.
A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:
- Set up the proper driver caps for this controller, similar as the
other AMD controller.
- Correct the DMA position reporting with the fixed FIFO size, which
is similar like as workaround used for VIA chip set.
- Even after the position correction, PulseAudio still shows
mysterious stalls of playback streams when a capture is triggered in
timer-scheduled mode. Since we have no clear way to eliminate the
stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
a temporary workaround.
This patch implements the workarounds. For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.
Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c1c6c877b0c79fd7e05c931435aa42211eaeebaf upstream.
The commit bfcba288b97f ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object. This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.
Fix the bug by setting the PCM hw info flag locally.
Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3d92aa45fbfd7319e3a19f4ec59fd32b3862b723 upstream.
In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.
To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.
Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1be3c1fae6c1e1f5bb982b255d2034034454527a upstream.
In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.
To fix the above issue, free 'b->packets' before returning the error code.
Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v2.6.39+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a67060201b746a308b1674f66bf289c9faef6d09 upstream.
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 3b8179944cb0dd53e5223996966746cdc8a60657 ]
Draining makes little sense in the situation of hardware overrun, as the
hardware will have consumed all its available samples. Additionally,
draining whilst the stream is paused would presumably get stuck as no
data is being consumed on the DSP side.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit a70ab8a8645083f3700814e757f2940a88b7ef88 ]
Partial drain and next track are intended for gapless playback and
don't really have an obvious interpretation for a capture stream, so
makes sense to not allow those operations on capture streams.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 26c3f1542f5064310ad26794c09321780d00c57d ]
Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call
snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which
allow a transition to SNDRV_PCM_STATE_SETUP. The stream should
only be able to move to the setup state once it has received a
SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing
those ioctls whilst in the open state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 4475f8c4ab7b248991a60d9c02808dbb813d6be8 ]
A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.
To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.
Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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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 74bf71ed792ab0f64631cc65ccdb54c356c36d45 upstream.
Distribution installation images such as Debian include different sets
of modules which can be downloaded dynamically. Such images may notably
include the hda sound modules but not the i915 DRM module, even if the
latter was enabled at build time, as reported on
https://bugs.debian.org/931507
In such a case hdac_i915 would be linked in and try to load the i915
module, fail since it is not there, but still wait for a whole minute
before giving up binding with it.
This fixes such as case by only waiting for the binding if the module
was properly loaded (or module support is disabled, in which case i915
is already compiled-in anyway).
Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Signed-off-by: Samuel Thibault <samuel.thibault@ens-lyon.org>
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 607975b30db41aad6edc846ed567191aa6b7d893 upstream.
put_device will call ac97_codec_release to free
ac97_codec_device and other resources, so remove the kfree
and other redundant code.
Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus")
Signed-off-by: Ding Xiang <dingxiang@cmss.chinamobile.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 fbc571290d9f7bfe089c50f4ac4028dd98ebfe98 upstream.
It assigned to wrong model. So, The headphone Mic can't work.
Fixes: 3f640970a414 ("ALSA: hda - Fix headset mic detection problem for several Dell laptops")
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 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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[ Upstream commit 0f6ff78540bd1b4df1e0f17806b0ce2e1dff0d78 ]
When we unload Skylake driver we may end up calling
hdac_component_master_unbind(), it uses acomp->audio_ops, which we set
in hdmi_codec_probe(), so we need to set it to NULL in hdmi_codec_remove(),
otherwise we will dereference no longer existing pointer.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@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 cb36ff785e868992e96e8b9e5a0c2822b680a9e2 ]
The content of SND_SOC_DAIFMT_FORMAT_MASK is a number, not a bitfield,
so the test to check if the format is i2s is wrong. Because of this the
clock setting may be wrong. For example, the sample clock gets inverted
in DSP B mode, when it should not.
Fix the lrclk invert helper function
Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver")
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 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 ca95c7bf3d29716916baccdc77c3c2284b703069 upstream.
Extension Unit (XU) is used to have a compatible layout with
Processing Unit (PU) on UAC1, and the usb-audio driver code assumed it
for parsing the descriptors. Meanwhile, on UAC2, XU became slightly
incompatible with PU; namely, XU has a one-byte bmControls bitmap
while PU has two bytes bmControls bitmap. This incompatibility
results in the read of a wrong address for the last iExtension field,
which ended up with an incorrect string for the mixer element name, as
recently reported for Focusrite Scarlett 18i20 device.
This patch corrects this misalignment by introducing a couple of new
macros and calling them depending on the descriptor type.
Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Reported-by: Stefan Sauer <ensonic@hora-obscura.de>
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 183ab39eb0ea9879bb68422a83e65f750f3192f0 ]
The recent commit 98081ca62cba ("ALSA: hda - Record the current power
state before suspend/resume calls") made the HD-audio driver to store
the PM state in power_state field. This forgot, however, the
initialization at power up. Although the codec drivers usually don't
need to refer to this field in the normal operation, let's initialize
it properly for consistency.
Fixes: 98081ca62cba ("ALSA: hda - Record the current power state before suspend/resume calls")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.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 503d90b30602a3295978e46d844ccc8167400fe6 upstream.
This adds 4 SND_PCI_QUIRK(...) lines for several barebone models of the ODM
Clevo. The model names are written in regex syntax to describe/match all clevo
models that are similar enough and use the same PCI SSID that this fixup works
for them.
Additionally the lines regarding SSID 0x96e1 and 0x97e1 didn't fix audio for the
all our Clevo notebooks using these SSIDs (models Clevo P960* and P970*) since
ALC1220_FIXP_CLEVO_PB51ED_PINS swapped pins that are not necesarry to be
swapped. This patch initiates ALC1220_FIXUP_CLEVO_P950 instead for these model
and fixes the audio.
Fixes: 80690a276f44 ("ALSA: hda/realtek - Add quirk for Tuxedo XC 1509")
Signed-off-by: Richard Sailer <rs@tuxedocomputers.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 6d647b736a6b1cbf2f8deab0e6a94c34a6ea9d60 ]
During the integration of HDaudio support, we changed the way in which
we get hdev in snd_hdac_ext_bus_device_init() to use one preallocated
with devm_kzalloc(), however it still left kfree(hdev) in
snd_hdac_ext_bus_device_exit(). It leads to oopses when trying to
rmmod and modprobe. Fix it, by just removing kfree call.
SOF also uses some of the snd_hdac_ functions for HDAudio support but
allocated the memory with kzalloc. A matching fix is provided
separately to align all users of the snd_hdac_ library.
Fixes: 6298542fa33b ("ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@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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Greg Kroah-Hartman
Takashi Iwai
Jeronimo Borque
Peter Ujfalusi
Cheng-Yi Chiang
Ricard Wanderlof
Charles Keepax
Hui Wang
Wenwen Wang
Hui Peng
Samuel Thibault
Kai-Heng Feng
Ding Xiang
Kailang Yang
Mark Brown
Amadeusz Sławiński
Jerome Brunet
Dennis Wassenberg
Richard Sailer
Colin Ian King
Takashi Sakamoto
Marcus Cooper