A collection of small fixes as usual:
- More coverage of USB-audio descriptor sanity checks
- A fix for mute LED regression on Conexant HD-audio codecs
- A few device-specific fixes and quirks for USB-audio and HD-audio
- A fix for (die-hard remaining) possible race in sequencer core
- FireWire oxfw regression fix that was introduced in 5.3-rc1
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Merge tag 'sound-5.3-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes as usual:
- More coverage of USB-audio descriptor sanity checks
- A fix for mute LED regression on Conexant HD-audio codecs
- A few device-specific fixes and quirks for USB-audio and HD-audio
- A fix for (die-hard remaining) possible race in sequencer core
- FireWire oxfw regression fix that was introduced in 5.3-rc1"
* tag 'sound-5.3-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: oxfw: fix to handle correct stream for PCM playback
ALSA: seq: Fix potential concurrent access to the deleted pool
ALSA: usb-audio: Check mixer unit bitmap yet more strictly
ALSA: line6: Fix memory leak at line6_init_pcm() error path
ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()
ALSA: hda/ca0132 - Add new SBZ quirk
ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604
ALSA: hda - Fixes inverted Conexant GPIO mic mute led
When userspace application calls ioctl(2) to configure hardware for PCM
playback substream, ALSA OXFW driver handles incoming AMDTP stream.
In this case, outgoing AMDTP stream should be handled.
This commit fixes the bug for v5.3-rc kernel.
Fixes: 4f380d0070 ("ALSA: oxfw: configure packet format in pcm.hw_params callback")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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: 0bfe5e434e ("ALSA: usb-audio: Check mixer unit descriptors more strictly")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: 3450121997 ("ALSA: line6: Fix write on zero-sized buffer")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: 8c558076c7 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new PCI subsys ID for the SBZ, as found and tested by
me and some reddit users.
Link: https://lore.kernel.org/lkml/20190819204008.14426-1-p.rekowski@gmail.com
Signed-off-by: Paweł Rekowski <p.rekowski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
"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: 184e302b46 ("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>
All small fixes targeted for stable:
- Two fixes for USB-audio with malformed descriptor, spotted by
fuzzers
- Two fixes Conexant HD-audio codec wrt power management
- Quirks for HD-audio AMD platform and HP laptop
- HD-audio memory leak fix
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Merge tag 'sound-5.3-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All small fixes targeted for stable:
- Two fixes for USB-audio with malformed descriptor, spotted by
fuzzers
- Two fixes Conexant HD-audio codec wrt power management
- Quirks for HD-audio AMD platform and HP laptop
- HD-audio memory leak fix"
* tag 'sound-5.3-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_term
ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unit
ALSA: hda - Add a generic reboot_notify
ALSA: hda - Let all conexant codec enter D3 when rebooting
ALSA: hda/realtek - Add quirk for HP Envy x360
ALSA: hda - Fix a memory leak bug
ALSA: hda - Apply workaround for another AMD chip 1022:1487
`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>
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>
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>
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>
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>
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: 352f7f914e ("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>
Lots of small fixes at this time since we've received the ASoC
fix batch now.
- Some coverage in ASoC core mostly for minor issues like NULL
checks for DPCM and proper error handling in DAI instantiation
- A collection of small device-specific changes in various ASoC
codec and platform drivers
- OF-tree refcount fixes in a few ASoC drivers
- Fixes of memory leaks in the error paths of various ASoC / ALSA
drivers
- A workaround for a long-standing issue on AMD HD-audio device
- Updates of MAINTAINERS, mail addresses, file permission fixups
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Merge tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Lots of small fixes at this time since we've received the ASoC fix
batch now.
- Some coverage in ASoC core mostly for minor issues like NULL checks
for DPCM and proper error handling in DAI instantiation
- A collection of small device-specific changes in various ASoC codec
and platform drivers
- OF-tree refcount fixes in a few ASoC drivers
- Fixes of memory leaks in the error paths of various ASoC / ALSA
drivers
- A workaround for a long-standing issue on AMD HD-audio device
- Updates of MAINTAINERS, mail addresses, file permission fixups"
* tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (38 commits)
ALSA: firewire: fix a memory leak bug
sound: fix a memory leak bug
ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
ALSA: hiface: fix multiple memory leak bugs
ALSA: hda - Don't override global PCM hw info flag
ALSA: usb-audio: fix a memory leak bug
ASoC: max98373: Remove executable bits
ASoC: amd: acp3x: use dma address for acp3x dma driver
ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
ASoC: max98373: add 88200 and 96000 sampling rate support
ASoC: sun4i-i2s: Incorrect SR and WSS computation
MAINTAINERS: Update Intel ASoC drivers maintainers
ASoC: ti: davinci-mcasp: Correct slot_width posed constraint
ASoC: rockchip: Fix mono capture
ASoC: Intel: Fix some acpi vs apci typo in somme comments
ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode
ASoC: Fail card instantiation if DAI format setup fails
ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread
ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
ASoC: dapm: fix a memory leak bug
...
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>
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: 31ef9134eb ("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>
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>
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>
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: a91c3fb2f8 ("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>
The commit bfcba288b9 ("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: bfcba288b9 ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: 7edf3b5e6a ("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>
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
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Merge tag 'asoc-fix-v5.3-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.3
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
This patch fixes page faults when IOMMU is enabled.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD platform device acp3x_rv_i2s created by parent PCI device
driver. Pass struct device of the parent to
snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use
correct dma_ops. Otherwise, it will use default dma_ops which
is nommu_dma_ops on x86_64 even when IOMMU is enabled and
set to non passthrough mode.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
- A further fix for syzcaller issues with USB-audio, addressing
NULL dereference that was introduced by the recent fix
- Avoid a long delay at boot with HD-audio when i915 module was
built but not installed, found on some Debian systems
- A fix of small race window at PCM draining
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Merge tag 'sound-5.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
- A further fix for syzcaller issues with USB-audio, addressing NULL
dereference that was introduced by the recent fix
- Avoid a long delay at boot with HD-audio when i915 module was built
but not installed, found on some Debian systems
- A fix of small race window at PCM draining
* tag 'sound-5.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check
ALSA: pcm: fix lost wakeup event scenarios in snd_pcm_drain
ALSA: hda: Fix 1-minute detection delay when i915 module is not available
88200 and 96000 sampling rate was not enabled on driver, so can't be played.
The error information:
max98373 3-0031：rate 96000 not supported
max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22
Signed-off-by: fengchunguo <chunguo.feng@amlogic.com>
Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The A64 audio codec uses the original I2S block but the SR and
WSS computation currently assigned is for the newer block.
Fixes: 619c15f7fa (ASoC: sun4i-i2s: Change SR and WSS computation)
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
syzbot found the following crash on:
general protection fault: 0000 [#1] SMP KASAN
RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75
Call Trace:
snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007
snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline]
snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280
usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576
usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361
really_probe+0x281/0x650 drivers/base/dd.c:548
....
It was introduced in commit 801ebf1043 for checking pipe and endpoint
types. It is fixed by adding a check of the ep pointer in question.
BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2
Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com>
Fixes: 801ebf1043 ("ALSA: usb-audio: Sanity checks for each pipe and EP types")
Cc: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Hillf Danton <hdanton@sina.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
lost wakeup can occur after enabling irq, therefore put task
into interruptible before enabling interrupts,
without this change, task can be put to sleep and snd_pcm_drain
will delay
Fixes: f2b3614cef ("ALSA: PCM - Don't check DMA time-out too shortly")
Signed-off-by: Yuki Tsunashima <ytsunashima@jp.adit-jv.com>
Signed-off-by: Suresh Udipi <sudipi@jp.adit-jv.com>
[ported from 4.9]
Signed-off-by: Adam Miartus <amiartus@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: f9b54e1961 ("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>
All relative small changes.
- A regression fix for PCM link code with CONFIG_REFCOUNT_FULL;
stumbled on a slight difference between atomic_t and refcount_t
- A couple of HD-audio stabilization patches addressing the too
slow PM resume seen on some Intel chips
- A series of ALSA compress-offload API fixes, including the
regression by the previous capture stream support
- Trivial LINE6 USB-audio driver fixes, a new Conexant HD-audio
chip coverage, and a fix in AC97 bus error path
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Merge tag 'sound-5.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All relatively small changes:
- a regression fix for PCM link code with CONFIG_REFCOUNT_FULL;
stumbled on a slight difference between atomic_t and refcount_t
- a couple of HD-audio stabilization patches addressing the too slow
PM resume seen on some Intel chips
- a series of ALSA compress-offload API fixes, including the
regression by the previous capture stream support
- trivial LINE6 USB-audio driver fixes, a new Conexant HD-audio chip
coverage, and a fix in AC97 bus error path"
* tag 'sound-5.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Add a conexant codec entry to let mute led work
ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips
ALSA: ac97: Fix double free of ac97_codec_device
ALSA: compress: Be more restrictive about when a drain is allowed
ALSA: compress: Don't allow paritial drain operations on capture streams
ALSA: compress: Prevent bypasses of set_params
ALSA: compress: Fix regression on compressed capture streams
ALSA: line6: Fix a typo
ALSA: pcm: Fix refcount_inc() on zero usage
ALSA: line6: Fix wrong altsetting for LINE6_PODHD500_1
ALSA: hda - Optimize resume for codecs without jack detection
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>
This reverts commit db51707b9c.
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>
Fix some typo to have the filaname given in a comment match the real name
of the file.
Some 'acpi' have erroneously been written 'apci'
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
When running McASP as master capture alone will not record any audio unless
a parallel playback stream is running. As soon as the playback stops the
captured data is going to be silent again.
In McASP master mode we need to set the PDIR for the clock pins and fix
the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above
AMUTE.
This went unnoticed as most of the boards uses McASP as slave and neither
of these issues are visible (audible) in those setups.
Fixes: ca3d943334 ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
It turned out that the recent Intel HD-audio controller chips show a
significant stall during the system PM resume intermittently. It
doesn't happen so often and usually it may read back successfully
after one or more seconds, but in some rare worst cases the driver
went into fallback mode.
After trial-and-error, we found out that the communication stall seems
covered by issuing the sync after each verb write, as already done for
AMD and other chipsets. So this patch enables the write-sync flag for
the recent Intel chips, Skylake and onward, as a workaround.
Also, since Broxton and co have the very same driver flags as Skylake,
refer to the Skylake driver flags instead of defining the same
contents again for simplification.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901
Reported-and-tested-by: Todd Brandt <todd.e.brandt@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
put_device will call ac97_codec_release to free
ac97_codec_device and other resources, so remove the kfree
and other redundant code.
Fixes: 74426fbff6 ("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>
Downgrade "nothing to do in IRQ thread" message from error to a debug
message in the IPC interrupt handler thread.
The spurious wake-up can happen if a HDA stream interrupt is
raised while the IPC interrupt thread is running. IPC functionality
is not impacted by this condition, so debug is a more appropriate
trace level.
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/20190722141402.7194-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
apq8016_sbc_parse_of() sets up multiple DAI links, depending on the
number of nodes in the device tree. However, at the moment
CPU and platform components are only allocated for the first link.
This causes an oops when more than one link is defined:
Internal error: Oops: 96000044 [#1] SMP
CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4
Call trace:
apq8016_sbc_platform_probe+0x1a8/0x3f0
platform_drv_probe+0x50/0xa0
...
Move the allocation inside the loop to ensure that each link is
properly initialized.
Fixes: 98b232ca9e ("ASoC: qcom: apq8016_sbc: use modern dai_link style")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
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: 4f2ab5e1d1 ("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>
Linus Torvalds