Kailang Yang [Fri, 26 Apr 2019 08:35:41 +0000 (16:35 +0800)]
ALSA: hda/realtek - EAPD turn on later
Let EAPD turn on after set pin output.
[ NOTE: This change is supposed to reduce the possible click noises at
(runtime) PM resume. The functionality should be same (i.e. the
verbs are executed correctly) no matter which order is, so this
should be safe to apply for all codecs -- tiwai ]
ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()
The error from snd_usb_mixer_apply_create_quirk() is ignored in the
current usb-audio driver code, which will continue the probing even
after the error. Let's take it more serious.
ALSA: ps3: Remove set but not used variables 'start_vaddr' and 'pcm_index'
Fixes gcc '-Wunused-but-set-variable' warnings:
sound/ppc/snd_ps3.c: In function 'snd_ps3_program_dma':
sound/ppc/snd_ps3.c:236:8: warning: variable 'start_vaddr' set but not used [-Wunused-but-set-variable]
sound/ppc/snd_ps3.c: In function 'snd_ps3_pcm_open':
sound/ppc/snd_ps3.c:529:6: warning: variable 'pcm_index' set but not used [-Wunused-but-set-variable]
ALSA: core: Don't refer to snd_cards array directly
The snd_cards[] array holds the card pointers that have been currently
registered, and it's exported for the external modules that may need
to refer a card object. But accessing to this array can be racy
against the driver probe or removal, as the card registration or free
may happen concurrently.
This patch gets rid of the direct access to snd_cards[] array and
provides a helper function to give the card object from the index
number with a refcount management. Then the caller can access to the
given card object safely, and releases it via snd_card_unref().
While we're at it, add a proper comment to snd_card_unref() and make
it an inlined function for type-safety, too.
ALSA: emu10k1: Drop superfluous id-uniquification behavior
The emu10k1 driver tries to create a unique id string by itself when
it's copied from the card list, but it's rather superfluous, as the
same thing will be done in ALSA core side at the card registration.
Let's drop the code. This allows us removing snd_cards export.
ALSA: seq: Correct unlock sequence at snd_seq_client_ioctl_unlock()
The doubly unlock sequence at snd_seq_client_ioctl_unlock() is tricky.
I took a direct unref call since I thought it would avoid
misunderstanding, but rather this seems more confusing. Let's use
snd_seq_client_unlock() consistently even if they look strange to be
called twice, and add more comments for avoiding reader's confusion.
Fixes: 6b580f523172 ("ALSA: seq: Protect racy pool manipulation from OSS sequencer") Reviewed-by: Kai Vehmanen <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
Roope Salmi [Sun, 14 Apr 2019 11:13:06 +0000 (14:13 +0300)]
ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo
The device reports Synch: Synchronous on the playback interface.
This causes regular audible napping on sample rates that are not multiples
of 1 kHz. Fix to Synch: Asynchronous.
Specifically observed on Focusrite Scarlett Solo 2nd generation. I assume
the first generation model has a different device ID. A first generation
Scarlett 2i2 I was able to test advertised Synch: Asynchronous by default.
For example, with a sample rate of 44100 Hz, a silent sample is played
every 40.96 seconds (likely 44.0 samples instead of 44.1 transmitted per
USB frame on average, 4096 being the size of some internal buffer).
There may be some other bug at play here since this doesn't happen
on other platforms. However, a feedback endpoint is listed and using it
fixes the issue. That is the only change in the quirk,
but I didn't find a way to declare only it.
Tested on two units and on two different computers.
ALSA: hda: Initialize ext-bus-specific fields in snd_hdac_bus_init(), too
Some fields in snd_hdac_bus are ext-bus specific, but they still
should be initialized in snd_hdac_bus_init() for consistency, at
least, for the ones that do need the explicit initialization like the
list head.
Also move the lock field to the more appropriate place and correct the
comment to reflect the recent change where it serves for both the
display power and the link management.
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 <[email protected]>
ALSA: seq: Protect racy pool manipulation from OSS sequencer
OSS sequencer emulation still allows to queue and issue the events
that manipulate the client pool concurrently in a racy way. This
patch serializes the access like the normal sequencer write / ioctl
via taking the client ioctl_mutex. Since the access to the sequencer
client is done indirectly via a client id number, a new helper to
take/release the mutex is introduced.
We have two helpers for queuing a sequencer event from the kernel
client, and both are used only from OSS sequencer layer without any
hop and atomic set. Let's simplify and unify two helpers into one.
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.
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.
Merge tag 'asoc-fix-v5.1-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.1
A few core fixes along with the driver specific ones, mainly fixing
small issues that only affect x86 platforms for various reasons (their
unusual machine enumeration mechanisms mainly, plus a fix for error
handling in topology).
There's some of the driver fixes that look larger than they are, like
the hdmi-codec changes which resulted in an indentation change, and most
of the other large changes are for new drivers like the STM32 changes.
snd_hdac_display_power() doesn't handle the concurrent calls carefully
enough, and it may lead to the doubly get_power or put_power calls,
when a runtime PM and an async work get called in racy way.
This patch addresses it by reusing the bus->lock mutex that has been
used for protecting the link state change in ext bus code, so that it
can protect against racy display state changes. The initialization of
bus->lock was moved from snd_hdac_ext_bus_init() to
snd_hdac_bus_init() as well accordingly.
When master clock is used, master clock rate is set exclusively.
Parent clocks of master clock cannot be changed after a call to
clk_set_rate_exclusive(). So the parent clock of SAI kernel clock
must be set before.
Ensure also that exclusive rate operations are balanced
in STM32 SAI driver.
Takashi Iwai [Thu, 28 Mar 2019 16:11:10 +0000 (17:11 +0100)]
ALSA: timer: Simplify error path in snd_timer_open()
Just a minor refactoring to use the standard goto for error paths in
snd_timer_open() instead of open code. The first mutex_lock() is
moved to the beginning of the function to make the code clearer.
ALSA: seq: Fix race of get-subscription call vs port-delete ioctls
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.
ALSA: seq: Protect in-kernel ioctl calls with mutex
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.
Takashi Iwai [Thu, 28 Mar 2019 15:21:01 +0000 (16:21 +0100)]
ALSA: seq: Remove superfluous irqsave flags
spin_lock_irqsave() is used unnecessarily in various places in
sequencer core code although it's pretty obvious that the context is
sleepable. Remove irqsave and use the plain spin_lock_irq() in such
places for simplicity.
Takashi Iwai [Thu, 28 Mar 2019 14:55:08 +0000 (15:55 +0100)]
ALSA: seq: Align temporary re-locking with irqsave version
In a few places in sequencer core, we temporarily unlock / re-lock the
pool spin lock while waiting for the allocation in the blocking mode.
There spin_unlock_irq() / spin_lock_irq() pairs are called while
initially spin_lock_irqsave() is used (and spin_lock_irqrestore() at
the end of the function again). This is likely OK for now, but it's a
bit confusing and error-prone.
This patch replaces these temporary relocking lines with the irqsave
variant to make the lock/unlock sequence more consistently.
ALSA: timer: Revert active callback sync check at close
This is essentially a revert of the commit a7588c896b05 ("ALSA: timer:
Check ack_list emptiness instead of bit flag"). The intended change
by the commit turns out to be insufficient, as snd_timer_close*()
always calls snd_timer_stop() that deletes the ack_list beforehand.
In theory, we can change the behavior of snd_timer_stop() to sync the
pending ack_list, but this will become a deadlock for the callback
like sequencer that calls again snd_timer_stop() from itself. So,
reverting the change is a more straightforward solution.
Fixes: a7588c896b05 ("ALSA: timer: Check ack_list emptiness instead of bit flag") Reported-by: [email protected] Signed-off-by: Takashi Iwai <[email protected]>
Hui Wang [Mon, 8 Apr 2019 07:58:11 +0000 (15:58 +0800)]
ALSA: hda - Add two more machines to the power_save_blacklist
Recently we set CONFIG_SND_HDA_POWER_SAVE_DEFAULT to 1 when
configuring the kernel, then two machines were reported to have noise
after installing the new kernel. Put them in the blacklist, the
noise disappears.
ASoC: pcm: update module refcount if module_get_upon_open is set
Setting the module_get_upon_open field for component driver
prevents the module refcount from being incremented during
component probe(). This could lead to the module being
allowed to be unloaded when a pcm stream is open. So,
if this field is set, the module's refcount should be
incremented during pcm open to prevent module removal
when the component is in use. And, the refcount should
be decremented upon pcm close.
ASoC: core: conditionally increase module refcount on component open
Recently, for Intel platforms the "ignore_module_refcount" field
was introduced for the component driver. In order to avoid a
deadlock preventing the PCI modules from being removed
even when the card was idle, the refcounts were not incremented
for the device driver module during component probe.
However, this change introduced a nasty side effect:
the device driver module can be unloaded while a pcm stream is open.
This patch proposes to change the field to be renamed as
"module_get_upon_open". When this field is set, the module
refcount should be incremented on pcm open amd decremented
upon pcm close. This will enable modules to be removed
when no PCM playback/capture happens and prevent removal
when the component is actually in use.
Also, align with the skylake component driver with the new name.
ASoC: topology: Use the correct dobj to free enum control values and texts
The control values and texts of the enum kcontrol associated
with a widget need to be freed when the widget is removed.
However, both struct snd_soc_dapm_widget and struct soc_enum
contain a dobj member, which resulted in a confusion.
The existing code generates a null pointer dereference by
attempting to free the values and texts from the dobj which
belongs to the widget instead of the dobj belonging to the
enum kcontrol.
The suggested fix is to use the correct dobj member (se->dobj)
of the enum kcontrol.
When ioctl calls are made with non-null-terminated userspace strings,
strlcpy causes an OOB-read from within strlen. Fix by changing to use
strscpy instead.
Charles Keepax [Thu, 4 Apr 2019 16:27:20 +0000 (17:27 +0100)]
ASoC: cs35l35: Disable regulators on driver removal
The chips main power supplies VA and VP are enabled during probe but
then never disabled, this will cause warnings from the regulator
framework on driver removal. Fix this by adding a remove callback and
disabling the supplies, whilst doing so follow best practice and put the
chip back into reset as well.
ASoC: simple-card: don't select DPCM via simple-audio-card
commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card")
merged simple-scu-audio-card which can handle DPCM into
simple-audio-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its CPU/Codec DAI count.
But, because of it, existing "simple-audio-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "simple-audio-card" user
can select "normal sound card", and "simple-scu-audio-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Fixes: da215354eb55c ("ASoC: simple-card: merge simple-scu-card") Signed-off-by: Kuninori Morimoto <[email protected]> Signed-off-by: Mark Brown <[email protected]>
ASoC: audio-graph-card: don't select DPCM via audio-graph-card
commit ae3cb5790906b ("ASoC: audio-graph-card: merge
audio-graph-scu-card") merged audio-graph-scu-card which can
handle DPCM into audio-graph-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its OF-graph endpoint connection.
But, because of it, existing "audio-graph-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "audio-graph-card" user
can select "normal sound card", and "audio-graph-scu-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Richard Sailer [Tue, 2 Apr 2019 13:52:04 +0000 (15:52 +0200)]
ALSA: hda/realtek - Add quirk for Tuxedo XC 1509
This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509.
The Tuxedo XC 1509 and the System76 oryp5 are the same barebone
notebooks manufactured by Clevo. To name the fixups both use after the
actual underlying hardware, this patch also changes System76_orpy5
to clevo_pb51ed in 2 enum symbols and one function name,
matching the other pci_quirk entries which are also named after the
device ODM.
Fixes: 7f665b1c3283 ("ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5") Signed-off-by: Richard Sailer <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
Kailang Yang [Wed, 3 Apr 2019 07:31:49 +0000 (15:31 +0800)]
ALSA: hda/realtek - Move to ACT_INIT state
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Just Implement of reset combo jack JD verb for ACT_PRE_PROBE state.
Intel test result was also failed.
It test passed until changed the initial state to ACT_INIT.
Mic JD will show every time.
This patch also changed the model name as 'alc-chrome-book' for
application of Chrome OS.
Fixes: 10f5b1b85ed1 ("ALSA: hda/realtek - Fixed Headset Mic JD not stable") Signed-off-by: Kailang Yang <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
Hans de Goede [Tue, 2 Apr 2019 10:20:49 +0000 (12:20 +0200)]
ASoC: Intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled
Users have been seeing sound stability issues with max98090 codecs since:
commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
At first that commit broke sound for Chromebook Swanky and Clapper models,
the problem was that the machine-driver has been controlling the wrong
clock on those models since support for them was added. This was hidden by
clk-pmc-atom.c keeping the actual clk on unconditionally.
With the machine-driver controlling the proper clock, sound works again
but we are seeing bug reports describing it as: low volume,
"sounds like played at 10x speed" and instable.
When these issues are hit the following message is seen in dmesg:
"max98090 i2c-193C9890:00: PLL unlocked".
Attempts have been made to fix this by inserting a delay between enabling
the clk and enabling and checking the pll, but this has not helped.
It seems that at least on boards which use pmc_plt_clk_0 as clock,
if we ever disable the clk, the pll looses its lock and after that we get
various issues.
This commit fixes this by enabling the clock once at probe time on
these boards. In essence this restores the old behavior of clk-pmc-atom.c
always keeping the clk on on these boards.
Charles Keepax [Tue, 2 Apr 2019 12:49:14 +0000 (13:49 +0100)]
ASoC: wm_adsp: Check for buffer in trigger stop
Trigger stop can be called in situations where trigger start failed
and as such it can't be assumed the buffer is already attached to
the compressed stream or a NULL pointer may be dereferenced.
Fixes: 639e5eb3c7d6 ("ASoC: wm_adsp: Correct handling of compressed streams that restart") Signed-off-by: Charles Keepax <[email protected]> Signed-off-by: Mark Brown <[email protected]>
Jian-Hong Pan [Mon, 1 Apr 2019 03:25:05 +0000 (11:25 +0800)]
ALSA: hda/realtek: Enable headset MIC of Acer TravelMate B114-21 with ALC233
The Acer TravelMate B114-21 laptop cannot detect and record sound from
headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb
quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix
this issue.
[ fixed the missing brace and reordered the entry -- tiwai ]
ASoC: dapm: set power_check callback for widgets that shouldnt be always on
Currently, buffers, schedulers, src's, encoders, decoders
and effect type dapm widgets remain always on as their
power_check method is not set. Setting this callback allows these
widgets in the audio path to be powered managed properly.
ASoC: dpcm: skip missing substream while applying symmetry
If for any reason, the backend does not have the requested substream
(like capture on a playback only backend), the BE will be skipped in
dpcm_be_dai_startup().
However, dpcm_apply_symmetry() does not skip those BE and will
dereference the be_substream (NULL) pointer anyway.
Like in dpcm_be_dai_startup(), just skip those BE.
Fixes: 906c7d690c3b ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: Jerome Brunet <[email protected]> Signed-off-by: Mark Brown <[email protected]>
Takashi Iwai [Sat, 16 Mar 2019 07:48:52 +0000 (08:48 +0100)]
ALSA: us122l: Use alloc_pages_exact()
alloc_pages_exact() is more suitable choice for allocating the sound
buffers, as it doesn't need to align with power-of-two. Along with
the conversion, we can drop __GFP_COMP as well.
The patch also replace the error messages to be more explicit.
Takashi Iwai [Fri, 23 Nov 2018 18:38:13 +0000 (19:38 +0100)]
ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers, take#2
snd_malloc_pages() and snd_free_pages() are merely thin wrappers of
the standard page allocator / free functions. Even the arguments are
compatible with some standard helpers, so there is little merit of
keeping these wrappers.
This patch replaces the all existing callers of snd_malloc_pages() and
snd_free_pages() with the direct calls of the standard helper
functions. In this version, we use a recently introduced one,
alloc_pages_exact(), which suits better than the old
snd_malloc_pages() implementation for our purposes. Then we can avoid
the waste of pages by alignment to power-of-two.
Since alloc_pages_exact() does split pages, we need no longer
__GFP_COMP flag; or better to say, we must not pass __GFP_COMP to
alloc_pages_exact(). So the former unconditional addition of
__GFP_COMP flag in snd_malloc_pages() is dropped, as well as in most
other places.
Takashi Iwai [Wed, 27 Mar 2019 16:02:40 +0000 (17:02 +0100)]
ALSA: timer: Make snd_timer_close() really kill pending actions
snd_timer_close() is supposed to close the timer instance and sync
with the deactivation of pending actions. However, there are still
some overlooked cases:
- It calls snd_timer_stop() at the beginning, but some other might
re-trigger the timer right after that.
- snd_timer_stop() calls del_timer_sync() only when all belonging
instances are closed. If multiple instances were assigned to a
timer object and one is closed, the timer is still running. Then
the pending action assigned to this timer might be left.
Actually either of the above is the likely cause of the reported
syzkaller UAF.
This patch plug these holes by introducing SNDRV_TIMER_IFLG_DEAD
flag. This is set at the beginning of snd_timer_close(), and the flag
is checked at snd_timer_start*() and else, so that no longer new
action is left after snd_timer_close().
Takashi Iwai [Wed, 27 Mar 2019 15:56:08 +0000 (16:56 +0100)]
ALSA: timer: Check ack_list emptiness instead of bit flag
For checking the pending timer instance that is still left on the
timer object that is being closed, we set/clear a bit flag
SNDRV_TIMER_IFLG_CALLBACK around the call of callbacks. This can be
simplified by replace with the list_empty() call for ti->ack_list.
This covers the existence more comprehensively and safely.
A gratis bonus is that we can get rid of SNDRV_TIMER_IFLG_CALLBACK bit
flag definition as well.
Takashi Iwai [Wed, 27 Mar 2019 15:51:58 +0000 (16:51 +0100)]
ALSA: timer: Make sure to clear pending ack list
When a card is under disconnection, we bail out immediately at each
timer interrupt or tasklet. This might leave some items left in ack
list. For a better integration of the upcoming change to check
ack_list emptiness, clear out the whole list upon the emergency exit
route.
Takashi Iwai [Wed, 27 Mar 2019 15:42:51 +0000 (16:42 +0100)]
ALSA: timer: Unify timer callback process code
The timer core has two almost identical code for processing callbacks:
once in snd_timer_interrupt() for fast callbacks and another in
snd_timer_tasklet() for delayed callbacks. Let's unify them.
In the new version, the resolution is read from ti->resolution at each
call, and this must be fine; ti->resolution is set in the preparation
step in snd_timer_interrupt().
Takashi Iwai [Tue, 26 Mar 2019 16:32:02 +0000 (17:32 +0100)]
ALSA: emux: Add support of loading GUS-patch
It's a feature request for the ancient sutff, but it's still valid;
the loading of a GUS-patch isn't available via hwdep device although
it's supported over OSS sequencer. The only missing piece is the call
of snd_soundfont_load_guspatch() in synth emux hwdep code.
Jenny TC [Sat, 23 Mar 2019 13:10:10 +0000 (18:40 +0530)]
ASoC: Intel: Skylake: enable S24_LE format support
To enable S24_LE format, sample_type in topology fw has to be set to 1.
But sample_type defined in topology firmware configuration is not
getting reflected in the dsp param. This patch sets sample_type in base
config so that the sample type defined in the topology firmware is reflected
in the dsp params. This issues was uncovered while debugging the S24_LE format
which require the MSB byte in 32 bit word to be skipped. Setting sample_type
in topology firmware to 1 helps to skip MSB byte word.
Takashi Iwai [Mon, 25 Mar 2019 09:38:58 +0000 (10:38 +0100)]
ALSA: pcm: Don't suspend stream in unrecoverable PCM state
Currently PCM core sets each opened stream forcibly to SUSPENDED state
via snd_pcm_suspend_all() call, and the user-space is responsible for
re-triggering the resume manually either via snd_pcm_resume() or
prepare call. The scheme works fine usually, but there are corner
cases where the stream can't be resumed by that call: the streams
still in OPEN state before finishing hw_params. When they are
suspended, user-space cannot perform resume or prepare because they
haven't been set up yet. The only possible recovery is to re-open the
device, which isn't nice at all. Similarly, when a stream is in
DISCONNECTED state, it makes no sense to change it to SUSPENDED
state. Ditto for in SETUP state; which you can re-prepare directly.
So, this patch addresses these issues by filtering the PCM streams to
be suspended by checking the PCM state. When a stream is in either
OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend
action is skipped.
To be noted, this problem was originally reported for the PCM runtime
PM on HD-audio. And, the runtime PM problem itself was already
addressed (although not intended) by the code refactoring commits 3d21ef0b49f8 ("ALSA: pcm: Suspend streams globally via device type PM
ops") and 17bc4815de58 ("ALSA: pci: Remove superfluous
snd_pcm_suspend*() calls"). These commits eliminated the
snd_pcm_suspend*() calls from the runtime PM suspend callback code
path, hence the racy OPEN state won't appear while runtime PM.
(FWIW, the race window is between snd_pcm_open_substream() and the
first power up in azx_pcm_open().)
Although the runtime PM issue was already "fixed", the same problem is
still present for the system PM, hence this patch is still needed.
And for stable trees, this patch alone should suffice for fixing the
runtime PM problem, too.
In the observed situation, the problem is seen because the codec driver
failed to probe due to a hardware problem.
max98090 i2c-193C9890:00: Failed to read device revision: -1
max98090 i2c-193C9890:00: ASoC: failed to probe component -1
cht-bsw-max98090 cht-bsw-max98090: ASoC: failed to instantiate card -1
cht-bsw-max98090 cht-bsw-max98090: snd_soc_register_card failed -1
cht-bsw-max98090: probe of cht-bsw-max98090 failed with error -1
The problem is similar to the problem solved with commit 2fc995a87f2e
("ASoC: intel: Fix crash at suspend/resume without card registration"),
but codec registration fails at a later point. At that time, the pointer
checked with the above mentioned commit is already set, but it is not
cleared if the device is subsequently removed. Adding a remove function
to clear the pointer fixes the problem.
Takashi Iwai [Fri, 22 Mar 2019 14:51:36 +0000 (15:51 +0100)]
ALSA: hda/ca0132 - Simplify alt firmware loading code
ca0132 codec driver loads the firmware selectively depending on the
model in addition to the fallback of the default firmware. The code
works good, but a minor problem is that the current code seems
confusing for Clang where it spews a warning about uninitialized
variable.
This patch simplifies the code flow for such a false-positive
warning. After this refactoring, the ca0132_spec.alt_firmware_present
field is no longer used, hence it's eliminated as well.
Takashi Iwai [Fri, 22 Mar 2019 15:00:54 +0000 (16:00 +0100)]
ALSA: pcm: Fix possible OOB access in PCM oss plugins
The PCM OSS emulation converts and transfers the data on the fly via
"plugins". The data is converted over the dynamically allocated
buffer for each plugin, and recently syzkaller caught OOB in this
flow.
Although the bisection by syzbot pointed out to the commit 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer
allocations"), this is merely a commit to replace vmalloc() with
kvmalloc(), hence it can't be the cause. The further debug action
revealed that this happens in the case where a slave PCM doesn't
support only the stereo channels while the OSS stream is set up for a
mono channel. Below is a brief explanation:
At each OSS parameter change, the driver sets up the PCM hw_params
again in snd_pcm_oss_change_params_lock(). This is also the place
where plugins are created and local buffers are allocated. The
problem is that the plugins are created before the final hw_params is
determined. Namely, two snd_pcm_hw_param_near() calls for setting the
period size and periods may influence on the final result of channels,
rates, etc, too, while the current code has already created plugins
beforehand with the premature values. So, the plugin believes that
channels=1, while the actual I/O is with channels=2, which makes the
driver reading/writing over the allocated buffer size.
The fix is simply to move the plugin allocation code after the final
hw_params call.
S.j. Wang [Wed, 27 Feb 2019 06:31:12 +0000 (06:31 +0000)]
ASoC: fsl_esai: fix channel swap issue when stream starts
There is very low possibility ( < 0.1% ) that channel swap happened
in beginning when multi output/input pin is enabled. The issue is
that hardware can't send data to correct pin in the beginning with
the normal enable flow.
This is hardware issue, but there is no errata, the workaround flow
is that: Each time playback/recording, firstly clear the xSMA/xSMB,
then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled
before xSMA). Which is to use the xSMA as the trigger start register,
previously the xCR_TE or xCR_RE is the bit for starting.
S.j. Wang [Sat, 2 Mar 2019 05:52:19 +0000 (05:52 +0000)]
ASoC: fsl_asrc: add constraint for the asrc of older version
There is a constraint for the channel number setting on the
asrc of older version (e.g. imx35), the channel number should
be even, odd number isn't valid.
So add this constraint when the asrc of older version is used.
Daniel Mack [Wed, 20 Mar 2019 21:41:56 +0000 (22:41 +0100)]
ASoC: cs4270: Set auto-increment bit for register writes
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
Fix this by sanitizing dev before using it to index dp->synths.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
Fix this by sanitizing info->stream before using it to index
rmidi->streams.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
Jian-Hong Pan [Fri, 15 Mar 2019 09:51:09 +0000 (17:51 +0800)]
ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed9623 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G") Fixes: b72f936f6b32 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G") Signed-off-by: Jian-Hong Pan <[email protected]> Reviewed-by: Kailang Yang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
Charles Keepax [Tue, 19 Mar 2019 11:52:07 +0000 (11:52 +0000)]
ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout
If a watchdog timeout is received from the DSP it is safe to assume the
DSP is not functioning anymore and as such any active compressed streams
should be put into an error state.
Charles Keepax [Tue, 19 Mar 2019 11:52:04 +0000 (11:52 +0000)]
ASoC: wm_adsp: Correct handling of compressed streams that restart
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40e6 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active") Signed-off-by: Charles Keepax <[email protected]> Signed-off-by: Mark Brown <[email protected]>
Hui Wang [Tue, 19 Mar 2019 01:28:44 +0000 (09:28 +0800)]
ALSA: hda - Enforces runtime_resume after S3 and S4 for each codec
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control") Signed-off-by: Hui Wang <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
Hui Wang [Tue, 19 Mar 2019 01:28:43 +0000 (09:28 +0800)]
ALSA: hda - Don't trigger jackpoll_work in azx_resume
The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Shuming Fan [Mon, 18 Mar 2019 07:17:42 +0000 (15:17 +0800)]
ASoC: rt5682: recording has no sound after booting
If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Shuming Fan [Fri, 8 Mar 2019 03:36:08 +0000 (11:36 +0800)]
ASoC: rt5682: Check JD status when system resume
The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Takashi Sakamoto [Sun, 17 Mar 2019 11:25:06 +0000 (20:25 +0900)]
ALSA: firewire-lib: use 8 byte header for IR context to get isochronous cycle
In kernel API of Linux FireWire subsystem, handlers of isochronous
receive (IR) context can get context headers as an argument of
callback. When 4 byte header is used, the context header includes
isochronous packet header for each packet. When 8 byte header is
used, it includes isochronous cycle as well.
ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous
cycle from the cycle of interrupt. The usage of 8 byte header can
obsolete the computation.
Furthermore, this change works well for a case that a series of
packet in one interrupt includes skipped isochronous cycle,
This commit uses 8 byte header to handle isochronous cycle.
Takashi Sakamoto [Sun, 17 Mar 2019 07:50:24 +0000 (16:50 +0900)]
ALSA: firewire-motu: add support MOTU 8pre FireWire
This commit adds support for MOTU 8pre FireWire, which was shipped 2007
and nowadays already discontinued. Userspace applications can transmit
and receive PCM frames and MIDI messages for this model via ALSA PCM
interface and RawMidi/Sequencer interfaces.
Like the other models of MOTU FireWire series, this model has many
quirks in its CIP.
At first, data channels for two pairs of optical interfaces. At lower
sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is
available for ADAT data, thus 8 data chunks are transferred by CIP.
At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz,
two pairs are available to keep 8 chunks for ADAT data, thus CIP
still includes 8 data chunks.
Apart from data chunks for optical interface, CIP includes fixed number
of data chunks. In tx stream, two chunks for status message, eight
chunks for samples from analog 1-8 input, two chunks for mix-return.
In rx stream, two chunks for control message, two chunks for main 1-2
output, two chunks for phone 1-2 output, two chunks for dummy 1-2.
CIP header in tx stream includes quirks for its dbs and dbc fields.
The value of dbs field is fixed to 0x13, against its actual size.
The value of dbc field is firstly updated to 0x07 from zero, then
it's incremented continuously according to actual number of data h
blocks.
Finally, the model has own bits to disable frame fetch.
This commit uses several options to absorb the above quirks.
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 0003991c directory_length 3, crc 39196
42c 120001f2 specifier id
430 1300000f version
434 17103800 model
Colin Ian King [Sun, 17 Mar 2019 23:21:24 +0000 (23:21 +0000)]
ALSA: opl3: fix mismatch between snd_opl3_drum_switch definition and declaration
The function snd_opl3_drum_switch declaration in the header file
has the order of the two arguments on_off and vel swapped when
compared to the definition arguments of vel and on_off. Fix this
by swapping them around to match the definition.
This error predates the git history, so no idea when this error
was introduced.
projects / linux.git / log