Christian A. Ehrhardt [Fri, 31 Dec 2021 13:44:32 +0000 (14:44 +0100)]
ALSA: hda/cs8409: Fix Jack detection after resume
The suspend code unconditionally sets ->hp_jack_in and ->mic_jack_in
to zero but without reporting this status change to the HDA core.
To compensate for this, always assume a status change on the
first unsol event after boot or resume.
Christian A. Ehrhardt [Fri, 31 Dec 2021 13:12:21 +0000 (14:12 +0100)]
ALSA: hda/cs8409: Increase delay during jack detection
Commit c8b4f0865e82 reduced delays related to cs42l42 jack
detection. However, the change was too aggressive. As a result
internal speakers on DELL Inspirion 3501 are not detected.
Increase the delay in cs42l42_run_jack_detect() a bit.
Christian Lachner [Mon, 3 Jan 2022 14:05:17 +0000 (15:05 +0100)]
ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master after reboot from Windows
This patch addresses an issue where after rebooting from Windows into Linux
there would be no audio output.
It turns out that the Realtek Audio driver on Windows changes some coeffs
which are not being reset/reinitialized when rebooting the machine. As a
result, there is no audio output until these coeffs are being reset to
their initial state. This patch takes care of that by setting known-good
(initial) values to the coeffs.
We initially relied upon alc1220_fixup_clevo_p950() to fix some pins in the
connection list. However, it also sets coef 0x7 which does not need to be
touched. Furthermore, to prevent mixing device-specific quirks I introduced
a new alc1220_fixup_gb_x570() which is heavily based on
alc1220_fixup_clevo_p950() but does not set coeff 0x7 and fixes the coeffs
that are actually needed instead.
This new alc1220_fixup_gb_x570() is believed to also work for other boards,
like the Gigabyte X570 Aorus Extreme and the newer Gigabyte Aorus X570S
Master. However, as there is no way for me to test these I initially only
enable this new behaviour for the mainboard I have which is the Gigabyte
X570(non-S) Aorus Master.
I tested this patch on the 5.15 branch as well as on master and it is
working well for me.
Sameer Pujar [Thu, 23 Dec 2021 11:53:49 +0000 (17:23 +0530)]
ALSA: hda/tegra: Fix Tegra194 HDA reset failure
HDA regression is recently reported on Tegra194 based platforms.
This happens because "hda2codec_2x" reset does not really exist
in Tegra194 and it causes probe failure. All the HDA based audio
tests fail at the moment. This underlying issue is exposed by
commit c045ceb5a145 ("reset: tegra-bpmp: Handle errors in BPMP
response") which now checks return code of BPMP command response.
Fix this issue by skipping unavailable reset on Tegra194.
Arie Geiger [Thu, 23 Dec 2021 23:28:57 +0000 (15:28 -0800)]
ALSA: hda/realtek: Add speaker fixup for some Yoga 15ITL5 devices
This patch adds another possible subsystem ID for the ALC287 used by
the Lenovo Yoga 15ITL5.
It uses the same initalization as the others.
This patch has been tested and works for my device.
Kai Vehmanen [Thu, 23 Dec 2021 07:34:23 +0000 (09:34 +0200)]
ALSA: hda: Add AlderLake-N PCI ID
Add HD Audio PCI ID for Intel AlderLake-N. Add rules to
snd_intel_dsp_find_config() to choose DSP-based SOF driver for ADL-N
systems with PCH-DMIC or Soundwire codecs, and plain HDA driver for the
rest (DSP not used).
Ville Syrjälä [Wed, 22 Dec 2021 14:53:50 +0000 (16:53 +0200)]
ALSA: hda/hdmi: Disable silent stream on GLK
The silent stream stuff recurses back into i915 audio
component .get_power() from the .pin_eld_notify() hook.
On GLK this will deadlock as i915 may already be holding
the relevant modeset locks during .pin_eld_notify() and
the GLK audio vs. CDCLK workaround will try to grab the
same locks from .get_power().
Until someone comes up with a better fix just disable the
silent stream support on GLK.
Mark Brown [Fri, 17 Dec 2021 13:02:12 +0000 (13:02 +0000)]
kselftest: alsa: Factor out check that values meet constraints
To simplify the code a bit and allow future reuse factor the checks that
values we read are valid out of test_ctl_get_value() into a separate
function which can be reused later. As part of this extend the test to
check all the values for the control, not just the first one.
Werner Sembach [Wed, 15 Dec 2021 19:16:46 +0000 (20:16 +0100)]
ALSA: hda/realtek: Fix quirk for Clevo NJ51CU
The Clevo NJ51CU comes either with the ALC293 or the ALC256 codec, but uses
the 0x8686 subproduct id in both cases. The ALC256 codec needs a different
quirk for the headset microphone working and and edditional quirk for sound
working after suspend and resume.
When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of
0x0020 on the ALC256 codec. Setting the value manually makes the sound
work again. This patch does this automatically.
[ minor coding style fix by tiwai ]
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Fixes: b5acfe152abaa ("ALSA: hda/realtek: Add some Clove SSID in the ALC293(ALC1220)") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211215191646.844644-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jaroslav Kysela [Sat, 18 Dec 2021 12:39:25 +0000 (13:39 +0100)]
ALSA: rawmidi - fix the uninitalized user_pversion
The user_pversion was uninitialized for the user space file structure
in the open function, because the file private structure use
kmalloc for the allocation.
The kernel ALSA sequencer code clears the file structure, so no additional
fixes are required.
Libin Yang [Tue, 21 Dec 2021 01:08:17 +0000 (09:08 +0800)]
ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2
In the HDAS ACPI scope, the SoundWire may not be the direct child of HDAS.
It needs to go through the ACPI table at max depth of 2 to find the
SoundWire device from HDAS.
Libin Yang [Tue, 21 Dec 2021 01:08:16 +0000 (09:08 +0800)]
ALSA: hda: intel-sdw-acpi: harden detection of controller
The existing code currently sets a pointer to an ACPI handle before
checking that it's actually a SoundWire controller. This can lead to
issues where the graph walk continues and eventually fails, but the
pointer was set already.
This patch changes the logic so that the information provided to
the caller is set when a controller is found.
Ville Syrjälä [Wed, 22 Dec 2021 14:53:50 +0000 (16:53 +0200)]
ALSA: hda/hdmi: Disable silent stream on GLK
The silent stream stuff recurses back into i915 audio
component .get_power() from the .pin_eld_notify() hook.
On GLK this will deadlock as i915 may already be holding
the relevant modeset locks during .pin_eld_notify() and
the GLK audio vs. CDCLK workaround will try to grab the
same locks from .get_power().
Until someone comes up with a better fix just disable the
silent stream support on GLK.
Takashi Iwai [Wed, 22 Dec 2021 17:07:27 +0000 (18:07 +0100)]
Merge tag 'asoc-fix-v5.16-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.16
This is a relatively large set of driver specific changes so it may make
sense to hold off to v5.17, though picking some over might be good.
It's a combination of new device IDs and fixes for various driver
specific things which are all small and of the usual "really bad if
you're running into them" level, especially the Tegra ones.
Martin Blumenstingl [Mon, 6 Dec 2021 21:08:04 +0000 (22:08 +0100)]
ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s
The out-of-tree vendor driver uses the following approach to set the
AIU_I2S_MISC register:
1) write AIU_MEM_I2S_START_PTR and AIU_MEM_I2S_RD_PTR
2) configure AIU_I2S_MUTE_SWAP[15:0]
3) write AIU_MEM_I2S_END_PTR
4) set AIU_I2S_MISC[2] to 1 (documented as: "put I2S interface in hold
mode")
5) set AIU_I2S_MISC[4] to 1 (depending on the driver revision it always
stays at 1 while for older drivers this bit is unset in step 4)
6) set AIU_I2S_MISC[2] to 0
7) write AIU_MEM_I2S_MASKS
8) toggle AIU_MEM_I2S_CONTROL[0]
9) toggle AIU_MEM_I2S_BUF_CNTL[0]
Move setting the AIU_I2S_MISC[2] bit to aiu_fifo_i2s_hw_params() so it
resembles the flow in the vendor kernel more closely. While here also
configure AIU_I2S_MISC[4] (documented as: "force each audio data to
left or right according to the bit attached with the audio data")
similar to how the vendor driver does this. This fixes the infamous and
long-standing "machine gun noise" issue (a buffer underrun issue).
Fixes: 6ae9ca9ce986bf ("ASoC: meson: aiu: add i2s and spdif support") Reported-by: Christian Hewitt <christianshewitt@gmail.com> Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com> Tested-by: Christian Hewitt <christianshewitt@gmail.com> Tested-by: Geraldo Nascimento <geraldogabriel@gmail.com> Acked-by: Jerome Brunet <jbrunet@baylibre.com> Cc: stable@vger.kernel.org Signed-off-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com> Link: https://lore.kernel.org/r/20211206210804.2512999-3-martin.blumenstingl@googlemail.com Signed-off-by: Mark Brown <broonie@kernel.org>
The FIFO registers which take an DMA-able address are only 32-bit wide
on AIU. Add dma_coerce_mask_and_coherent() to make the DMA core aware of
this limitation.
Derek Fang [Tue, 14 Dec 2021 10:50:33 +0000 (18:50 +0800)]
ASoC: rt5682: fix the wrong jack type detected
Some powers were changed during the jack insert detection
and clk's enable/disable in CCF.
If in parallel, the influence has a chance to detect
the wrong jack type, so add a lock.
Bradley Scott [Mon, 13 Dec 2021 16:22:47 +0000 (11:22 -0500)]
ALSA: hda/realtek: Add new alc285-hp-amp-init model
Adds a new "alc285-hp-amp-init" model that can be used to apply the ALC285
HP speaker amplifier initialization fixup to devices that are not already
known by passing "hda_model=alc285-hp-amp-init" to the
snd-sof-intel-hda-common module or "model=alc285-hp-amp-init" to the
snd-hda-intel module, depending on which is being used.
Dmitry Osipenko [Sat, 11 Dec 2021 23:11:46 +0000 (02:11 +0300)]
ASoC: tegra: Restore headphones jack name on Nyan Big
UCM of Acer Chromebook (Nyan) uses a different name for the headphones
jack. The name was changed during unification of the machine drivers and
UCM fails now to load because of that. Restore the old jack name.
Cc: <stable@vger.kernel.org> Fixes: cc8f70f ("ASoC: tegra: Unify ASoC machine drivers") Reported-by: Thomas Graichen <thomas.graichen@gmail.com> # T124 Nyan Big Tested-by: Thomas Graichen <thomas.graichen@gmail.com> # T124 Nyan Big Signed-off-by: Dmitry Osipenko <digetx@gmail.com> Link: https://lore.kernel.org/r/20211211231146.6137-2-digetx@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dmitry Osipenko [Sat, 11 Dec 2021 23:11:45 +0000 (02:11 +0300)]
ASoC: tegra: Add DAPM switches for headphones and mic jack
UCM of Acer Chromebook (Nyan) uses DAPM switches of headphones and mic
jack. These switches were lost by accident during unification of the
machine drivers, restore them.
Cc: <stable@vger.kernel.org> Fixes: cc8f70f ("ASoC: tegra: Unify ASoC machine drivers") Reported-by: Thomas Graichen <thomas.graichen@gmail.com> # T124 Nyan Big Tested-by: Thomas Graichen <thomas.graichen@gmail.com> # T124 Nyan Big Signed-off-by: Dmitry Osipenko <digetx@gmail.com> Link: https://lore.kernel.org/r/20211211231146.6137-1-digetx@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Mon, 13 Dec 2021 14:15:12 +0000 (15:15 +0100)]
ALSA: gus: Fix memory leaks at memory allocator error paths
When snd_gf1_mem_xalloc() returns NULL, the current code still leaves
the formerly allocated block.name string but returns an error
immediately. This patch does code-refactoring to move the kstrdup()
call itself into snd_gf1_mem_xalloc() and deals with the resource free
in the helper code by itself for fixing those memory leaks.
Takashi Iwai [Mon, 13 Dec 2021 13:24:43 +0000 (14:24 +0100)]
ALSA: gus: Fix erroneous memory allocation
snd_gf1_mem_xalloc() returns NULL incorrectly when the memory chunk is
allocated in the middle of the chain. This patch corrects the return
value to treat it properly.
Xiaoke Wang [Mon, 13 Dec 2021 10:52:32 +0000 (18:52 +0800)]
ALSA: sound/isa/gus: check the return value of kstrdup()
kstrdup() returns NULL when some internal memory errors happen, it is
better to check the return value of it. Otherwise, we may not to be able
to catch some memory errors in time.
Colin Ian King [Sun, 12 Dec 2021 17:20:25 +0000 (17:20 +0000)]
ALSA: drivers: opl3: Fix incorrect use of vp->state
Static analysis with scan-build has found an assignment to vp2 that is
never used. It seems that the check on vp->state > 0 should be actually
on vp2->state instead. Fix this.
This dates back to 2002, I found the offending commit from the git
history git://git.kernel.org/pub/scm/linux/kernel/git/tglx/history.git,
commit 91e39521bbf6 ("[PATCH] ALSA patch for 2.5.4")
Takashi Sakamoto [Sat, 29 May 2021 03:33:53 +0000 (12:33 +0900)]
ALSA: pcm: comment about relation between msbits hw parameter and [S|U]32 formats
Regarding to handling [U|S][32|24] PCM formats, many userspace
application developers and driver developers have confusion, since they
require them to understand justification or padding. It easily
loses consistency and soundness to operate with many type of devices. In
this commit, I attempt to solve the situation by adding comment about
relation between [S|U]32 formats and 'msbits' hardware parameter.
The formats are used for 'left-justified' sample format, and the available
bit count in most significant bit is delivered to userspace in msbits
hardware parameter (struct snd_pcm_hw_params.msbits), which is decided by
msbits constraint added by pcm drivers (snd_pcm_hw_constraint_msbits()).
In driver side, the msbits constraint includes two elements; the physical
width of format and the available width of the format in most significant
bit. The former is used to match SAMPLE_BITS of format. (For my
convenience, I ignore wildcard in the usage of the constraint.)
As a result of interaction between ALSA pcm core and ALSA pcm application,
when the format in which SAMPLE_BITS equals to physical width of the
msbits constaint, the msbits parameter is set by referring to the
available width of the constraint. When the msbits parameter is not
changed in the above process, ALSA pcm core set it alternatively with
SAMPLE_BIT of chosen format.
In userspace application side, the msbits is only available after calling
ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS request. Even if the hardware
parameter structure includes somewhat value of SAMPLE_BITS interval
parameter as width of format, all of the width is not always available
since msbits can be less than the width.
I note that [S|U]24 formats are used for 'right-justified' 24 bit sample
formats within 32 bit frame. The first byte in most significant bit
should be invalidated. Although the msbits exposed to userspace should be
zero as invalid value, actually it is 32 from physical width of format.
Jaroslav Kysela [Fri, 10 Dec 2021 18:54:10 +0000 (18:54 +0000)]
kselftest: alsa: Use private alsa-lib configuration in mixer test
As mentined by Takashi Sakamoto, the system-wide alsa-lib configuration
may override the standard device declarations. This patch use the private
alsa-lib configuration to set the predictable environment.
Takashi Sakamoto [Fri, 10 Dec 2021 18:54:09 +0000 (18:54 +0000)]
kselftest: alsa: optimization for SNDRV_CTL_ELEM_ACCESS_VOLATILE
The volatile attribute of control element means that the hardware can
voluntarily change the state of control element independent of any
operation by software. ALSA control core necessarily sends notification
to userspace subscribers for any change from userspace application, while
it doesn't for the hardware's voluntary change.
This commit adds optimization for the attribute. Even if read value is
different from written value, the test reports success as long as the
target control element has the attribute. On the other hand, the
difference is itself reported for developers' convenience.
Mark Brown [Fri, 10 Dec 2021 18:54:08 +0000 (18:54 +0000)]
kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest
Add a basic test for the mixer control interface. For every control on
every sound card in the system it checks that it can read and write the
default value where the control supports that and for writeable controls
attempts to write all valid values, restoring the default values after
each test to minimise disruption for users.
There are quite a few areas for improvement - currently no coverage of the
generation of notifications, several of the control types don't have any
coverage for the values and we don't have any testing of error handling
when we attempt to write out of range values - but this provides some basic
coverage.
This is added as a kselftest since unlike other ALSA test programs it does
not require either physical setup of the device or interactive monitoring
by users and kselftest is one of the test suites that is frequently run by
people doing general automated testing so should increase coverage. It is
written in terms of alsa-lib since tinyalsa is not generally packaged for
distributions which makes things harder for general users interested in
kselftest as a whole but it will be a barrier to people with Android.
Jason Wang [Sun, 12 Dec 2021 07:04:22 +0000 (15:04 +0800)]
ALSA: sparc: no need to initialise statics to 0
Static variables do not need to be initialised to 0, because compiler
will initialise all uninitialised statics to 0. Thus, remove the
unneeded initializations.
Takashi Iwai [Tue, 7 Dec 2021 16:51:46 +0000 (17:51 +0100)]
ALSA: seq: Set upper limit of processed events
Currently ALSA sequencer core tries to process the queued events as
much as possible when they become dispatchable. If applications try
to queue too massive events to be processed at the very same timing,
the sequencer core would still try to process such all events, either
in the interrupt context or via some notifier; in either away, it
might be a cause of RCU stall or such problems.
As a potential workaround for those problems, this patch adds the
upper limit of the amount of events to be processed. The remaining
events are processed in the next batch, so they won't be lost.
For the time being, it's limited up to 1000 events per queue, which
should be high enough for any normal usages.
Takashi Iwai [Tue, 7 Dec 2021 15:33:23 +0000 (16:33 +0100)]
ALSA: mixart: Add sanity check for timer notify streams
The miXart timer notification is a variable length, and if a hardware
is screwed up, we may access over the actual data size. Let's add a
sanity check and bail out if an invalid value is received.
Anders Roxell [Tue, 7 Dec 2021 11:00:53 +0000 (12:00 +0100)]
ALSA: ppc: beep: fix clang -Wimplicit-fallthrough
Clang warns:
sound/ppc/beep.c:103:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough]
case SND_TONE: break;
^
sound/ppc/beep.c:103:2: note: insert 'break;' to avoid fall-through
case SND_TONE: break;
^
break;
1 warning generated.
Clang is more pedantic than GCC, which does not warn when failing
through to a case that is just break or return. Clang's version
is more in line with the kernel's own stance in deprecated.rst.
Add athe missing break to silence the warning.
Kees Cook [Tue, 7 Dec 2021 06:29:41 +0000 (22:29 -0800)]
ALSA: mixart: Reduce size of mixart_timer_notify
The mixart_timer_notify structure was larger than could be represented
by the mixart_msg_data array storage. Adjust the size to as large as
possible to fix the warning seen with -Warray-bounds builds:
sound/pci/mixart/mixart_core.c: In function 'snd_mixart_threaded_irq':
sound/pci/mixart/mixart_core.c:447:50: error: array subscript 'struct mixart_timer_notify[0]' is partly outside array bounds of 'u32[128]' {aka 'unsigned int[128]'} [-Werror=array-bounds]
447 | for(i=0; i<notify->stream_count; i++) {
| ^~
sound/pci/mixart/mixart_core.c:328:12: note: while referencing 'mixart_msg_data'
328 | static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4];
| ^~~~~~~~~~~~~~~
Takashi Iwai [Mon, 6 Dec 2021 16:25:10 +0000 (17:25 +0100)]
Merge tag 'asoc-fix-v5.16-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.16
A relatively large collection of updates, the size is increased quite a
bit by there being some repetitive changes for similar issues that occur
multiple times with both notifying control value changes and runtime PM.
The Rockchip update looks at first glance like a cleanup but fixes
instantiation of the hardware on some systems.
Olivia Mackintosh has posted to alsa-devel reporting that
there's a potential bug that could break mixer quirks for Pioneer
devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7
"ALSA: usb-audio: Add support for the Pioneer DJM 750MK2
Mixer/Soundcard".
This happened because the DJM 750 MK2 was added last to the Pioneer DJM
device table index and defined as 0x4 but was added to snd_djm_devices[]
just after the DJM 750 (MK1) entry instead of last, after the DJM 900
NXS2. This escaped review.
To prevent that from ever happening again, Takashi Iwai suggested to use
C99 array designators in snd_djm_devices[] instead of simply reordering
the entries.
Fixes: 6d2778816036 ("ALSA: usb-audio: Add support for the Pioneer DJM 750MK2") Reported-by: Olivia Mackintosh <livvy@base.nu> Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com> Link: https://lore.kernel.org/r/Yau46FDzoql0SNnW@geday Signed-off-by: Takashi Iwai <tiwai@suse.de>
Werner Sembach [Thu, 2 Dec 2021 16:50:10 +0000 (17:50 +0100)]
ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1
This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This
fixes the issue of sound not working after s3 suspend.
When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of
0x0020. Setting the value manually makes the sound work again. This patch
does this automatically.
While being on it, I also fixed the comment formatting of the quirk and
shortened variable and function names.
Alan Young [Thu, 2 Dec 2021 15:06:07 +0000 (15:06 +0000)]
ALSA: ctl: Fix copy of updated id with element read/write
When control_compat.c:copy_ctl_value_to_user() is used, by
ctl_elem_read_user() & ctl_elem_write_user(), it must also copy back the
snd_ctl_elem_id value that may have been updated (filled in) by the call
to snd_ctl_elem_read/snd_ctl_elem_write().
This matches the functionality provided by snd_ctl_elem_read_user() and
snd_ctl_elem_write_user(), via snd_ctl_build_ioff().
Without this, and without making additional calls to snd_ctl_info()
which are unnecessary when using the non-compat calls, a userspace
application will not know the numid value for the element and
consequently will not be able to use the poll/read interface on the
control file to determine which elements have updates.
Takashi Iwai [Wed, 1 Dec 2021 07:36:06 +0000 (08:36 +0100)]
ALSA: pcm: oss: Handle missing errors in snd_pcm_oss_change_params*()
A couple of calls in snd_pcm_oss_change_params_locked() ignore the
possible errors. Catch those errors and abort the operation for
avoiding further problems.
Takashi Iwai [Wed, 1 Dec 2021 07:36:05 +0000 (08:36 +0100)]
ALSA: pcm: oss: Limit the period size to 16MB
Set the practical limit to the period size (the fragment shift in OSS)
instead of a full 31bit; a too large value could lead to the exhaust
of memory as we allocate temporary buffers of the period size, too.
As of this patch, we set to 16MB limit, which should cover all use
cases.
Takashi Iwai [Wed, 1 Dec 2021 07:36:04 +0000 (08:36 +0100)]
ALSA: pcm: oss: Fix negative period/buffer sizes
The period size calculation in OSS layer may receive a negative value
as an error, but the code there assumes only the positive values and
handle them with size_t. Due to that, a too big value may be passed
to the lower layers.
This patch changes the code to handle with ssize_t and adds the proper
error checks appropriately.
Srinivas Kandagatla [Tue, 30 Nov 2021 16:05:07 +0000 (16:05 +0000)]
ASoC: codecs: wsa881x: fix return values from kcontrol put
wsa881x_set_port() and wsa881x_put_pa_gain() currently returns zero eventhough
it changes the value. Fix this, so that change notifications are sent
correctly.
Srinivas Kandagatla [Tue, 30 Nov 2021 16:05:04 +0000 (16:05 +0000)]
ASoC: codecs: wcd934x: handle channel mappping list correctly
Currently each channel is added as list to dai channel list, however
there is danger of adding same channel to multiple dai channel list
which endups corrupting the other list where its already added.
This patch ensures that the channel is actually free before adding to
the dai channel list and also ensures that the channel is on the list
before deleting it.
This check was missing previously, and we did not hit this issue as
we were testing very simple usecases with sequence of amixer commands.
Bixuan Cui [Wed, 1 Dec 2021 08:58:54 +0000 (16:58 +0800)]
ALSA: oss: fix compile error when OSS_DEBUG is enabled
Fix compile error when OSS_DEBUG is enabled:
sound/core/oss/pcm_oss.c: In function 'snd_pcm_oss_set_trigger':
sound/core/oss/pcm_oss.c:2055:10: error: 'substream' undeclared (first
use in this function); did you mean 'csubstream'?
pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger);
^
Amadeusz Sławiński [Fri, 26 Nov 2021 14:03:55 +0000 (15:03 +0100)]
ASoC: Intel: Skylake: Use NHLT API to search for blob
With NHLT enriched with new search functions, remove local code in
favour of them. This also fixes broken behaviour: search should be based
on significant bits count rather than container size.
Amadeusz Sławiński [Fri, 26 Nov 2021 14:03:53 +0000 (15:03 +0100)]
ALSA: hda: Fill gaps in NHLT endpoint-interface
Two key operations missings are: endpoint presence-check and retrieval
of matching endpoint hardware configuration (blob). Add operations for
both use cases.
Hui Wang [Tue, 30 Nov 2021 09:06:06 +0000 (11:06 +0200)]
ASoC: SOF: Intel: Retry codec probing if it fails
On the latest Lenovo Thinkstation laptops, we often experience the
speaker failure after rebooting, check the dmesg, we could see:
sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5
The analogue codec on the machine is ALC287, then we designed a
testcase to reboot and check the codec probing result repeatedly, we
found the analogue codec probing always failed at least once within
several minutes to several hours (roughly 1 reboot per min). This
issue happens on all laptops of this Thinkstation model, but with
legacy HDA driver, we couldn't reproduce this issue on those laptops.
And so far, this issue is not reproduced on machines which don't
belong to this model.
We tried to make the hda_dsp_ctrl_init_chip() same as
hda_intel_init_chip() which is the controller init routine in the
legacy HDA driver, but it didn't help.
We found when issue happens, the resp is -1, and if we let driver
re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the
desktop and all audio function work well.
Here adding codec probing retries to 3 times, it could fix the issue
on this Thinkstation model, and it doesn't bring impact to other
machines.
Reviewed-by: Bard Liao <bard.liao@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20211130090606.529348-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dan Carpenter [Tue, 30 Nov 2021 12:56:33 +0000 (15:56 +0300)]
ASoC: amd: fix uninitialized variable in snd_acp6x_probe()
The "index" is potentially used without being initialized on the error
path.
Fixes: fc329c1de498 ("ASoC: amd: add platform devices for acp6x pdm driver and dmic driver") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Link: https://lore.kernel.org/r/20211130125633.GA24941@kili Signed-off-by: Mark Brown <broonie@kernel.org>
Nicolas Frattaroli [Thu, 25 Nov 2021 08:48:59 +0000 (09:48 +0100)]
ASoC: rockchip: i2s_tdm: Dup static DAI template
Previously, the DAI template was used directly, which lead to
fun bugs such as "why is my channels_max changing?" when one
instantiated more than one i2s_tdm IP block in a device tree.
This change makes it so that we instead duplicate the template
struct, and then use that.
Thomas Gleixner [Wed, 24 Nov 2021 22:40:01 +0000 (23:40 +0100)]
ALSA: hda: Make proper use of timecounter
HDA uses a timecounter to read a hardware clock running at 24 MHz. The
conversion factor is set with a mult value of 125 and a shift value of 0,
which is not converting the hardware clock to nanoseconds, it is converting
to 1/3 nanoseconds because the conversion factor from 24Mhz to nanoseconds
is 125/3. The usage sites divide the "nanoseconds" value returned by
timecounter_read() by 3 to get a real nanoseconds value.
There is a lengthy comment in azx_timecounter_init() explaining this
choice. That comment makes blatantly wrong assumptions about how
timecounters work and what can overflow.
The comment says:
* Applying the 1/3 factor as part of the multiplication
* requires at least 20 bits for a decent precision, however
* overflows occur after about 4 hours or less, not a option.
timecounters operate on time deltas between two readouts of a clock and use
the mult/shift pair to calculate a precise nanoseconds value:
delta_nsec = (delta_clock * mult) >> shift;
The fractional part is also taken into account and preserved to prevent
accumulated rounding errors. For details see cyclecounter_cyc2ns().
The mult/shift pair has to be chosen so that the multiplication of the
maximum expected delta value does not result in a 64bit overflow. As the
counter wraps around on 32bit, the maximum observable delta between two
reads is (1 << 32) - 1 which is about 178.9 seconds.
That in turn means the maximum multiplication factor which fits into an u32
will not cause a 64bit overflow ever because it's guaranteed that:
((1 << 32) - 1) ^ 2 < (1 << 64)
The resulting correct multiplication factor is 2796202667 and the shift
value is 26, i.e. 26 bit precision. The overflow of the multiplication
would happen exactly at a clock readout delta of 6597069765 which is way
after the wrap around of the hardware clock at around 274.8 seconds which
is off from the claimed 4 hours by more than an order of magnitude.
If the counter ever wraps around the last read value then the calculation
is off by the number of wrap arounds times 178.9 seconds because the
overflow cannot be observed.
Use clocks_calc_mult_shift(), which calculates the most accurate mult/shift
pair based on the given clock frequency, and remove the bogus comment along
with the divisions at the readout sites.
Fixes: 5d890f591d15 ("ALSA: hda: support for wallclock timestamps") Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/871r35kwji.ffs@tglx Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stefan Binding [Sun, 28 Nov 2021 11:55:58 +0000 (11:55 +0000)]
ALSA: hda/cs8409: Set PMSG_ON earlier inside cs8409 driver
For cs8409, it is required to run Jack Detect on resume.
Jack Detect on cs8409+cs42l42 requires an interrupt from
cs42l42 to be sent to cs8409 which is propogated to the driver
via an unsolicited event.
However, the hda_codec drops unsolicited events if the power_state
is not set to PMSG_ON. Which is set at the end of the resume call.
This means there is a race condition between setting power_state
to PMSG_ON and receiving the interrupt.
To solve this, we can add an API to set the power_state earlier
and call that before we start Jack Detect.
This does not cause issues, since we know inside our driver that
we are already initialized, and ready to handle the unsolicited
events.
Mark Brown [Sat, 27 Nov 2021 01:27:20 +0000 (01:27 +0000)]
Suspend related fixes on Tegra
Merge series from Sameer Pujar <spujar@nvidia.com>:
This series addresses following problems:
* The runtime PM is not balanced in MVC driver, whenever
mute or volume mixer controls are set.
* Some of the AHUB devices (SFC, MVC, Mixer, AMX and ADX)
use late system sleep. Suspend failure is seen on Jetson
TX2 platform.
Rob Clark [Thu, 18 Nov 2021 01:04:53 +0000 (17:04 -0800)]
ASoC: rt5682s: Fix crash due to out of scope stack vars
Move the declaration of temporary arrays to somewhere that won't go out
of scope before the devm_clk_hw_register() call, lest we be at the whim
of the compiler for whether those stack variables get overwritten.
Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1)
Fixes: bdd229ab26be ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Rob Clark <robdclark@chromium.org> Reviewed-by: Stephen Boyd <swboyd@chromium.org> Link: https://lore.kernel.org/r/20211118010453.843286-2-robdclark@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Rob Clark [Thu, 18 Nov 2021 01:04:52 +0000 (17:04 -0800)]
ASoC: rt5682: Fix crash due to out of scope stack vars
Move the declaration of temporary arrays to somewhere that won't go out
of scope before the devm_clk_hw_register() call, lest we be at the whim
of the compiler for whether those stack variables get overwritten.
Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1)
Fixes: edbd24ea1e5c ("ASoC: rt5682: Drop usage of __clk_get_name()") Signed-off-by: Rob Clark <robdclark@chromium.org> Reviewed-by: Stephen Boyd <swboyd@chromium.org> Link: https://lore.kernel.org/r/20211118010453.843286-1-robdclark@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Sameer Pujar [Tue, 23 Nov 2021 14:07:39 +0000 (19:37 +0530)]
ASoC: tegra: Use normal system sleep for ADX
The driver currently subscribes for a late system sleep call.
The initcall_debug log shows that suspend call for ADX device
happens after the parent device (AHUB). This seems to cause
suspend failure on Jetson TX2 platform. Also there is no use
of having late system sleep specifically for ADX device. Fix
the order by using normal system sleep.
Sameer Pujar [Tue, 23 Nov 2021 14:07:38 +0000 (19:37 +0530)]
ASoC: tegra: Use normal system sleep for AMX
The driver currently subscribes for a late system sleep call.
The initcall_debug log shows that suspend call for AMX device
happens after the parent device (AHUB). This seems to cause
suspend failure on Jetson TX2 platform. Also there is no use
of having late system sleep specifically for AMX device. Fix
the order by using normal system sleep.
Sameer Pujar [Tue, 23 Nov 2021 14:07:37 +0000 (19:37 +0530)]
ASoC: tegra: Use normal system sleep for Mixer
The driver currently subscribes for a late system sleep call.
The initcall_debug log shows that suspend call for Mixer device
happens after the parent device (AHUB). This seems to cause
suspend failure on Jetson TX2 platform. Also there is no use
of having late system sleep specifically for Mixer device. Fix
the order by using normal system sleep.
Sameer Pujar [Tue, 23 Nov 2021 14:07:36 +0000 (19:37 +0530)]
ASoC: tegra: Use normal system sleep for MVC
The driver currently subscribes for a late system sleep call.
The initcall_debug log shows that suspend call for MVC device
happens after the parent device (AHUB). This seems to cause
suspend failure on Jetson TX2 platform. Also there is no use
of having late system sleep specifically for MVC device. Fix
the order by using normal system sleep.
Sameer Pujar [Tue, 23 Nov 2021 14:07:35 +0000 (19:37 +0530)]
ASoC: tegra: Use normal system sleep for SFC
The driver currently subscribes for a late system sleep call.
The initcall_debug log shows that suspend call for SFC device
happens after the parent device (AHUB). This seems to cause
suspend failure on Jetson TX2 platform. Also there is no use
of having late system sleep specifically for SFC device. Fix
the order by using normal system sleep.
Sameer Pujar [Tue, 23 Nov 2021 14:07:34 +0000 (19:37 +0530)]
ASoC: tegra: Balance runtime PM count
After successful application of volume/mute settings via mixer control
put calls, the control returns without balancing the runtime PM count.
This makes device to be always runtime active. Fix this by allowing
control to reach pm_runtime_put() call.
Takashi Iwai [Thu, 25 Nov 2021 13:35:24 +0000 (14:35 +0100)]
Merge tag 'asoc-fix-v5.16-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.16
There's a large but repetitive set of fixes here for issues with the
Tegra kcontrols not correctly reporting changes to userspace, a fix for
some issues with matching on older x86 platforms introduced during the
merge window together with a set of smaller fixes and one new system
quirk.
Both, hardware and drivers does support interrupts.
Fix warnings as:
arch/arm/boot/dts/tegra30-microsoft-surface-rt-efi.dt.yaml: audio-codec@1a: 'interrupt-parent', 'interrupts' do not match any of the regexes: 'pinctrl-[0-9]+'
From schema: /home/runner/work/linux/linux/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml
Fixes: cd51b942f344 ("ASoC: dt-bindings: wlf,wm8962: Convert to json-schema") Signed-off-by: David Heidelberg <david@ixit.cz> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be> Link: https://lore.kernel.org/r/20211124155101.59694-1-david@ixit.cz Signed-off-by: Mark Brown <broonie@kernel.org>
Ranjani Sridharan [Tue, 23 Nov 2021 16:57:59 +0000 (18:57 +0200)]
ASoC: SOF: hda: reset DAI widget before reconfiguring it
It is not unusual for ALSA/ASoC hw_params callbacks to be invoked
multiple times. Reset and free the DAI widget before reconfiguring
it to keep the DAI widget use_count balanced.
Fixes: 0acb48dd31e3 ("ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com> Reviewed-by: Bard Liao <bard.liao@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20211123165759.127884-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Lucas Tanure [Tue, 23 Nov 2021 16:31:39 +0000 (16:31 +0000)]
ASoC: cs35l41: Set the max SPI speed for the whole device
Higher speeds are only supported when PLL is enabled, but
the current driver doesn't enable PLL outside of stream
use cases, so better to set the lowest SPI speed accepted
by the entire device.
Move the current frequency set to the spi sub-driver so
the whole device can benefit from that speed.
spi-max-frequency property could be used, but ACPI systems don't
support it, so by setting it in the spi sub-driver probe
both Device Trees and ACPI systems are supported.
Hans de Goede [Thu, 18 Nov 2021 15:30:14 +0000 (16:30 +0100)]
ASoC: soc-acpi: Set mach->id field on comp_ids matches
Commit dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using
compatible IDs") and commit 959ae8215a9e ("ASoC: Intel: soc-acpi-cht:
shrink tables using compatible IDs") simplified the match tables in
soc-acpi-intel-byt-match.c and soc-acpi-intel-cht-match.c by merging
identical entries using the new .comp_ids snd_soc_acpi_mach field to
point a single entry to multiple ACPI HIDs and clearing the previously
unique per entry .id field.
But various machine drivers from sound/soc/intel/boards rely on mach->id
in one or more ways, e.g. some drivers contain the following snippets:
if (!strncmp(snd_soc_cards[i].codec_id, mach->id, 8)) { ...
All of which are broken by the match table shrinking.
Make the snd_soc_acpi_mach.id field non const (the storage for the tables
already is non const) and on a comps_ids match copy the matching HID to
the id field to fix this.
Fixes: dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs") Fixes: 959ae8215a9e ("ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs") Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Brent Lu <brent.lu@intel.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211118153014.349222-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Fri, 19 Nov 2021 16:27:30 +0000 (17:27 +0100)]
ALSA: hda: Remove redundant runtime PM calls
The previous fix for more comprehensive runtime PM calls turned out to
be not good as hoped; a few calls including pm_runtime_enable() and
pm_runtime_disable() are rather utterly superfluous for PCI devices,
even triggering a kernel error message. Better to drop those calls.
Note that the problem we wanted to solve with that commit seems
irrelevant with the fix itself; the original bug (a GPF at
azx_remove()) was likely a regression by the recent PCI core cleanup,
and the buggy PCI change has been already reverted. So basically we
were scratching a wrong surface. OTOH, making the runtime PM calls
symmetric for both probe and remove is more consistent, and maybe
that's a sensible outcome.
Pierre-Louis Bossart [Wed, 27 Oct 2021 02:32:54 +0000 (10:32 +0800)]
ALSA: intel-dsp-config: add quirk for JSL devices based on ES8336 codec
These devices are based on an I2C/I2S device, we need to force the use
of the SOF driver otherwise the legacy HDaudio driver will be loaded -
only HDMI will be supported.
We previously added support for other Intel platforms but missed
JasperLake.
Takashi Iwai [Fri, 19 Nov 2021 10:24:59 +0000 (11:24 +0100)]
ALSA: usb-audio: Switch back to non-latency mode at a later point
The recent regression report revealed that the judgment of the
low-latency playback mode based on the runtime->stop_threshold cannot
work reliably at the prepare stage, as sw_params call may happen at
any time, and PCM dmix actually sets it up after the prepare call.
This ended up with the stall of the stream as PCM ack won't be issued
at all.
For addressing this, check the free-wheeling mode again at the PCM
trigger right before starting the stream again, and allow switching to
the non-LL mode at a late stage.
Takashi Iwai [Thu, 18 Nov 2021 21:57:29 +0000 (22:57 +0100)]
ALSA: ctxfi: Fix out-of-range access
The master and next_conj of rcs_ops are used for iterating the
resource list entries, and currently those are supposed to return the
current value. The problem is that next_conf may go over the last
entry before the loop abort condition is evaluated, and it may return
the "current" value that is beyond the array size. It was caught
recently as a GPF, for example.
Those return values are, however, never actually evaluated, hence
basically we don't have to consider the current value as the return at
all. By dropping those return values, the potential out-of-range
access above is also fixed automatically.
This patch changes the return type of master and next_conj callbacks
to void and drop the superfluous code accordingly.
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