Takashi Sakamoto [Sun, 17 Mar 2024 02:40:50 +0000 (11:40 +0900)]
ALSA: core: add kunitconfig
It is helpful to add .kunitconfig if we work with the tools provided by
KUnit project. The file describes the series of kernel configurations to
satisfy the dependency to build the target test.
Ian Murphy [Sat, 16 Mar 2024 09:41:57 +0000 (09:41 +0000)]
ALSA: hda/realtek: add in quirk for Acer Swift Go 16 - SFG16-71
Keyboard has an LED that is ON/OFF when mic is muted/active
- LED is controlled by GPIO pin
- Patch enables led to appear in /sys/class/leds/ as hda::micmute
- Enables LED when mic is MUTED
- Disables LED when mic is active
[ fixed white spaces by tiwai ]
Signed-off-by: Ian Murphy <iano200@gmail.com>
Message-ID: <20240316094157.13890-1-iano200@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fix is applied globally to all devices, and it may change the
existing control names. When the devices are managed with the fixed
configuration like UCM, such control name mismatch may lead to
significant regressions.
For avoiding that kind of regression, we would need to apply such
changes conditionally, but it'd take time to settle down.
While the original fix is a good thing in general, in order to address
the regression, let's revert the change for now.
Jichi Zhang [Fri, 15 Mar 2024 08:19:56 +0000 (01:19 -0700)]
ALSA: hda/realtek: Add quirk for Lenovo Yoga 9 14IMH9
The speakers on the Lenovo Yoga 9 14IMH9 are similar to previous generations
such as the 14IAP7, and the bass speakers can be fixed using similar methods
with one caveat: 14IMH9 uses CS35L41 amplifiers which need to be activated
separately.
Valentine Altair [Tue, 12 Mar 2024 14:42:00 +0000 (14:42 +0000)]
ALSA: hda/realtek - ALC236 fix volume mute & mic mute LED on some HP models
Some HP laptops have received revisions that altered their board IDs
and therefore the current patches/quirks do not apply to them.
Specifically, for my Probook 440 G8, I have a board ID of 8a74.
It is necessary to add a line for that specific model.
Takashi Iwai [Mon, 11 Mar 2024 15:18:47 +0000 (16:18 +0100)]
Merge tag 'asoc-v6.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.9
This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
Geoffrey D. Bennett [Mon, 11 Mar 2024 09:26:08 +0000 (19:56 +1030)]
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again
The 4th Gen input preamp gain range is 0dB to +69dB, although the
control values range from 0 to 70. Replace SCARLETT2_MAX_GAIN with
SCARLETT2_MAX_GAIN_VALUE and SCARLETT2_MAX_GAIN_DB, and update the TLV
again.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu> Fixes: a45cf0a08347 ("ALSA: scarlett2: Fix Scarlett 4th Gen input gain range")
Message-ID: <Ze7OMA8ntG7KteGa@m.b4.vu> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Geoffrey D. Bennett [Sun, 10 Mar 2024 10:34:59 +0000 (21:04 +1030)]
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range
The input gain range TLV was declared as -70dB to 0dB, but the preamp
gain range is actually 0dB to +70dB. Rename SCARLETT2_GAIN_BIAS to
SCARLETT2_MAX_GAIN and update the TLV to fix.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu> Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain") Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <9168317b5ac5335943d3f14dbcd1cc2d9b2299d0.1710047969.git.g@b4.vu>
Geoffrey D. Bennett [Sun, 10 Mar 2024 10:34:41 +0000 (21:04 +1030)]
ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values
The meanings of the raw_auto_gain_status values were originally
guessed through experimentation, but the official names have now been
discovered. Update the autogain status control strings accordingly.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu> Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain") Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <8bd12a5e7dc714801dd9887c4bc5cb35c384e27c.1710047969.git.g@b4.vu>
Geoffrey D. Bennett [Sun, 10 Mar 2024 10:34:27 +0000 (21:04 +1030)]
ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection
The value currently being read to determine the low-voltage state is
actually the front panel state. Fix the code to use the correct offset
for the low-voltage state.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu> Fixes: d7cfa2fdfc8a ("ALSA: scarlett2: Add power status control") Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d97b7d87f43b0e54f37e1552394be2f3ae182704.1710047969.git.g@b4.vu>
Gergo Koteles [Fri, 8 Mar 2024 17:41:43 +0000 (18:41 +0100)]
ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend
The runtime_resume function calls prmg_load and apply_calibration
functions, but system_resume also calls them, so calling
pm_runtime_force_resume before reset is unnecessary.
For consistency, do not call the pm_runtime_force_suspend in
system_suspend, as runtime_suspend does the same.
Athaariq Ardhiansyah [Sun, 10 Mar 2024 13:58:44 +0000 (20:58 +0700)]
ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops
Realtek codec on HP Envy laptop series are heavily modified by vendor.
Therefore, need intervention to make it work properly. The patch fixes:
- B&O soundbar speakers (between lid and keyboard) activation
- Enable LED on mute button
- Add missing process coefficient which affects the output amplifier
- Volume control synchronization between B&O soundbar and side speakers
- Unmute headset output on several HP Envy models
- Auto-enable headset mic when plugged
This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285
The only unsolved problem is output amplifier of all built-in speakers
is too weak, which causes volume of built-in speakers cannot be loud
as vendor's proprietary driver due to missing _DSD parameter in the
firmware. The solution is currently on research. Expected to has another
patch in the future.
Potential fix to related issues, need test before close those issues:
Simon Trimmer [Fri, 8 Mar 2024 13:58:59 +0000 (13:58 +0000)]
ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57
Add the HID for the CS35L54 and CS35L57 Boosted Smart Amplifiers. These
have the same control interface as the CS35L56 so are handled by the
cs35l56-hda driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-3-rf@opensource.cirrus.com>
Simon Trimmer [Fri, 8 Mar 2024 13:58:58 +0000 (13:58 +0000)]
ASoC: cs35l56: Add support for CS35L54 and CS35L57
The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has
I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire
control and audio.
The hardware differences between L54, L56 and L57 do not affect the
driver control interface so they can all be handled by the same driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Acked-by: Mark Brown <broonie@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-2-rf@opensource.cirrus.com>
Richard Fitzgerald [Thu, 7 Mar 2024 11:12:16 +0000 (11:12 +0000)]
ALSA: hda: hda_component: Include sound/hda_codec.h
hda_component.h uses hda_codec_dev from sound/hda_codec.h.
Include sound/hda_codec.h instead of assuming that it has already
been included by the parent .c file.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-2-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the conventional include guards around the content of the
hda_component.h header file. This prevents double-declaration of
struct hda_component if the header gets included multiple times.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-1-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Thu, 7 Mar 2024 15:20:35 +0000 (15:20 +0000)]
ASoC: codecs: ES8326: change members of private
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We found that using 0x45 as the default value for interrupt-clk
would cause a headset detection error.So we took 0x00 as the default
value for interrupt-clk and passed the test.
We removed mic1-src and mic2-src, which were not used.
There are clearly multiple calls, one per component, but they cannot be
discriminated from each other.
Change the ftrace events to also print the component name, to make it clear
which part of the code is involved. This requires changing the passed value
from a struct snd_soc_card, where the DAPM context is not kwown, to a
struct snd_soc_dapm_context where it is obviously known but the a card
pointer is also available.
intel-mid.h is providing some core parts of the South Complex PM,
which are usually are not used by individual drivers. In particular,
this driver doesn't use it, so simply remove the unused header.
The documented vendor prefix for LG Electronics is 'lg' not 'lge'. Just
change the example to 'lg' as there doesn't appear to be any dependency
on the existing compatible string.
Chancel Liu [Mon, 4 Mar 2024 07:21:28 +0000 (16:21 +0900)]
ASoC: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get()
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name
for dlc (snd_soc_dai_link_component). In this function call
dlc->dai_name is parsed via snd_soc_dai_name_get() (B).
(B) has a priority to return dai->name as dlc->dai_name. In most cases
card can probe successfully. However it has an issue that ASoC tries to
rebind card. Here is a simplified flow for example:
| a) Card probes successfully at first
| b) One of the component bound to this card is removed for some
| reason the component->dev is released
| c) That component is re-registered
v d) ASoC calls snd_soc_try_rebind_card()
a) points dlc->dai_name to dai->name. b) releases all resource of the
old DAI. c) creates new DAI structure. In result d) can not use
dlc->dai_name to add new created DAI.
So it's reasonable that prefer to return dai->driver->name in
snd_soc_dai_name_get() because dai->driver is a pre-defined global
variable. Also update snd_soc_is_matching_dai() for alignment.
Richard Fitzgerald [Mon, 4 Mar 2024 14:37:05 +0000 (14:37 +0000)]
ASoC: cs-amp-lib: Add KUnit test for calibration helpers
Add a KUnit test for the cs-amp-lib library. This has test cases
for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs().
A KUNIT_STATIC_STUB_REDIRECT() has been added to
cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the
KUnit test can redirect these to test harness functions.
Much of the testing involves invoking the same function with different
parameters, i.e. the number of amps and the amp index within the array.
This uses parameterization rather than looping. The idea is to avoid
looping over configurations within one test case as that has a higher
chance of having a bug that doesn't actually test all the expected cases.
Having the test run exactly one configuration, and then tear-down, is less
prone to accidentally skipped configurations.
Andreas Pape [Fri, 1 Mar 2024 08:50:03 +0000 (09:50 +0100)]
ASoC: rcar: adg: correct TIMSEL setting for SSI9
Timing select registers for SRC and CMD are by default
referring to the corresponding SSI word select.
The calculation rule from HW spec skips SSI8, which has
no clock connection.
Fix the erroneous prohibited setting of timsel value 1111 (0xf) for SSI9
by using timsel value 1110 (0xe) instead. This is possible because SSI8
is not connected as shown by <GAP> in the table above.
Cezary Rojewski [Mon, 26 Feb 2024 12:44:30 +0000 (13:44 +0100)]
ASoC: Intel: avs: Ignore codecs with no suppoting driver
HDMI codecs which are present and functional from audio perspective lack
i915 support on drm side what results in -ENODEV during the probing
sequence. There is no reason to perform recovery procedure e.g.: reset
the HDAudio controller if this is the case.
Cezary Rojewski [Mon, 26 Feb 2024 12:44:29 +0000 (13:44 +0100)]
ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing
If i915 does not support given platform but the hardware i.e.: HDAudio
codec is still there, the codec-probing procedure will succeed for such
device but the follow up initialization will always end up with -ENODEV.
While bus could filter out address '2' which Intel's HDMI/DP codecs
always enumerate on, more robust approach is to check for i915 presence
before registering display codecs.
Cezary Rojewski [Mon, 26 Feb 2024 12:44:28 +0000 (13:44 +0100)]
ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms
Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends
removed support for i915 for all CNL-based platforms. HDAudio library,
however, still treats such platforms as valid candidates for i915
binding. Update query mechanism to reflect changes made in drm tree.
At the same time, i915 support for LKF-based platforms has not been
provided so remove them from valid binding candidates.
Kenny Levinsen [Fri, 1 Mar 2024 23:11:07 +0000 (00:11 +0100)]
ALSA: usb-audio: Name feature ctl using output if input is PCM
When building feature controls from a unit without a name, we try to
derive a name first from the feature unit's input, then fall back to the
output terminal.
If a feature unit connects directly to a "USB Streaming" input terminal
rather than a mixer or other virtual type, the control receives the
somewhat meaningless name "PCM", even if the output had a descriptive
type such as "Headset" or "Speaker".
Here is an example of such AudioControl descriptor from a USB headset
which ends up named "PCM Playback" and is therefore not recognized as
headphones by userspace:
Other headsets and DACs I tried that used their output terminal for
naming only did so due to their input being an unnamed sidetone mixer.
Instead of always starting with the input terminal, check the type of it
first. If it seems uninteresting, invert the order and use the output
terminal first for naming.
This makes userspace recognize headsets with simple controls as
headphones, and leads to more consistent naming of playback devices
based on their outputs irrespective of sidetone mixers.
Stefan Binding [Fri, 1 Mar 2024 16:01:54 +0000 (16:01 +0000)]
ALSA: hda: cs35l41: Overwrite CS35L41 configuration for ASUS UM5302LA
Whilst this laptop contains _DSD inside the BIOS, there is an error in
this configuration. Override the _DSD in the BIOS with the correct
configuration for this laptop.
Nathan Chancellor [Fri, 1 Mar 2024 17:08:22 +0000 (10:08 -0700)]
ALSA: hwdep: Move put_user() call out of scoped_guard() in snd_hwdep_control_ioctl()
Clang prior to 17.0.0 has a bug in its asm goto jump scope analysis to
determine that no variables with the cleanup attribute are skipped by an
indirect jump. Instead of only checking the scope of each label that is
a possible target of each asm goto statement, it checks the scope of
every label, which can cause an error when a variable with the cleanup
attribute is used between two asm goto statements with different scopes,
even if they have completely different label targets:
sound/core/hwdep.c:273:8: error: cannot jump from this asm goto statement to one of its possible targets
if (get_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user'
__get_user(x, _gu_addr) : \
^
arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user'
__get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \
^
arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed'
__get_user_size_goto(x, ptr, size, __gus_failed); \
^
arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto'
case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \
^
arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto'
asm_volatile_goto( \
^
include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto'
#define asm_volatile_goto(x...) asm goto(x)
^
sound/core/hwdep.c:291:9: note: possible target of asm goto statement
if (put_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user'
__put_user(x, _pu_addr) : -EFAULT; \
^
arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user'
\
^
sound/core/hwdep.c:276:4: note: jump bypasses initialization of variable with __attribute__((cleanup))
scoped_guard(mutex, ®ister_mutex) {
^
include/linux/cleanup.h:169:20: note: expanded from macro 'scoped_guard'
for (CLASS(_name, scope)(args), \
To avoid this issue, move the put_user() call out of the scoped_guard()
scope, which allows the asm goto scope analysis to see that the variable
with the cleanup attribute will never be skipped by the asm goto
statements.
There should be no functional change because prior to the refactoring,
put_user() was not called under register_mutex, so this call does not
even need to be in the scoped_guard() in the first place.
songxiebing [Fri, 1 Mar 2024 01:18:41 +0000 (09:18 +0800)]
ALSA: hda: optimize the probe codec process
In azx_probe_codecs function, when bus->codec_mask is becomes to 0(no codecs),
execute azx_init_chip, bus->codec_mask will be initialized to a value again,
this causes snd_hda_codec_new function to run, the process is as follows:
-->snd_hda_codec_new
-->snd_hda_codec_device_init
-->snd_hdac_device_init---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_SUBSYSTEM_ID) 2s
---snd_hdac_read_parm(...AC_PAR_REV_ID) 2s
---snd_hdac_read_parm(...AC_PAR_NODE_COUNT) 2s
when no codecs, read communication is error, each command will be polled for
2 second, a total of 10s, it is easy to some problem.
like this:
2 [ 14.833404][ 6] [ T164] hda 0006:00: Codec #0 probe error; disabling it...
3 [ 14.844178][ 6] [ T164] hda 0006:00: codec_mask = 0x1
4 [ 14.880532][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
5 [ 15.891988][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
6 [ 16.978090][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0001
7 [ 18.140895][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0002
8 [ 19.135516][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0004
10 [ 19.900086][ 6] [ T164] hda 0006:00: no codecs initialized
11 [ 45.573398][ 2] [ C2] watchdog: BUG: soft lockup - CPU#2 stuck for 22s! [kworker/2:0:25]
Here, when bus->codec_mask is 0, use a direct break to avoid execute snd_hda_codec_new function.
Takashi Iwai [Thu, 29 Feb 2024 07:29:04 +0000 (08:29 +0100)]
Merge tag 'asoc-fix-v6.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
A few small fixes, some driver specific and one slightly larger one
from Richard which adds a new core helper and updates a small clutch of
drivers to deal with the fact that they were using a helper which
requires that the lock for the list of controls without holding that
lock. We also have some quirks for new AMD based Lenovo systems.
Takashi Iwai [Tue, 27 Feb 2024 08:52:51 +0000 (09:52 +0100)]
ALSA: rawmidi: Use guard() for locking
We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.
There are a few remaining explicit mutex and spinlock calls, and those
are the places where the temporary unlock/relocking happens -- which
guard() doens't cover well yet.
Only the code refactoring, and no functional changes.
Takashi Iwai [Tue, 27 Feb 2024 08:52:50 +0000 (09:52 +0100)]
ALSA: control: Use guard() for locking
We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.
The lops calls under multiple rwsems are factored out as a simple
macro, so that it can be called easily from snd_ctl_dev_register()
and snd_ctl_dev_disconnect().
There are a few remaining explicit rwsem and spinlock calls, and those
are the places where the lock downgrade happens or where the temporary
unlock/relocking happens -- which guard() doens't cover well yet.
Only the code refactoring, and no functional changes.
Takashi Iwai [Tue, 27 Feb 2024 08:52:47 +0000 (09:52 +0100)]
ALSA: hwdep: Use guard() for locking
We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.
There are still a few remaining explicit mutex_lock/unlock calls, and
those are for the places where we do temporary unlock/relock, which
doesn't fit well with the guard(), so far.
Only the code refactoring, and no functional changes.
Takashi Iwai [Tue, 27 Feb 2024 08:52:45 +0000 (09:52 +0100)]
ALSA: timer: Use guard() for locking
We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.
For making changes easier, some functions widen the application of
register_mutex, but those shouldn't influence on any actual
performance.
Also, one code block was factored out as a function so that guard()
can be applied cleanly without much indentation.
There are still a few remaining explicit spin_lock/unlock calls, and
those are for the places where we do temporary unlock/relock, which
doesn't fit well with the guard(), so far.
Only the code refactoring, and no functional changes.
Richard Fitzgerald [Tue, 27 Feb 2024 10:00:42 +0000 (10:00 +0000)]
ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid()
Cast u8 values to u32 when using them to build a 32-bit unsigned value
that is then stored in a u64. This avoids the possibility of a bad sign
extension where the u8 is implicitly extended to an int, thus changing it
from an unsigned to a signed value.
Whether this is a real problem is debatable, but it does no harm to
ensure that the u8 are cast to a suitable type for shifting.
Mark Brown [Mon, 26 Feb 2024 18:30:38 +0000 (18:30 +0000)]
ASoC: meson: axg fixes and clean-up
Merge series from Jerome Brunet <jbrunet@baylibre.com>:
This are various fixes and clean up gathered while working on Amlogic audio
support. These help better handle higher and unusual clock configuration
for TDM, SPDIF or PDM.
Mark Brown [Mon, 26 Feb 2024 17:20:43 +0000 (17:20 +0000)]
ASoC: codecs: Simplify mclk initialization
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
The patchset may not cover all codecs found in the codecs/ directory -
noticed a possible improvement and grepped for similar pattern across C
files found in the directory. Those addressed here seem pretty
straightforward.
Most of clk_xxx() functions do check if provided clk-pointer is
non-NULL. These do not check if the pointer is an error-pointer.
Providing such to a clk_xxx() results in a panic.
By utilizing _optional() variant of devm_clk_get() the driver code is
both simplified and more robust. There is no need to remember about
IS_ERR(clk) checks each time mclk is accessed.
Jerome Brunet [Fri, 23 Feb 2024 17:51:11 +0000 (18:51 +0100)]
ASoC: meson: axg-fifo: take continuous rates
The rate of the stream does not matter for the fifos of the axg family.
Fifos will just push or pull data to/from the DDR according to consumption
or production of the downstream element, which is the DPCM backend.
Drop the rate list and allow continuous rates. The lower and upper rate are
set according what is known to work with the different backends
This allows the PDM input backend to also use continuous rates.
Jerome Brunet [Fri, 23 Feb 2024 17:51:10 +0000 (18:51 +0100)]
ASoC: meson: axg-spdifin: use max width for rate detection
Use maximum width between 2 edges to setup spdifin thresholds
and detect the input sample rate. This comes from Amlogic SDK and
seems to be marginally more reliable than minimum width.
This is done to align with a future eARC support.
No issue was reported with minimum width so far, this is considered
to be an update so no Fixes tag is set.
According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.
The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.
Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.
For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.
Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.
projects / users / jedix / linux-maple.git / log