Takashi Iwai [Wed, 5 Jan 2022 14:39:24 +0000 (15:39 +0100)]
Merge tag 'asoc-v5.17' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.17
Not much going on framework release this time, but a big update for
drivers especially the Intel and SOF ones.
- Refinements and cleanups around the delay() APIs.
- Wider use of dev_err_probe().
- Continuing cleanups and improvements to the SOF code.
- Support for pin switches in simple-card derived cards.
- Support for AMD Renoir ACP, Asahi Kasei Microdevices AKM4375, Intel
systems using NAU8825 and MAX98390, Mediatek MT8915, nVidia Tegra20
S/PDIF, Qualcomm systems using ALC5682I-VS and Texas Instruments
TLV320ADC3xxx.
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.
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.
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.
This series of patches repairs some problems for pcmif BE dai.
The unexpected control flow is corrected, and the missing playback
support of DPCM is added.
Lucas Tanure [Fri, 17 Dec 2021 11:57:02 +0000 (11:57 +0000)]
ASoC: cs35l41: Create shared function for errata patches
ASoC and HDA systems require the same errata patches, so
move it to the shared code using a function the correctly
applies the patches by revision
Also, move CS35L41_DSP1_CCM_CORE_CTRL write to errata
patch function as is required to be written at boot,
but not in regmap_register_patch sequence as will affect
waking up from hibernation
Lucas Tanure [Fri, 17 Dec 2021 11:56:59 +0000 (11:56 +0000)]
ASoC: cs35l41: Convert tables to shared source code
To support CS35L41 in HDA systems the HDA driver
for CS35L41 would have to duplicate some functions
that already exist on ASoC driver
So instead of duplicate the code, use the new lib
source as a shared resource for both ASoC and HDA
Also, change the way CONFIG_SND_SOC_CS35L41 is
selected, as reported by Intel Kernel test robot,
it is possible to build SND_SOC_CS35L41_SPI/I2C
without the main driver, which would lead to build
failures.
Trevor Wu [Thu, 30 Dec 2021 08:47:30 +0000 (16:47 +0800)]
ASoC: mediatek: mt8195: correct pcmif BE dai control flow
Originally, the conditions for preventing reentry are not correct.
dai->component->active is not the state specifically for pcmif dai, so it
is not a correct condition to indicate the status of pcmif dai.
On the other hand, snd_soc_dai_stream_actvie() in prepare ops for both
playback and capture possibly return true at the first entry when these
two streams are opened at the same time.
In the patch, I refer to the implementation in mt8192-dai-pcm.c.
Clock and enabling bit for PCMIF are managed by DAPM, and the condition
for prepare ops is replaced by the status of dai widget.
Derek Fang [Mon, 27 Dec 2021 05:54:46 +0000 (13:54 +0800)]
ASoC: rt5682: Register wclk with its parent_hws instead of parent_data
The mclk might not be registered as a fixed clk name "mclk" on some
platforms.
In those platforms, if the mclk needed to be controlled by codec driver
and acquired by a fixed name, it would be a problem.
This patch to fix the issue that wclk becomes an orphan due to the fixed
mclk's name.
Trevor Wu [Tue, 28 Dec 2021 06:48:21 +0000 (14:48 +0800)]
ASoC: mediatek: mt8195: update control for RT5682 series
Playback pop is observed and the root cause is the reference clock
provided by MT8195 is diabled before RT5682 finishes the control flow.
To ensure the reference clock supplied to RT5682 is disabled after RT5682
finishes all register controls. We replace BCLK with MCLK for RT5682
reference clock, and makes use of set_bias_level_post to handle MCLK
which guarantees MCLK is off after all RT5682 register access.
Jiasheng Jiang [Tue, 28 Dec 2021 03:40:26 +0000 (11:40 +0800)]
ASoC: samsung: idma: Check of ioremap return value
Because of the potential failure of the ioremap(), the buf->area could
be NULL.
Therefore, we need to check it and return -ENOMEM in order to transfer
the error.
Fabio Estevam [Wed, 22 Dec 2021 14:19:19 +0000 (11:19 -0300)]
ASoC: cs4265: Fix part number ID error message
The Chip ID - Register 01h contains the following description
as per the CS4265 datasheet:
"Bits 7 through 4 are the part number ID, which is 1101b (0Dh)"
The current error message is incorrect as it prints CS4265_CHIP_ID,
which is the register number, instead of printing the expected
part number ID value.
To make it clearer, also do a shift by 4, so that the error message
would become:
[ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd
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.
This series contains three topics.
1. SoundWire: Intel: remove pdm support
2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire
3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines
The topics are independent but the changes are dependent. So please
allow me to send them in one series.
ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback.
Configure the speaker gpio pin based on power sequence of the DAPM
speaker events.
Enable speaker after widget power up and Disable before widget powerdown.
ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio
Overloading the tx_mask with a linear value is asking for trouble and
only works because the codec_dai hw_params() is called before the
cpu_dai hw_params().
Move to the more generic set_stream() API to pass the hdac_stream
information.
The HDAudio ASoC support relies on the set_tdm_slots() helper to store
the HDaudio stream tag in the tx_mask. This only works because of the
pre-existing order in soc-pcm.c, where the hw_params() is handled for
codec_dais *before* cpu_dais. When the order is reversed, the
stream_tag is used as a mask in the codec fixup functions:
/* fixup params based on TDM slot masks */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
codec_dai->tx_mask)
soc_pcm_codec_params_fixup(&codec_params,
codec_dai->tx_mask);
As a result of this confusion, the codec_params_fixup() ends-up
generating bad channel masks, depending on what stream_tag was
allocated.
We could add a flag to state that the tx_mask is really not a mask,
but it would be quite ugly to persist in overloading concepts.
Instead, this patch suggests a more generic get/set 'stream' API based
on the existing model for SoundWire. We can expand the concept to
store 'stream' opaque information that is specific to different DAI
types. In the case of HDAudio DAIs, we only need to store a stream tag
as an unsigned char pointer. The TDM rx_ and tx_masks should really
only be used to store masks.
Rename get_sdw_stream/set_sdw_stream callbacks and helpers as
get_stream/set_stream. No functionality change beyond the rename.
This patch provides both a simplification of the suspend flows and a
better balanced operation during suspend/resume transition, as part of
the transition of Sound Open Firmware (SOF) to dynamic pipelines: the
DSP resources are only enabled when required instead of enabled on
startup.
The exiting code relies on a convoluted way of dealing with suspend
signals. Since there is no .suspend DAI callback, we used the
component .suspend and marked all the component DAI dmas as
'suspended'. The information was used in the .prepare stage to
differentiate resume operations from xrun handling, and only
reinitialize SHIM registers and DMA in the former case.
While this solution has been working reliably for about 2 years, there
is a much better solution consisting in trapping the TRIGGER_SUSPEND
in the .trigger DAI ops. The DMA is still marked in the same way for
the .prepare op to run, but in addition the callbacks sent to DSP
firmware are now balanced.
Normal operation:
hw_params -> intel_params_stream
hw_free -> intel_free_stream
This balanced operation was not required with existing SOF firmware
relying on static pipelines instantiated at every boot. With the
on-going transition to dynamic pipelines, it's however a requirement
to keep the use count for the DAI widget balanced across all
transitions.
The component suspend is not removed but instead modified to deal with
a corner case: when a substream is PAUSED, the ALSA core does not
throw the TRIGGER_SUSPEND. This is problematic since the refcount for
all pipelines and widgets is not balanced, leading to issues on
resume. The trigger callback keeps track of the 'paused' state with a
new flag, which is tested during the component suspend called later to
release the remaining DSP resources. These resources will be
re-enabled in the .prepare step.
The IPC used in the TRIGGER_SUSPEND to release DSP resources is not a
problem since the BE dailink is already marked as non-atomic.
ASoC/soundwire: intel: simplify callbacks for params/hw_free
We don't really need to pass a substream to the callback, we only need
the direction. No functionality change, only simplification to enable
improve suspend with paused streams.
Some sound card setups might require extra pin switches to allow
turning off certain audio components. simple-card supports this
already using the "pin-switches" and "widgets" device tree property.
This series makes it possible to use the same properties for the Qcom
sound cards.
To implement that, the function that parses the "pin-switches" property
in simple-card-utils.c is first moved into the ASoC core. Then two
simple function calls are added to the common Qcom sound card DT parser.
Finally there is a small patch for the msm8916-wcd-analog codec to make
it possible to model sound card setups used in some MSM8916 smartphones.
(See PATCH 2/4 for an explanation of some real example use cases.)
Using pin switches rather than patching codec drivers with switches was
originally suggested by Mark Brown on a patch for the tfa989x codec:
https://lore.kernel.org/alsa-devel/[email protected]/
Stephan Gerhold [Tue, 14 Dec 2021 14:20:49 +0000 (15:20 +0100)]
ASoC: msm8916-wcd-analog: Use separate outputs for HPH_L/HPH_R
The analog codec has separate output paths for the left headphone channel
(HPH_L) and the right headphone channel (HPH_R). While they are usually
used together for actual headphones output, some devices also have an
analog speaker amplifier connected to one of the headphone channels.
To allow modelling that properly (and to avoid powering on the unneeded
output path), HPH_L and HPH_R should be represented by separate outputs
rather than a shared HEADPHONE output that always activates both paths.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:48 +0000 (15:20 +0100)]
ASoC: qcom: common: Parse "pin-switches" and "widgets" from DT
Use the DT helpers in the ASoC core to parse the "pin-switches" and
"widgets" properties from the device tree. This allows adding extra
mixers to disable e.g. an extra speaker amplifier that would be
normally powered on automatically because it is connected to a shared
output pin.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:47 +0000 (15:20 +0100)]
ASoC: dt-bindings: qcom: sm8250: Document "pin-switches" and "widgets"
Some sound card setups might require extra pin switches to allow
turning off certain audio components. There are two real examples for
this in smartphones/tablets based on MSM8916:
1. Analog speaker amplifiers connected to headphone outputs.
The MSM8916 analog codec does not have a separate "Line Out" port
so some devices have an analog speaker amplifier connected to one
of the headphone outputs. A pin switch is necessary to allow
playback on headphones without also activating the speaker.
2. External speaker codec also used as earpiece.
Some smartphones have two front-facing (stereo) speakers that can
be also configured to act as an earpiece during voice calls. A pin
switch is needed to allow disabling the second speaker during
voice calls.
There are existing bindings that allow setting up such pin switches in
simple-card.yaml. Document the same for Qcom sound cards.
One variant of example 1 above is added to the examples in the DT
schema: There is an analog speaker amplifier connected to the HPH_R
(right headphone channel) output. Adding a "Speaker" pin switch and
widget allows turning off the speaker when audio should be only played
via the connected headphones.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:46 +0000 (15:20 +0100)]
ASoC: core: Add snd_soc_of_parse_pin_switches() from simple-card-utils
The ASoC core already has several helpers to parse card properties
from the device tree. Move the parsing code for "pin-switches" from
simple-card-utils to a shared snd_soc_of_parse_pin_switches() function
so other drivers can also use it to set up pin switches configured in
the device tree.
this series will improve how we are tracking the firmware's state to be able to
avoid communication with it when it is not going to answer due to a panic and
we will attempt to force power cycle the DSP to recover at the next runtime
suspend time.
The state handling brings in other improvements on the way the kernel reports
errors and DSP panics to reduce the printed lines for normal users, but at the
same time allowing developers (or for bug reports) to have more precise
information available to track down the issue.
We can now place messages easily in the correct debug level and not bound to the
static ERROR for some of the print chains, causing excess amount or partial
information to be printed, confusing users and machines (CI).
I would have prefered to split this series up, but it was developed together to
achieve a single goal to reduce the noise, but also provide the details we need
to be able to rootcause issues.
This is used in meson-gx. Add the property to the binding.
This fixes the dtschema warning:
audio-controller@5400: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
This is used in meson-axg, meson-g12 and meson-gx. Add the property to
the binding.
This fixes the dtschema warning:
audio-codec-0: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:28 +0000 (13:36 +0200)]
ASoC: SOF: Intel: hda: Use DEBUG log level for optional prints
If the user requested to see all dumps (even the optional ones) then use
KERN_DEBUG level for the optional dumps as they are only for debugging
purposes.
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:27 +0000 (13:36 +0200)]
ASoC: SOF: debug: Use DEBUG log level for optional prints
If the user requested to see all dumps (even the optional ones) then use
KERN_DEBUG level for the optional dumps as they are only for debugging
purposes.
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:26 +0000 (13:36 +0200)]
ASoC: SOF: Add clarifying comments for sof_core_debug and DSP dump flags
Update the comment for the global SOF level debug flags and add one for
the flags used to control the DSP dump functionality.
Document the expected behavior when the SOF_DBG_DUMP_OPTIONAL is passed
for the DSP dump:
Only print the dump if SOF_DBG_PRINT_ALL_DUMPS is set
Print must use KERN_DEBUG log level
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:25 +0000 (13:36 +0200)]
ASoC: SOF: Rename snd_sof_get_status() and add kernel log level parameter
The snd_sof_get_status() is not the best name for a function which in fact
is tasked to print out DSP oops and stack. Rename it to
sof_print_oops_and_stack().
At the same time add a new parameter to specify the desired kernel log
level to be used for the prints.
When updating the users of the function, pass KERN_ERR for now to make sure
that there is no functional change happens.
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:21 +0000 (13:36 +0200)]
ASoC: SOF: pm: Force DSP off on suspend in BOOT_FAILED state also
Try to force the DSP to be turned off next time if the fw_state is either
CRASHED or BOOT_FAILED when a suspend happens in order to attempt a clean
boot to recover.
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:19 +0000 (13:36 +0200)]
ASoC: SOF: ipc: Only allow sending of an IPC in SOF_FW_BOOT_COMPLETE state
If the state of the firmware is not BOOT_COMPLETE, it means that the
firmware is not functioning, thus it is not capable of handling IPC
messages.
Do not try to send IPC if the state is not BOOT_COMPLETE
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:13 +0000 (13:36 +0200)]
ASoC: SOF: Add 'non_recoverable' parameter to snd_sof_dsp_panic()
Some platforms use retries during firmware boot to overcome DSP startup
issues.
In these cases we might receive a DSP panic message which should not be
treated as fatal if it happens during boot.
Pass this information to snd_sof_dsp_panic() and omit the panic print if
it is not fatal or the user does not want to see all dumps.
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:09 +0000 (13:36 +0200)]
ASoC: SOF: ops: Use dev_warn() if the panic offsets differ
Catch the cases when the stored sdev->dsp_oops_offset and the offset
received via the panic message differs and print a warning, but keep using
the dsp_oops_offset for the oops query.
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.
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.
This is used in meson-sm1 and meson-g12 .dtsi. Add the property to
the binding.
This fixes the dtschema warning:
audio-controller@740: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
This is used in meson-gxl and meson-g12-common .dtsi. Add the property to
the binding.
This fixes the dtschema warning:
audio-controller@32000: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
Vincent Knecht [Mon, 20 Dec 2021 19:37:25 +0000 (20:37 +0100)]
ASoC: Add AK4375 support
AK4375 is a 32-bit stereo DAC with headphones amplifier.
There's no documentation for it on akm.com, and only a brief
datasheet can be found floating on the internets [1].
Thanks to Oriane BAYERD <[email protected]>
for finally answering my inquiries through akm.com, if only to tell
me that this chip is EOL following AKM factory burning in october 2020
and thus no detailed documentation is available anymore...
AK4331 is advertised [2] as pin and register compatible with AK4375
so some scraps of its datasheet were used and this driver might be
used as a base for it, but this is totally untested.
So this driver is mainly based on downstream code [3] and [4]
by Hu Jin from AKM (no known email).
Tested on msm8916-alcatel-idol347 and msm8939-alcatel-idol3,
which both use PLL driven clock with bypass of SRC (sample rate
converter), so only this setup is supported for now.
Lad Prabhakar [Tue, 21 Dec 2021 17:01:00 +0000 (17:01 +0000)]
ASoC: bcm: Use platform_get_irq() to get the interrupt
platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static
allocation of IRQ resources in DT core code, this causes an issue
when using hierarchical interrupt domains using "interrupts" property
in the node as this bypasses the hierarchical setup and messes up the
irq chaining.
In preparation for removal of static setup of IRQ resource from DT core
code use platform_get_irq().
While at it also drop "r_irq" member from struct bcm_i2s_priv as there
are no users of it.
Lad Prabhakar [Tue, 21 Dec 2021 17:00:59 +0000 (17:00 +0000)]
ASoC: xlnx: Use platform_get_irq() to get the interrupt
platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static
allocation of IRQ resources in DT core code, this causes an issue
when using hierarchical interrupt domains using "interrupts" property
in the node as this bypasses the hierarchical setup and messes up the
irq chaining.
In preparation for removal of static setup of IRQ resource from DT core
code use platform_get_irq().
sound/soc/sof/amd/acp.c:222:9: warning: Identical condition and return
expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/sof/amd/acp.c:213:6: note: If condition 'ret' is true, the
function will return/exit
if (ret)
^
sound/soc/sof/amd/acp.c:222:9: note: Returning identical expression 'ret'
return ret;
^
Dmitry Osipenko [Thu, 16 Dec 2021 16:02:29 +0000 (19:02 +0300)]
ASoC: tegra-audio-rt5677: Correct example
Remove non-existent properties from the example of the binding. These
properties were borrower from the old txt binding, but they were never
used in practice and aren't documented in the new binding. They aren't
reported by the binding checker because dtschema needs extra patch that
hasn't been upstreamed yet to make unevaluatedProperties work properly.
ASoC: amd: acp-config: Update sof_tplg_filename for SOF machines
SOF machines support different codec end points and hence required
different topologies configuration. Update tplg filename in machine
struct to load different topology files for SOF machines.
projects / linux.git / log