Dmitry Osipenko [Wed, 12 Jan 2022 19:50:39 +0000 (22:50 +0300)]
ASoC: hdmi-codec: Fix OOB memory accesses
Correct size of iec_status array by changing it to the size of status
array of the struct snd_aes_iec958. This fixes out-of-bounds slab
read accesses made by memcpy() of the hdmi-codec driver. This problem
is reported by KASAN.
ASoC: qdsp6: q6apm-dai: only stop graphs that are started
Its possible that the sound card is just opened and closed without actually
playing stream, ex: if the audio file itself is missing.
Even in such cases we do call stop on graphs that are not yet started.
DSP can throw errors in such cases, so add a check to see if the graph
was started before stopping it.
ASoC: codecs: wcd938x: fix return value of mixer put function
wcd938x_ear_pa_put_gain, wcd938x_set_swr_port and wcd938x_set_compander
currently returns zero eventhough it changes the value.
Fix this, so that change notifications are sent correctly.
For some reason we ended up with incorrect register offfset calcuations
for sidetone. regmap clearly throw errors when accessing these incorrect
registers as these do not belong to any read/write ranges.
so fix them to point to correct register offsets.
ASoC: codecs: wcd938x: fix incorrect used of portid
Mixer controls have the channel id in mixer->reg, which is not same
as port id. port id should be derived from chan_info array.
So fix this. Without this, its possible that we could corrupt
struct wcd938x_sdw_priv by accessing port_map array out of range
with channel id instead of port id.
Mark Brown [Mon, 24 Jan 2022 15:32:53 +0000 (15:32 +0000)]
ASoC: ops: Reject out of bounds values in snd_soc_put_xr_sx()
We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.
Mark Brown [Mon, 24 Jan 2022 15:32:52 +0000 (15:32 +0000)]
ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx()
We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.
Mark Brown [Mon, 24 Jan 2022 15:32:51 +0000 (15:32 +0000)]
ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()
We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.
ASoC: amd: acp-mach: Fix Left and Right rt1019 amp devices
We're setting wrong card codec conf for rt1019 amp devices in our
machine driver. Due to this left and right amp channels data are
reversed in our machines as wrong device prefix results in wrong
value for "Mono LR Select" rt1019 mixer control. Reverse dev ids
in codec conf with Left and Right name_prefix to fix such issue.
This is because SND_SOC_MT8195_MT6359_RT1011_RT5682
selects SND_SOC_DMIC without selecting or depending on
GPIOLIB, depsite SND_SOC_DMIC depending on GPIOLIB.
This unmet dependency bug was detected by Kismet,
a static analysis tool for Kconfig. Please advise
if this is not the appropriate solution.
Dan Carpenter [Wed, 19 Jan 2022 12:31:01 +0000 (15:31 +0300)]
ASoC: max9759: fix underflow in speaker_gain_control_put()
Check for negative values of "priv->gain" to prevent an out of bounds
access. The concern is that these might come from the user via:
-> snd_ctl_elem_write_user()
-> snd_ctl_elem_write()
-> kctl->put()
Jiasheng Jiang [Tue, 11 Jan 2022 02:50:48 +0000 (10:50 +0800)]
ASoC: cpcap: Check for NULL pointer after calling of_get_child_by_name
If the device does not exist, of_get_child_by_name() will return NULL
pointer.
And devm_snd_soc_register_component() does not check it.
Also, I have noticed that cpcap_codec_driver has not been used yet.
Therefore, it should be better to check it in order to avoid the future
dereference of the NULL pointer.
Robert Hancock [Fri, 7 Jan 2022 21:47:10 +0000 (15:47 -0600)]
ASoC: simple-card: fix probe failure on platform component
A previous change to simple-card resulted in asoc_simple_parse_dai
attempting to retrieve the dai_name for platform components, which are
unlikely to have a valid DAI name. This caused simple-card to fail to
probe when using the xlnx_formatter_pcm as the platform component, since
it does not register any DAI components.
Since the dai_name is not used for platform components, just skip trying
to retrieve it for those.
Robert Hancock [Fri, 7 Jan 2022 21:47:06 +0000 (15:47 -0600)]
ASoC: xilinx: xlnx_formatter_pcm: Make buffer bytes multiple of period bytes
This patch is based on one in the Xilinx kernel tree, "ASoc: xlnx: Make
buffer bytes multiple of period bytes" by Devarsh Thakkar. The same
issue exists in the mainline version of the driver. The original
patch description is as follows:
"The Xilinx Audio Formatter IP has a constraint on period
bytes to be multiple of 64. This leads to driver changing
the period size to suitable frames such that period bytes
are multiple of 64.
Now since period bytes and period size are updated but not
the buffer bytes, this may make the buffer bytes unaligned
and not multiple of period bytes.
When this happens we hear popping noise as while DMA is being
done the buffer bytes are not enough to complete DMA access
for last period of frame within the application buffer boundary.
To avoid this, align buffer bytes too as multiple of 64, and
set another constraint to always enforce number of periods as
integer. Now since, there is already a rule in alsa core
to enforce Buffer size = Number of Periods * Period Size
this automatically aligns buffer bytes as multiple of period
bytes."
Charles Keepax [Fri, 7 Jan 2022 16:06:35 +0000 (16:06 +0000)]
ASoC: cs35l41: Update handling of test key registers
In preparation for the addition of PM runtime support move the test
key out of the register patches themselves. This is necessary to
allow the test key to be held during cache synchronisation, which is
required by the OTP settings which were unpacked from the device and
written by the driver.
Also whilst at it, the driver uses a mixture of accessing the test key
register by name and by address, consistently use the name.
Yassine Oudjana [Tue, 4 Jan 2022 03:35:36 +0000 (03:35 +0000)]
ASoC: wcd9335: Keep a RX port value for each SLIM RX mux
Currently, rx_port_value is a single unsigned int that gets overwritten
when slim_rx_mux_put() is called for any RX mux, then the same value is
read when slim_rx_mux_get() is called for any of them. This results in
slim_rx_mux_get() reporting the last value set by slim_rx_mux_put()
regardless of which SLIM RX mux is in question.
Turn rx_port_value into an array and store a separate value for each
SLIM RX mux.
ASoC: amd: acp: acp-mach: Change default RT1019 amp dev id
RT1019 components was initially registered with i2c1 and i2c2 but
now changed to i2c0 and i2c1 in most of our AMD platforms. Change
default rt1019 components to 10EC1019:00 and 10EC1019:01 which is
aligned with most of AMD machines.
Any exception to rt1019 device ids in near future board design can
be handled using dmi based quirk for that machine.
Shengjiu Wang [Wed, 5 Jan 2022 11:08:03 +0000 (19:08 +0800)]
ASoC: fsl_asrc: refine the check of available clock divider
According to RM, the clock divider range is from 1 to 8, clock
prescaling ratio may be any power of 2 from 1 to 128.
So the supported divider is not all the value between
1 and 1024, just limited value in that range.
Create table for the supported divder and add function to
check the clock divider is available by comparing with
the table.
Hans de Goede [Thu, 6 Jan 2022 11:01:28 +0000 (12:01 +0100)]
ASoC: Intel: bytcr_rt5640: Add support for external GPIO jack-detect
Some boards have the codec IRQ hooked-up as normally, so the driver can
still do things like headset vs headphones and button-press detection,
but instead of using one of the JD pins of the codec, an external GPIO
is used to report the jack-presence switch status of the jack.
Add support for boards which have this setup and which specify which
external GPIO to use in the special Android AMCR0F28 ACPI device.
And add a quirk for the Asus TF103C tablet which uses this setup.
Hans de Goede [Thu, 6 Jan 2022 11:01:27 +0000 (12:01 +0100)]
ASoC: Intel: bytcr_rt5640: Support retrieving the codec IRQ from the AMCR0F28 ACPI dev
Some X86 tablets, which ship with Android as factory installed OS,
specify codec IRQs/GPIOS in a special Android AMCR0F28 ACPI device.
Add support for retrieving the codec IRQ from this ACPI device instead
of from the 10EC5640 device describing the codec itself and enable this
on Asus MemoPad 7 ME176C tablets.
This fixes jack-detect not working on these tablets.
Hans de Goede [Thu, 6 Jan 2022 11:01:26 +0000 (12:01 +0100)]
ASoC: rt5640: Add support for boards with an external jack-detect GPIO
Some boards have the codec IRQ hooked-up as normally, so the driver can
still do things like headset vs headphones and button-press detection,
but instead of using one of the JD pins of the codec, an external GPIO
is used to report the jack-presence switch status of the jack.
Hans de Goede [Thu, 6 Jan 2022 11:01:25 +0000 (12:01 +0100)]
ASoC: rt5640: Allow snd_soc_component_set_jack() to override the codec IRQ
On some boards where the firmware/fwnode information is in essence
read-only (x86 + ACPI boards) the i2c_client for the codec may contain
the wrong IRQ or no IRQ at all.
Since we only request the IRQ once snd_soc_component_set_jack() gets
called, allow machine drivers to override the IRQ with the proper one
through the data parameter to snd_soc_component_set_jack().
Hans de Goede [Thu, 6 Jan 2022 11:01:24 +0000 (12:01 +0100)]
ASoC: rt5640: Change jack_work to a delayed_work
Change jack_work from a struct work_struct to a struct delayed_work, this
is a preparation patch for adding support for boards where an external
GPIO is used for jack-detect, rather then one of the JD pins of the codec.
Hans de Goede [Thu, 6 Jan 2022 11:01:23 +0000 (12:01 +0100)]
ASoC: rt5640: Fix possible NULL pointer deref on resume
Commit 2b9c8d2b3c89 ("ASoC: rt5640: Add the HDA header support") adds
re-queuing of the jack_work on resume when rt5640->jd_src != 0.
But the jack_work will unconditionally deref rt5640->jack and that might
be NULL. E.g. the sound/soc/intel/boards/bytcr_rt5640.c machine driver
call snd_soc_component_set_jack(codec, NULL, NULL) from pre_suspend to
disable the IRQ to avoid spurious wakeups, so when rt5640_resume()
runs rt5640->jack will be NULL in this case.
Make the queueing of the work conditional on rt5640->jack instead of
on rt5640->jd_src to fix this.
Shengjiu Wang [Tue, 4 Jan 2022 10:40:35 +0000 (18:40 +0800)]
ASoC: imx-card: improve the sound quality for low rate
According to RM, on auto mode:
For codec AK4458 and AK4497, the lowest ratio of MLCK/FS is 256
if sample rate is 8kHz-48kHz,
For codec AK5558, the lowest ratio of MLCK/FS is 512 if sample
rate is 8kHz-48kHz.
With these setting the sound quality for 8kHz-48kHz can be improved.
Shengjiu Wang [Tue, 4 Jan 2022 10:40:34 +0000 (18:40 +0800)]
ASoC: imx-card: Fix mclk calculation issue for akcodec
Transfer the refined slots and slot_width to akcodec_get_mclk_rate()
for mclk calculation, otherwise the mclk frequency does not match
with the slots and slot_width for S16_LE format, because the default
slot_width is 32.
Shengjiu Wang [Tue, 4 Jan 2022 10:40:33 +0000 (18:40 +0800)]
ASoC: imx-card: Need special setting for ak4497 on i.MX8MQ
The SAI on i.MX8MQ don't support one2one ratio for mclk:bclk, so
the mclk frequency exceeds the supported range of codec for
the case that sample rate is larger than 705kHZ and format is
S32_LE. Update the supported width for such case.
Takashi Iwai [Wed, 5 Jan 2022 16:24:09 +0000 (17:24 +0100)]
ASoC: ak4375: Fix unused function error
A randconfig caught a compile warning that is now treated as a fatal
error:
sound/soc/codecs/ak4375.c:415:13: error: ‘ak4375_power_off’ defined but not used [-Werror=unused-function]
where ak4375_power_off() is used only from the PM handler.
As both suspend and resumes are already marked with __maybe_unused,
let's rip off the superfluous ifdef CONFIG_PM, so that the error above
can be avoided.
This patch series adds support for the low power hibernation feature
on cs35l41. This allows the DSP memory to be retained whilst the
device enters a very low power state.
Charles Keepax [Wed, 5 Jan 2022 11:30:24 +0000 (11:30 +0000)]
ASoC: wm_adsp: Add support for "toggle" preloaders
In the case a device can support retaining the firmware memory across
low power states it is useful for the preloader widget to only power up
whilst actually loading/unloading the core, as opposed to the normal
operation where the widget is powered for the entire time a firmware is
preloaded onto the core. Add support for this mode and a flag to enable
it.
Charles Keepax [Wed, 5 Jan 2022 11:30:23 +0000 (11:30 +0000)]
firmware: cs_dsp: Clear core reset for cache
If the Halo registers are kept in the register cache the
HALO_CORE_RESET bit will be retained as 1 after reset is triggered in
cs_dsp_halo_start_core. This will cause subsequent writes to reset
the core which is not desired. Apart from this bit the rest of the
register bits are cacheable, so for safety sake clear the bit to
ensure the cache is consistent.
Charles Keepax [Wed, 5 Jan 2022 11:30:22 +0000 (11:30 +0000)]
ASoC: cs35l41: Correct handling of some registers in the cache
It makes no sense to cache the test/user key registers, since they
require values written at specific times, mark them volatile. It is
probably best if they can't be accessed from user-space either, so
mark them precious as well.
The interrupt force, edge, polarity and debounce are all settings
applied to the IRQ rather than status bits and as such should not be
volatile.
The OTP trim values will require re-application in the event of a
cache sync and as such should not be volatile. The OTPID however
should be volatile.
The DSP scratch registers are used to read back an error/debug code
from the DSP on shutdown, as such these should be marked volatile.
Finally, add some missing defaults, add TST_FS_MON0, and allow the
DSP core control register to be cached.
Charles Keepax [Wed, 5 Jan 2022 11:30:21 +0000 (11:30 +0000)]
ASoC: cs35l41: Correct DSP power down
The wm_adsp_event should be called before the early_event on power
down, event stops the core running and early_event then powers down
the core. Additionally, the core should only be stopped if it was
actually running in the first place.
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
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.
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