ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts
The headset type detection must run to set the analogue switches
correctly for the attached headset type. Without this only headsets
with wiring matching the chip default will have a functioning mic.
commit c26a5289e865 ("ASoC: cs42l42: Add support for set_jack calls")
moved the interrupt unmasking to the component set_jack() callback.
But it's not mandatory for a machine driver to register a struct
snd_soc_jack handler. Without a registered handler the type detection
would not have run and so the mic would not work on some types of
headset.
This patch restores the unmasking of TS_PLUG and TS_UNPLUG interrupts
during probe.
ASoC: cs42l42: Fix WARN in remove() if running without an interrupt
The driver must free the IRQ in remove() to prevent the potential race
where an IRQ starts to be handled while the driver is being removed but
devres has not yet called free_irq(). However, the driver can run without
an interrupt but devm_free_irq() will hit a WARN() if no devres-managed
interrupt was ever created.
Fix this by only attempting to create the interrupt handler if the hardware
config specified an interrupt, and failing probe() if the interrupt could
not be created. This means that in cs42l42_remove() an interrupt must have
been registered if the irq number is valid and therefore it is safe to call
devm_free_irq().
According to the datasheet the SRC MCLK must be as near as possible to
(125 * sample rate). This means it should be ~6MHz for rates up to 48k
and ~12MHz for rates above that. As per datasheet table 4-21.
ASoC: cs42l42: Allow time for HP/ADC to power-up after enable
After enabling the HP or ADC by writing the corresponding PDN=0,
it takes around 20 milliseconds for it to power up and the midrail
supply to be stable. Add this wait into a DAPM widget callback.
If HP and ADC are both powering up in a DAPM sequence, there's no
need to do the wait twice. The widget will perform one wait in the
POST_PMU if there was a PRE_PMU for one or both.
It isn't possible to switch MCLK between 12MHz and 24MHz rate groups
on-the-fly - this can only be done when cs42l42 is powered-down.
All "normal" SCLK rates use an MCLK in the 12MHz group, so change the
configs for SCLK > 12.288 MHz to use the PLL to generate an MCLK in
the 12MHz group.
As this means MCLK_DIV is always 0 it can be removed from the pll
configuration setup.
The driver currently only supports configuring for sample rates <= 96k
and it isn't possible to setup a configuration that will support all
sample rates up to 192k.
For sample rates up to 96k MCLK is in the 12MHz group.
However, although 192k only requires an I2S clock in the 12MHz group,
the cs42l42 audio path is not natively 192k so the audio must be
resampled. But for 192k the SRC requires a 24MHz MCLK.
It is not possible to switch MCLK between 12MHz and 24MHz groups
on-the-fly. The 12MHz group supports all sample rates up to 96k.
ASoC: cs42l42: Defer probe if request_threaded_irq() returns EPROBE_DEFER
The driver can run without an interrupt so if devm_request_threaded_irq()
failed, the probe() just carried on. But if this was EPROBE_DEFER the
driver would continue without an interrupt instead of deferring to wait
for the interrupt to become available.
ASoC: cs42l42: Always configure both ASP TX channels
An I2S frame always has two slots (left and right) even when sending
mono. The right channel (channel 2) of ASP TX will always have the
same bit width as the left channel and will always be on the high
phase of LRCLK.
The previous implementation always passed the field masks for both
channels to snd_soc_component_update_bits() but for mono the written value
only contained the settings for channel 1. The result was that for mono
channel 2 was set to 8-bit (which is an invalid configuration) with both
channels on the low phase of LRCLK.
ASoC: cs42l42: Don't reconfigure the PLL while it is running
When capture and playback substreams are both running at the same time,
cs42l42_pcm_hw_params() would be called for each direction. The first
call will configure the PLL. The second call must not write the PLL
configuration registers again if the first substream is already running,
as this could destabilize the PLL.
The DAI is marked symmetric sample bits and sample rate, so the two
directions will always have the same SCLK (I2S always has 2 channel slots
so the DAI does not need to require symmetric channels to guarantee the
same SCLK). However, since cs42l42_pll_config() is checking for an active
stream it may as well test that the requested SCLK is the same as the
currently active configuration.
Shengjiu Wang [Wed, 13 Oct 2021 05:17:04 +0000 (13:17 +0800)]
ASoC: wm8960: Fix clock configuration on slave mode
There is a noise issue for 8kHz sample rate on slave mode.
Compared with master mode, the difference is the DACDIV
setting, after correcting the DACDIV, the noise is gone.
There is no noise issue for 48kHz sample rate, because
the default value of DACDIV is correct for 48kHz.
So wm8960_configure_clocking() should be functional for
ADC and DAC function even if it is slave mode.
In order to be compatible for old use case, just add
condition for checking that sysclk is zero with
slave mode.
Stefan Binding [Mon, 11 Oct 2021 14:49:03 +0000 (15:49 +0100)]
ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets
Ensure the default 0dB playback path is always used.
The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was
spurious, and resulted in a -6dB attenuation being accidentally
inserted into the playback path.
Yang Yingliang [Sat, 9 Oct 2021 06:58:40 +0000 (14:58 +0800)]
ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
'component' is allocated in snd_soc_register_component(), but component->list
is not initalized, this may cause snd_soc_del_component_unlocked() deref null
ptr in the error handing case.
Takashi Iwai [Wed, 6 Oct 2021 14:17:12 +0000 (16:17 +0200)]
ASoC: DAPM: Fix missing kctl change notifications
The put callback of a kcontrol is supposed to return 1 when the value
is changed, and this will be notified to user-space. However, some
DAPM kcontrols always return 0 (except for errors), hence the
user-space misses the update of a control value.
This patch corrects the behavior by properly returning 1 when the
value gets updated.
Hans de Goede [Wed, 29 Sep 2021 20:15:12 +0000 (22:15 +0200)]
ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working
Commit 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling")
replaced the nau8824_dapm_enable_pin() helper with direct calls to
snd_soc_dapm_enable_pin(), but the helper was using
snd_soc_dapm_force_enable_pin() and not forcing the MICBIAS + SAR
supplies on breaks headphone vs headset and button-press detection.
Replace the snd_soc_dapm_enable_pin() calls with
snd_soc_dapm_force_enable_pin() to fix this.
Mark Brown [Fri, 24 Sep 2021 19:48:44 +0000 (20:48 +0100)]
ASoC: cs4341: Add SPI device ID table
Currently autoloading for SPI devices does not use the DT ID table, it uses
SPI modalises. Supporting OF modalises is going to be difficult if not
impractical, an attempt was made but has been reverted, so ensure that
module autoloading works for this driver by adding SPI IDs for parts that
only have a compatible listed.
Mark Brown [Fri, 24 Sep 2021 19:49:56 +0000 (20:49 +0100)]
ASoC: pcm179x: Add missing entries SPI to device ID table
Currently autoloading for SPI devices does not use the DT ID table, it uses
SPI modalises. Supporting OF modalises is going to be difficult if not
impractical, an attempt was made but has been reverted, so ensure that
module autoloading works for this driver by adding SPI IDs for parts that
only have a compatible listed.
Shengjiu Wang [Fri, 10 Sep 2021 09:18:30 +0000 (17:18 +0800)]
ASoC: fsl_xcvr: Fix channel swap issue with ARC
With pause and resume test for ARC, there is occasionally
channel swap issue. The reason is that currently driver set
the DPATH out of reset first, then start the DMA, the first
data got from FIFO may not be the Left channel.
Moving DPATH out of reset operation after the dma enablement
to fix this issue.
Peter Rosin [Mon, 20 Sep 2021 14:49:39 +0000 (16:49 +0200)]
ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers
Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats")
breaks the TSE-850 device, which is using a pcm5142 in I2S and
CBM_CFS mode (maybe not relevant). Without this fix, the result
is:
pcm512x 0-004c: Failed to set data format: -16
And after that, no sound.
This fix is not 100% correct. The datasheet of at least the pcm5142
states that four bits (0xcc) in the I2S_1 register are "RSV"
("Reserved. Do not access.") and no hint is given as to what the
initial values are supposed to be. So, specifying defaults for
these bits is wrong. But perhaps better than a broken driver?
Peter Ujfalusi [Fri, 17 Sep 2021 08:51:08 +0000 (11:51 +0300)]
ASoC: SOF: trace: Omit error print when waking up trace sleepers
Do not print error message from snd_sof_trace_notify_for_error() when
possible sleeping trace work is woken up to flush the remaining debug
information.
This action by itself is not an error, it is just an action we take when
an error occurs to make sure that all information have been fed to the
userspace (if we have trace in use).
Peter Ujfalusi [Thu, 16 Sep 2021 08:53:42 +0000 (11:53 +0300)]
ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication
In case the firmware is missing we will have the following in the kernel
log:
1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2
2 | error: request firmware intel/sof/sof-tgl-h.ri failed err: -2
3 | you may need to download the firmware from https://github.com/thesofproject/sof-bin/
4 | error: failed to load DSP firmware -2
5 | error: sof_probe_work failed err: -2
The first line is the standard, request_firmware() warning.
The second and third line is printed in snd_sof_load_firmware_raw()
Note that the first and second line is mostly identical.
With this patch the log will be changed to:
1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2
2 | error: sof firmware file is missing, you might need to
3 | download it from https://github.com/thesofproject/sof-bin/
4 | error: failed to load DSP firmware -2
5 | error: sof_probe_work failed err: -2
Marc Herbert [Thu, 16 Sep 2021 08:50:08 +0000 (11:50 +0300)]
ASoC: SOF: loader: release_firmware() on load failure to avoid batching
Invoke release_firmware() when the firmware fails to boot in
sof_probe_continue().
The request_firmware() framework must be informed of failures in
sof_probe_continue() otherwise its internal "batching"
feature (different from caching) cached the firmware image
forever. Attempts to correct the file in /lib/firmware/ were then
silently and confusingly ignored until the next reboot. Unloading the
drivers did not help because from their disconnected perspective the
firmware had failed so there was nothing to release.
Also leverage the new snd_sof_fw_unload() function to simplify the
snd_sof_device_remove() function.
Mark Brown [Fri, 10 Sep 2021 14:46:13 +0000 (15:46 +0100)]
Merge series "ASoC: fsl: register platform component before registering cpu dai" from Shengjiu Wang <[email protected]>:
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
This patch set is to fix this issue for SAI, ESAI, MICFIL, SPDIF,
XCVR drivers.
Shengjiu Wang (5):
ASoC: fsl_sai: register platform component before registering cpu dai
ASoC: fsl_esai: register platform component before registering cpu dai
ASoC: fsl_micfil: register platform component before registering cpu
dai
ASoC: fsl_spdif: register platform component before registering cpu
dai
ASoC: fsl_xcvr: register platform component before registering cpu dai
ASoC: mediatek: common: handle NULL case in suspend/resume function
When memory allocation for afe->reg_back_up fails, reg_back_up can't
be used.
Keep the suspend/resume flow but skip register backup when
afe->reg_back_up is NULL, in case illegal memory access happens.
Shengjiu Wang [Fri, 3 Sep 2021 10:30:06 +0000 (18:30 +0800)]
ASoC: fsl_xcvr: register platform component before registering cpu dai
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
Shengjiu Wang [Fri, 3 Sep 2021 10:30:05 +0000 (18:30 +0800)]
ASoC: fsl_spdif: register platform component before registering cpu dai
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
Shengjiu Wang [Fri, 3 Sep 2021 10:30:04 +0000 (18:30 +0800)]
ASoC: fsl_micfil: register platform component before registering cpu dai
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
Shengjiu Wang [Fri, 3 Sep 2021 10:30:03 +0000 (18:30 +0800)]
ASoC: fsl_esai: register platform component before registering cpu dai
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
Shengjiu Wang [Fri, 3 Sep 2021 10:30:02 +0000 (18:30 +0800)]
ASoC: fsl_sai: register platform component before registering cpu dai
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
MAINTAINERS: fix update references to stm32 audio bindings
The 00d38fd8d2524 ("MAINTAINERS: update references to stm32 audio bindings")
commit update the bindings reference, by
removing bindings/sound/st,stm32-adfsdm.txt, to set the
new reference to bindings/iio/adc/st,stm32-*.yaml.
This leads to "get_maintainer finds" the match for the
Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml,
but also to the IIO bindings
Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml
And The commit fixes only a part of the problem:
Documentation/devicetree/bindings/sound/st,stm32-*.txt file have been
also moved to yaml.
Update references to include all stm32 audio bindings file and
exclude the st,stm32-adc.yaml bindings file.
Because SND_SOC_MT8195 depends on COMPILE_TEST, it's possible to build
MT8195 driver in many different config combinations.
Add all dependent config for SND_SOC_MT8195 in case some errors happen
when COMPILE_TEST is enabled.
ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic
The SoundWire BEs make use of 'stream' functions for .prepare and
.trigger. These functions will in turn force a Bank Switch, which
implies a wait operation.
Mark SoundWire BEs as nonatomic for consistency, but keep all other
types of BEs as is. The initialization of .nonatomic is done outside
of the create_sdw_dailink helper to avoid adding more parameters to
deal with a single exception to the rule that BEs are atomic.
ASoC: mt8195: correct the dts parsing logic about DPTX and HDMITX
According to the description in dt-bindings, phandle assignment of
HDMI TX and DP TX are not required properties, but driver regards them
as required properties.
In real use case, it's expected that DP TX and HDMI TX are optional
features, so correct the behavior in driver.
When CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH is enabled without
CONFIG_EXPERT, there is a Kconfig warning about unmet dependencies:
WARNING: unmet direct dependencies detected for SND_SOC_SDW_MOCKUP
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] &&
EXPERT [=n] && SOUNDWIRE [=y]
Selected by [y]:
- SND_SOC_INTEL_SOUNDWIRE_SOF_MACH [=y] && ...
Selecting a symbol does not account for dependencies. There are three
ways to resolve this:
1. Make CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select
CONFIG_SND_SOC_SDW_MOCKUP only if CONFIG_EXPERT is set.
2. Make CONFIG_SND_SOC_SDW_MOCKUP's prompt depend on CONFIG_EXPERT so
that it can be selected by options that only depend on
CONFIG_SOUNDWIRE but still appear as a prompt to the user when
CONFIG_EXPERT is set.
3. Make CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH imply
CONFIG_SND_SOC_SDW_MOCKUP, which will select
CONFIG_SND_SOC_SDW_MOCKUP when its dependencies are enabled but still
allow the user to disable it.
Go with the third option as it gives the most flexibility while
retaining the original intent of the select.
Mark Brown [Wed, 1 Sep 2021 16:30:37 +0000 (17:30 +0100)]
Merge tag 'asoc-v5.15' into asoc-5.15
ASoC: Updates for v5.15
Quite a quiet release this time, mostly a combination of cleanups
and a good set of new drivers.
- Lots of cleanups and improvements to the Intel drivers,
including some new systems support.
- New support for AMD Vangoh, CUI CMM-4030D-261, Mediatek
Mt8195, Renesas RZ/G2L Mediatek Mt8195, RealTek RT101P,
Renesas RZ/G2L,, Rockchip RK3568 S/PDIF.
ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK
The Mediatek MT8195 sound hardware is only present on Mediatek MT8195
SoCs. Hence add a dependency on ARCH_MEDIATEK, to prevent asking the
user about this driver when configuring a kernel without Mediatek SoC
support.
commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support")
removed Platform support from audio-graph, because it doesn't have
"plat" support on DT (simple-card has).
But, Platform support is needed if user is using
snd_dmaengine_pcm_register() which adds generic DMA as Platform.
And this Platform dev is using CPU dev.
Without this patch, at least STM32MP15 audio sound card is no more
functional (v5.13 or later). This patch respawn Platform Support on
audio-graph again.
Mark Brown [Thu, 26 Aug 2021 18:15:14 +0000 (19:15 +0100)]
Merge series "ASoC: wcd9335: Firx some resources leak in the probe and remove function" from Christophe JAILLET <[email protected]>:
The first 2 patches are sraightforward and look logical to me.
However, the 3rd one in purely speculative. It is based on the fact that a
comment states that we enable some irqs on some slave ports. That said, it writes
0xFF in some registers.
So, I guess that we should disable these irqs when the driver is removed. That
said, writing 0x00 at the same place looks logical to me.
This cis untested and NOT based on any documentation. Just a blind fix.
Review with care.
You'll be warned :)
Christophe JAILLET (3):
ASoC: wcd9335: Fix a double irq free in the remove function
ASoC: wcd9335: Fix a memory leak in the error handling path of the
probe function
ASoC: wcd9335: Disable irq on slave ports in the remove function
On start/pause_release/resume, when more than one FE is connected to
the same BE, it's possible that the trigger is sent more than
once. This is not desirable, we only want to trigger a BE once, which
is straightforward to implement with a refcount.
For stop/pause/suspend, the problem is more complicated: the check
implemented in snd_soc_dpcm_can_be_free_stop() may fail due to a
conceptual deadlock when we trigger the BE before the FE. In this
case, the FE states have not yet changed, so there are corner cases
where the TRIGGER_STOP is never sent - the dual case of start where
multiple triggers might be sent.
This patch suggests an unconditional trigger in all cases, without
checking the FE states, using a refcount protected by a spinlock.
ASoC: soc-pcm: protect BE dailink state changes in trigger
When more than one FE is connected to a BE, e.g. in a mixing use case,
the BE can be triggered multiple times when the FE are opened/started
concurrently. This race condition is problematic in the case of
SoundWire BE dailinks, and this is not desirable in a general
case. The code carefully checks when the BE can be stopped or
hw_free'ed, but the trigger code does not use any mutual exclusion.
Fix by using the same spinlock already used to check FE states, and
set the state before the trigger. In case of errors, the initial
state will be restored.
This patch does not change how the triggers are handled, it only makes
sure the states are handled in critical sections.
ASoC: wcd9335: Fix a double irq free in the remove function
There is no point in calling 'free_irq()' explicitly for
'WCD9335_IRQ_SLIMBUS' in the remove function.
The irqs are requested in 'wcd9335_setup_irqs()' using a resource managed
function (i.e. 'devm_request_threaded_irq()').
'wcd9335_setup_irqs()' requests all what is defined in the 'wcd9335_irqs'
structure.
This structure has only one entry for 'WCD9335_IRQ_SLIMBUS'.
So 'devm_request...irq()' + explicit 'free_irq()' would lead to a double
free.
Remove the unneeded 'free_irq()' from the remove function.
Mark Brown [Thu, 26 Aug 2021 14:08:30 +0000 (15:08 +0100)]
Merge series "Patches to update for rockchip i2s" from Sugar Zhang <[email protected]>:
These patches fixup or update for rockchip i2s.
Changes in v3:
- Drop property 'rockchip,playback-only', 'rockchip,capture-only'.
Implement it by 'dma-names' of DT instead.
Changes in v2:
- split property trcm into single 'trcm-sync-tx-only' and
'trcm-sync-rx-only' suggested by Nicolas.
- split property trcm into single 'trcm-sync-tx-only' and
'trcm-sync-rx-only' suggested by Nicolas.
- drop change-id
Sugar Zhang (12):
ASoC: rockchip: i2s: Add support for set bclk ratio
ASoC: rockchip: i2s: Fixup clk div error
ASoC: rockchip: i2s: Improve dma data transfer efficiency
ASoC: rockchip: i2s: Fix regmap_ops hang
ASoC: rockchip: i2s: Fix concurrency between tx/rx
ASoC: rockchip: i2s: Reset the controller if soft reset failed
ASoC: dt-bindings: rockchip: Document reset property for i2s
ASoC: rockchip: i2s: Make playback/capture optional
ASoC: rockchip: i2s: Add compatible for more SoCs
ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs
ASoC: rockchip: i2s: Add support for frame inversion
ASoC: dt-bindings: rockchip: i2s: Document property TRCM
Xiaotan Luo (1):
ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
Xing Zheng (1):
ASoC: rockchip: i2s: Add support for TRCM property
Robin Murphy [Wed, 25 Aug 2021 15:42:03 +0000 (16:42 +0100)]
ASoC: dwc: Get IRQ optionally
The IRQ is explicitly optional, so use platform_get_irq_optional() and
avoid platform_get_irq() logging a spurious error when trying to use the
thing in DMA mode.
Peter Ujfalusi [Wed, 25 Aug 2021 12:25:19 +0000 (15:25 +0300)]
ASoC: Intel: bytcr_rt5640: Make rt5640_jack_gpio/rt5640_jack2_gpio static
Marking the two jack gpio as static fixes the following Sparse errors:
sound/soc/intel/boards/bytcr_rt5640.c:468:26: error: symbol 'rt5640_jack_gpio' was not declared. Should it be static?
sound/soc/intel/boards/bytcr_rt5640.c:475:26: error: symbol 'rt5640_jack2_gpio' was not declared. Should it be static?
Mark Brown [Wed, 25 Aug 2021 09:50:21 +0000 (10:50 +0100)]
Merge series "ASoC: mediatek: Add support for MT8195 SoC" from Trevor Wu <[email protected]>:
This series of patches adds support for Mediatek AFE of MT8195 SoC.
Patches are based on broonie tree "for-next" branch.
Changes since v4:
- removed sof related code
Changes since v3:
- fixed warnings found by kernel test robot
- removed unused critical section
- corrected the lock protected sections on etdm driver
- added DPTX and HDMITX audio support
Changes since v2:
- added audio clock gate control
- added 'mediatek' prefix to private dts properties
- added consumed clocks to dt-bindins and adopted suggestions from Rob
- refined clock usage and remove unused clock and control code
- fixed typos
Changes since v1:
- fixed some problems related to dt-bindings
- added some missing properties to dt-bindings
- added depency declaration on dt-bindings
- fixed some warnings found by kernel test robot
Trevor Wu (11):
ASoC: mediatek: mt8195: update mediatek common driver
ASoC: mediatek: mt8195: support audsys clock control
ASoC: mediatek: mt8195: support etdm in platform driver
ASoC: mediatek: mt8195: support adda in platform driver
ASoC: mediatek: mt8195: support pcm in platform driver
ASoC: mediatek: mt8195: add platform driver
dt-bindings: mediatek: mt8195: add audio afe document
ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and
rt5682
ASoC: mediatek: mt8195: add DPTX audio support
ASoC: mediatek: mt8195: add HDMITX audio support
dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682
document
Trevor Wu [Thu, 19 Aug 2021 08:41:37 +0000 (16:41 +0800)]
ASoC: mediatek: mt8195: support adda in platform driver
This patch adds mt8195 adda dai driver.
audio_h clock is used by ADSP bus and ADDA module.
When ADDA requires audio_h clock, it is switched to APLL1, otherwise
it is switched to Xtal_26m so that APLL1 can be turned off when audio
feature is not used.
ADSP bus only requires that the clock is on, so dynamic reparenting
is used for the purpose of lowering power consumption.
Trevor Wu [Thu, 19 Aug 2021 08:41:36 +0000 (16:41 +0800)]
ASoC: mediatek: mt8195: support etdm in platform driver
This patch adds mt8195 tdm/i2s dai driver.
MCLK clock tree is as follows.
PLL -> MUX -> DIVIDER -> MCLK
For PLL source of MCLK, driver only supports APLL1 and APLL2 now.
APLL3 and APLL4 are used to track external clock source, so they are
only used when slave input is connected.
For example,
case 1: (HDMI RX connected)
DL memif (a1sys) -> etdm out2 (clk from apll1/apll2) -> codec
case 2: (HDMI RX disconnected)
HDMI RX -> a3sys -> UL memif (a3sys) -> DL memif (a3sys) -> .... ->
etdm out2 (clk from apll3) -> codec
We keep all modules in the pipeline working on the same clock domain.
MCLK is expected to output the clock generated from the same clock
source as the pipeline, so dynamic reparenting is required for MCLK
configuration.
As a result, clk_set_parent() is used to select PLL source,
and clk_set_rate() is used to configure divider to get MCLK output rate.
Charles Keepax [Tue, 24 Aug 2021 10:15:52 +0000 (11:15 +0100)]
ASoC: wm_adsp: Put debugfs_remove_recursive back in
This patch reverts commit acbf58e53041 ("ASoC: wm_adsp: Let
soc_cleanup_component_debugfs remove debugfs"), and adds an
alternate solution to the issue. That patch removes the call to
debugfs_remove_recursive, which cleans up the DSPs debugfs. The
intention was to avoid an unbinding issue on an out of tree
driver/platform.
The issue with the patch is it means the driver no longer cleans up
its own debugfs, instead relying on ASoC to remove recurive on the
parent debugfs node. This is conceptually rather unclean, but also it
would prevent DSPs being added/removed independently of ASoC and soon
we are going to be upstreaming some non-audio parts with these DSPs,
which will require this.
Finally, it seems the issue on the platform is a result of the
wm_adsp2_cleanup_debugfs getting called twice. This is very likely a
problem on the platform side and will be resolved there. But in the mean
time make the code a little more robust to such issues, and again
conceptually a bit nicer, but clearing the debugfs_root variable in the
DSP structure when the debugfs is removed.
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