ASoC is now supporting multi DAI, but, current
simple-card / audio-graph are assuming fixed single DAI.
Now, asoc_simple_parse_xxx() macro is assuming single DAI.
To support multi-CPU/Codec, this patch unpack asoc_simple_parse_xxx()
macro, and uses "&dai_link->cpus[i]" instead of "dai_link->cpus".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pmz0wf9u.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card / audio-graph creates 1xCPU + 1xCodec + 1xPlatform
for all dai_link, but some of them is not needed.
For example Platform is not needed for DPCM BE case.
Moreover, we can share snd-soc-dummy DAI for CPU-dummy / dummy-Codec
in DPCM.
This patch adds dummy DAI and share it when DPCM case,
I beliave it can contribute to reduce memory.
By this patch, CPU-dummy / dummy-CPU are set at asoc_simple_init_priv(),
thus, its settings are no longer needed at DPCM detecting timing
on simple-card / audio-graph.
Moreover, we can remove triky Platform settings code for DPCM BE,
because un-needed Platform is not created.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tuoqod22.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA control interface allows users to add arbitrary control elements
(called "user controls" or "user elements"), and its resource usage is
limited just by the max number of control sets (currently 32). This
limit, however, is quite loose: each allocation of control set may
have 1028 elements, and each element may have up to 512 bytes (ILP32) or
1024 bytes (LP64) of value data. Moreover, each control set may contain
the enum strings and TLV data, which can be up to 64kB and 128kB,
respectively. Totally, the whole memory consumption may go over 38MB --
it's quite large, and we'd rather like to reduce the size.
OTOH, there have been other requests even to increase the max number
of user elements; e.g. ALSA firewire stack require the more user
controls, hence we want to raise the bar, too.
For satisfying both requirements, this patch changes the management of
user controls: instead of setting the upper limit of the number of
user controls, we check the actual memory allocation size and set the
upper limit of the total allocation in bytes. As long as the memory
consumption stays below the limit, more user controls are allowed than
the current limit 32. At the same time, we set the lower limit (8MB)
as default than the current theoretical limit, in order to lower the
risk of DoS.
As a compromise for lowering the default limit, now the actual memory
limit is defined as a module option, 'max_user_ctl_alloc_size', so that
user can increase/decrease the limit if really needed, too.
Link: https://lore.kernel.org/r/s5htur3zl5e.wl-tiwai@suse.de
Co-developed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210408103149.40357-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: control - add generic LED API
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
simple-card / audio-graph are assuming single CPU/Codec/Platform on
dai_link. Because of it, it is difficult to support Multi-CPU/Codec.
This patch allocs CPU/Codec/Platform dai_link imformation
instead of using existing props information. It can update to
multi-CPU/Codec, but is still assuming single-CPU/Codec for now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blb61tpv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: control - add generic LED API
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent laptops have usually two LEDs assigned to reflect
the speaker and microphone mute state. This implementation
adds a tiny layer on top of the control API which calculates
the state for those LEDs using the driver callbacks.
Two new access flags are introduced to describe the controls
which affects the audio path settings (an easy code change
for drivers).
The LED resource can be shared with multiple sound cards with
this code. The user space controls may be added to the state
chain on demand, too.
This code should replace the LED code in the HDA driver and
add a possibility to easy extend the other drivers (ASoC
codecs etc.).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210317172945.842280-4-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core,
Linux is running on A core, RTOS is running on M core.
The audio hardware device can be controlled by Cortex-M device,
So audio playback/capture can be handled by M core.
Rpmsg is the interface for sending and receiving msg to and from M
core, that we can create a virtual sound on Cortex-A core side.
A core will tell the Cortex-M core sound format/rate/channel,
where is the data buffer, what is the period size, when to start,
when to stop and when suspend or resume happen, each of this behavior
there is defined rpmsg command.
Especially we designed the low power audio case, that is to
allocate a large buffer and fill the data, then Cortex-A core can go
to sleep mode, Cortex-M core continue to play the sound, when the
buffer is consumed, Cortex-M core will trigger the Cortex-A core to
wakeup to fill data.
changes in v5:
- remove unneeded property in binding doc and driver
- update binding doc according to Rob's comments.
- Fix link issue reported by kernel test robot
changes in v4:
- remove the sound card node, merge the property to cpu dai node
according to Rob's comments.
- sound card device will be registered by cpu dai driver.
- Fix do_div issue reported by kernel test robot
changes in v3:
- add local refcount for clk enablement in hw_params()
- update the document according Rob's comments
changes in v2:
- update codes and comments according to Mark's comments
Shengjiu Wang (6):
ASoC: soc-component: Add snd_soc_pcm_component_ack
ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg
ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg audio device
ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel
ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg
ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg
.../devicetree/bindings/sound/fsl,rpmsg.yaml | 108 +++
include/sound/soc-component.h | 3 +
sound/soc/fsl/Kconfig | 30 +
sound/soc/fsl/Makefile | 6 +
sound/soc/fsl/fsl_rpmsg.c | 279 ++++++
sound/soc/fsl/fsl_rpmsg.h | 35 +
sound/soc/fsl/imx-audio-rpmsg.c | 140 +++
sound/soc/fsl/imx-pcm-rpmsg.c | 918 ++++++++++++++++++
sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++
sound/soc/fsl/imx-rpmsg.c | 150 +++
sound/soc/soc-component.c | 14 +
sound/soc/soc-pcm.c | 2 +
12 files changed, 2197 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
create mode 100644 sound/soc/fsl/fsl_rpmsg.c
create mode 100644 sound/soc/fsl/fsl_rpmsg.h
create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h
create mode 100644 sound/soc/fsl/imx-rpmsg.c
--
2.27.0
Current snd_soc_fixup_dai_links_platform_name() creates name first (A),
and checks setup target pointer (B), and set it (C).
We should check target pointer first IMO.
This patch exchange the order to (B) -> (A) -> (C).
int snd_soc_fixup_dai_links_platform_name(...)
{
...
/* set platform name for each dailink */
for_each_card_prelinks(card, i, dai_link) {
(A) name = devm_kstrdup(...);
if (!name)
return -ENOMEM;
(B) if (!dai_link->platforms)
return -EINVAL;
/* only single platform is supported for now */
(C) dai_link->platforms->name = name;
}
return 0;
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8735wnaoon.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-pcm has very similar but different DPCM BE DAI stop operation at
1) dpcm_be_dai_startup() error case rollback
2) dpcm_be_dai_startup_unwind()
3) dpcm_be_dai_shutdown()
The differences are
1) for rollback
2) Doesn't check by snd_soc_dpcm_be_can_update() (Is this bug ?)
3) Do soc_pcm_hw_free() if it was not !OPENed and !HW_FREEed,
and call soc_pcm_close().
We can share same code by
1) hw_free is not needed. Needs last dpcm as rollback.
2) hw_free is not needed.
3) hw_free is needed.
This patch adds new dpcm_be_dai_stop() and share these 3.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a6rduoam.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5645.c is the only user of the rt5645_platform_data,
move its definition to sound/soc/codecs/rt5645.c and remove the now
empty include/sound/rt5645.h file.
Note since the DMI quirk mechanism uses pointers to the
rt5645_platform_data struct we can NOT simply add its members to
the rt5645_priv struct and completely remove the struct.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210306230223.516566-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pointer to of_phandle_args passed to snd_soc_get_dai_name() and
of_xlate_dai_name() implementations is not modified. Since it is being
used only to translate passed OF node to a DAI name, it should not be
modified, so mark it as const for correctness and safer code.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20210221153024.453583-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA/ASoC/SOF/SoundWire: fix Kconfig issues
In January, Intel kbuild bot and Arnd Bergmann reported multiple
issues with randconfig. This patchset builds on Arnd's suggestions to
a) expose ACPI and PCI devices in separate modules, while sof-acpi-dev
and sof-pci-dev become helpers. This will result in minor changes
required for developers/testers, i.e. modprobe snd-sof-pci will no
longer result in a probe. The SOF CI was already updated to deal with
this module dependency change and introduction of new modules.
b) Fix SOF/SoundWire/DSP_config dependencies by moving the code
required to detect SoundWire presence in ACPI tables to sound/hda.
Link: https://lore.kernel.org/r/20210302003125.1178419-1-pierre-louis.bossart@linux.intel.com
Multiple bug reports report issues with the SOF and SST drivers when
dealing with single microphone cases.
We currently read the DMIC array information unconditionally but we
don't check that the configuration type is actually a mic array.
When the DMIC link does not rely on a mic array configuration, the
recommendation is to check the format information to infer the maximum
number of channels, and map this to the number of microphones.
This leaves a potential for a mismatch between actual microphones
available in hardware and what the ACPI table contains, but we have no
other source of information.
Note that single microphone configurations can alternatively be
handled with a 'mic array' configuration along with a 'vendor-defined'
geometry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201251
BugLink: https://github.com/thesofproject/linux/issues/2725
Fixes: 7a33ea70e1 ('ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210302000146.1177770-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Updates for v5.12
Another quiet release in terms of features, though several of the
drivers got quite a bit of work and there were a lot of general changes
resulting from Morimoto-san's ongoing cleanup work.
- As ever, lots of hard work by Morimoto-san cleaning up the code and
making it more consistent.
- Many improvements in the Intel drivers including a wide range of
quirks and bug fixes.
- A KUnit testsuite for the topology code.
- Support for Ingenic JZ4760(B), Intel AlderLake-P, DT configured
nVidia cards, Qualcomm lpass-rx-macro and lpass-tx-macro
- Removal of obsolete SIRF prima/atlas, Txx9 and ZTE zx drivers.
The below patch series are to support Audio over HDMI.
The modification in this patch series shall allow I2S driver
to playback standard PCM format and IEC958 encoded format to
the ADV7511 HDMI chip.
ALSA IEC958 plugin will be used to compose the IEC958 format.
Existing hdmi-codec driver only support standard pcm format.
Support of IEC958 encoded format passdown from ALSA IEC958 plugin
is needed so that the IEC958 encoded data can be streamed to the
HDMI chip.
Sia Jee Heng (4):
ASoC: codec: hdmi-codec: Support IEC958 encoded PCM format
drm: bridge: adv7511: Support I2S IEC958 encoded PCM format
dt-bindings: sound: Intel, Keembay-i2s: Add hdmi-i2s compatible string
ASoC: Intel: KMB: Support IEC958 encoded PCM format
.../bindings/sound/intel,keembay-i2s.yaml | 1 +
drivers/gpu/drm/bridge/adv7511/adv7511.h | 1 +
.../gpu/drm/bridge/adv7511/adv7511_audio.c | 6 ++
include/sound/hdmi-codec.h | 5 ++
sound/soc/codecs/hdmi-codec.c | 4 +-
sound/soc/intel/keembay/kmb_platform.c | 73 ++++++++++++++++++-
sound/soc/intel/keembay/kmb_platform.h | 1 +
7 files changed, 89 insertions(+), 2 deletions(-)
base-commit: 2557c711b87cd42bb22be9ca6ff3fce038624f30
--
2.18.0
The extended HDA bus (hdac_ext) provides interfaces for more
fine-grained control of individual links than what plain HDA
provides for. Links can be powered off when they are not used and if
all links are released, controller can shut down the command DMA.
These interfaces are currently not used by common HDA codec drivers.
When a HDA codec is runtime suspended, it calls snd_hdac_codec_link_down(),
but there is no link to the HDA extended bus, and on controller side
the links are shut down only when all codecs are suspended.
This patch adds link_power() to hdac_bus ops. Controllers using the HDA
extended core, can use this to plug in snd_hdac_ext_bus_link_power() to
implement more fine-grained control of link power.
No change is needed for plain HDA controllers nor to existing HDA
codec drivers.
Co-developed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210205184630.1938761-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit e7bbb7acab ("dmaengine: add peripheral configuration")
adds peripheral configuration for dma_slave_config.
This configuration is useful for some audio peripherals, for
example, the peripheral supports multi fifos, we can
let the DMA know which fifos are selected. So also add
this configuration for snd_dmaengine_dai_dma_data.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1612509985-11063-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This change adds audio jack injection feature through debugfs, with
this feature, we could validate alsa userspace changes by injecting
plugin or plugout events to the non-phantom audio jacks.
With this change, the sound core will build the folders
$debugfs_mount_dir/sound/cardN if SND_DEBUG and DEBUG_FS are enabled.
And if users also enable the SND_JACK_INJECTION_DEBUG, the jack
injection nodes will be built in the folder cardN like below:
$tree $debugfs_mount_dir/sound
$debugfs_mount_dir/sound
├── card0
│ ├── HDMI_DP_pcm_10_Jack
│ │ ├── jackin_inject
│ │ ├── kctl_id
│ │ ├── mask_bits
│ │ ├── status
│ │ ├── sw_inject_enable
│ │ └── type
...
│ └── HDMI_DP_pcm_9_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── card1
├── HDMI_DP_pcm_5_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
...
├── Headphone_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── Headset_Mic_Jack
├── jackin_inject
├── kctl_id
├── mask_bits
├── status
├── sw_inject_enable
└── type
The nodes kctl_id, mask_bits, status and type are read-only, users
could check jack or jack_kctl's information through them.
The nodes sw_inject_enable and jackin_inject are directly used for
injection. The sw_inject_enable is read-write, users could check if
software injection is enabled or not on this jack, and users could
echo 1 or 0 to enable or disable software injection on this jack. Once
the injection is enabled, the jack will not change by hardware events
anymore, once the injection is disabled, the jack will restore the
last reported hardware events to the jack. The jackin_inject is
write-only, if the injection is enabled, users could echo 1 or 0 to
this node to inject plugin or plugout events to this jack.
For the detailed usage information on these nodes, please refer to
Documentation/sound/designs/jack-injection.rst.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210127085639.74954-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's often the case that we would write or read a particular field
in register. With the current soc_component apis, reading a particular
field in register would involve first read the register and then
perform shift operations.
Ex:
to read from a field mask of 0xf0
val = snd_soc_component_read(component, reg);
field = ((val & 0xf0) >> 0x4);
This is sometimes prone to errors and code become less readable!
With this new api we could just do
field = snd_soc_component_read_field(component, reg, 0xf0);
this makes it bit simple, easy to write and less error prone!
This also applies to writing!
There are various places in kernel which provides such field interfaces
however soc_component seems to be missing this.
This patch is inspired by FIELD_GET/FIELD_PREP macros in include/linux/bitfield.h
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210126171749.1863-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The fix for a long-standing USB-audio bug required one more dependency
variable to be added to the hw constraints. Unfortunately I didn't
realize at debugging that the new addition may result in the overflow
of the dependency array of each snd_pcm_hw_rule (up to three plus a
sentinel), because USB-audio driver adds one more dependency only for
a certain device and bus, hence it works as is for many devices. But
in a bad case, a simple open always results in -EINVAL (with kernel
WARNING if CONFIG_SND_DEBUG is set) no matter what is passed.
Since the dependencies are real and unavoidable (USB-audio restricts
the hw_params per looping over the format/rate/channels combos), the
only good solution seems to raise the bar for one more dependency for
snd_pcm_hw_rule -- so does this patch: now the hw constraint
dependencies can be up to four.
Fixes: 506c203cc3 ("ALSA: usb-audio: Fix hw constraints dependencies")
Reported-by: Jamie Heilman <jamie@audible.transient.net>
Link: https://lore.kernel.org/r/20210123155730.22576-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_runtime / snd_soc_dai / snd_soc_dai_driver / snd_soc_dai_link
have related parameter which is similar but not same naming.
struct snd_pcm_runtime {
...
(A) unsigned int rate;
...
(B) unsigned int sample_bits;
...
};
struct snd_soc_dai {
...
(A) unsigned int rate;
(B) unsigned int sample_bits;
...
};
struct snd_soc_dai_driver {
...
(A) unsigned int symmetric_rates:1;
(B) unsigned int symmetric_samplebits:1;
...
};
struct snd_soc_dai_link {
...
(A) unsigned int symmetric_rates:1;
(B) unsigned int symmetric_samplebits:1;
...
};
Because it is similar but not same naming rule,
code can be verbose / can't share macro.
This patch sync naming rule for framework.
- xxx_rates;
+ xxx_rate;
- xxx_samplebits;
+ xxx_sample_bits;
old name will be removed if all drivers were switched
to new naming rule.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wnweolj6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ECS EF20EA laptop use gpio for jack detection instead of rt5645
rt5645 JD. However, the GPIO polarity is inverse for hp-detect based
on the _DSD property of the RTK2 device.
Name (_DSD, Package () {
ToUUID("daffd814-6eba-4d8c-8a91-bc9bbf4aa301"),
Package () {
Package () {"hp-detect-gpio", Package() {^RTK2, 0, 0, 1 }},
}
})
This flag will invert the hp-detect gpio polarity.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-4-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
