ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw
Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC
connected over SoundWire. This device has two SoundWire devices RX and
TX respectively. This bindings is for those slave devices on WCD9380/WCD9385.
ASoC: dt-bindings: wcd938x: add bindings for wcd938x
Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC
connected over SoundWire. This device has two SoundWire device RX and
TX respectively, supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH,
7 x TX diff inputs, 8 DMICs, MBHC.
Maxime Ripard [Tue, 25 May 2021 13:23:47 +0000 (15:23 +0200)]
ASoC: hdmi-codec: Add a prepare hook
The IEC958 status bit is usually set by the userspace after hw_params
has been called, so in order to use whatever is set by the userspace, we
need to implement the prepare hook. Let's add it to the hdmi_codec_ops,
and mandate that either prepare or hw_params is implemented.
Maxime Ripard [Tue, 25 May 2021 13:23:46 +0000 (15:23 +0200)]
ASoC: hdmi-codec: Add iec958 controls
The IEC958 status bits can be exposed and modified by the userspace
through dedicated ALSA controls.
This patch implements those controls for the hdmi-codec driver. It
relies on a default value being setup at probe time that can later be
overridden by the control put.
The hw_params callback is then called with a buffer filled with the
proper bits for the current parameters being passed on so the underlying
driver can just reuse those bits as is.
Maxime Ripard [Tue, 25 May 2021 13:23:44 +0000 (15:23 +0200)]
ALSA: iec958: Split status creation and fill
In some situations, like a codec probe, we need to provide an IEC status
default but don't have access to the sampling rate and width yet since
no stream has been configured yet.
Each and every driver has its own default, whereas the core iec958 code
also has some buried in the snd_pcm_create_iec958_consumer functions.
Let's split these functions in two to provide a default that doesn't
rely on the sampling rate and width, and another function to fill them
when available.
Maxime Ripard [Tue, 25 May 2021 13:23:43 +0000 (15:23 +0200)]
ALSA: doc: Clarify IEC958 controls iface
The doc currently mentions that the IEC958 Playback Default should be
exposed on the PCM iface, and the Playback Mask on the mixer iface.
It's a bit confusing to advise to have two related controls on two
separate ifaces, and it looks like the drivers that currently expose
those controls use any combination of the mixer and PCM ifaces.
Let's try to clarify the situation a bit, and encourage to at least have
the controls on the same iface.
ASoC: mediatek: mtk-btcvsd: Fix an error handling path in 'mtk_btcvsd_snd_probe()'
If an error occurs after a successful 'of_iomap()' call, it must be undone
by a corresponding 'iounmap()' call, as already done in the remove
function.
While at it, remove the useless initialization of 'ret' at the beginning of
the function.
Mark Brown [Mon, 7 Jun 2021 18:53:01 +0000 (19:53 +0100)]
Merge series "ASoC: adds new .auto_selectable_formats support" from Kuninori Morimoto <[email protected]>:
Hi Mark
These are v3 of "ASoC: adds new .get_fmt support",
but renamed Subject.
This is a little bit challenging patch-set.
The idea/code is almost same as v1 / v2.
v3 has "priority" support.
We need to set dai_link->dai_fmt to select CPU/Codec settings,
and it is selected by Sound Card Driver, today.
Because of it, Sound Card user need to know both CPU / Codec
available dai_fmt, and needs to select it.
For example simple-card / audio-graph case, it is selected by
"format" and "bitclock/frame-master/inversion" on DT.
But, it can be automatically selected if both CPU and Codec drivers
indicate it to ALSA SoC Framework, somehow.
By this patch, dai_fmt can be automatically selected from each
driver if both CPU / Codec driver had .auto_selectable_formats.
Automatically selectable *field* is depends on each drivers.
For example, some driver want to select format "automatically",
but want to select other fields "manually", because of complex limitation.
Or other example, in case of both CPU and Codec are possible to be
clock provider, but the quality was different.
In these case, user need/want to *manually* select each fields
from Sound Card driver.
It uses Sound Card specified fields preferentially, and try to select
non-specific fields from CPU and Codec driver settings if driver had
.auto_selectable_formats.
In other words, we can select all dai_fmt via Sound Card driver
same as before.
Select dai_fmt 100% automatically is very difficult and will be very complex,
but select automatically some fields only is very easy, I guess.
This patch-set is based on such assumption.
v1 -> v2
- Add more detail explanation on git-log, code, comment.
- Possible to be Clock/Frame provider is depends on driver's situation.
v2 -> v3
- has priority
- tidyup function explanation for snd_soc_dai_get_fmt()
- Each driver don't try to have SND_SOC_DAIFMT_CBx_CFx to avoid confusion
Link: https://lore.kernel.org/r/[email protected] Link: https://lore.kernel.org/r/[email protected]
Kuninori Morimoto (7):
ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside
ASoC: soc-core: add snd_soc_runtime_get_dai_fmt()
ASoC: ak4613: add .auto_selectable_formats support
ASoC: pcm3168a: add .auto_selectable_formats support
ASoC: rsnd: add .auto_selectable_formats support
ASoC: fsi: add .auto_selectable_formats support
ASoC: hdmi-codec: add .auto_selectable_formats support
The pointer node is being initialized with a value that is never read and
it is being updated later with a new value. The initialization is
redundant and can be removed.
The function is missing a of_node_put on node, fix this by adding the call
before returning.
ASoC is using dai_link which specify DAI format (= dai_link->dai_fmt),
and it is selected by "Sound Card" driver in corrent implementation.
In other words, Sound Card *needs* to setup it.
But, it should be possible to automatically selected from CPU and
Codec driver settings.
This patch adds new .auto_selectable_formats support
at snd_soc_dai_ops.
By this patch, dai_fmt can be automatically selected from each
driver if both CPU / Codec driver had it.
Automatically selectable *field* is depends on each drivers.
For example, some driver want to select format "automatically",
but want to select other fields "manually", because of complex limitation.
Or other example, in case of both CPU and Codec are possible to be
clock provider, but the quality was different.
In these case, user need/want to *manually* select each fields
from Sound Card driver.
This .auto_selectable_formats can set priority.
For example, no limitaion format can be HI priority,
supported but has picky limitation format can be next priority, etc.
It uses Sound Card specified fields preferentially, and try to select
non-specific fields from CPU and Codec driver automatically
if all drivers have .auto_selectable_formats.
In other words, we can select all dai_fmt via Sound Card driver
same as before.
Mark Brown [Fri, 4 Jun 2021 16:26:54 +0000 (17:26 +0100)]
Merge series "ASoC: codecs: wcd934x: add Headset and button detection support" from Srinivas Kandagatla <[email protected]>:
This patchset adds support to MBHC(Multi Button Headset Control) block found in
Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and
electrical insert/removal detection along with 8 buttons detection,
Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R.
Eventhough MBHC block supports things like OverCurrent detection, Currently its
reported as a kernel debug message. Should this be reported as an uevent to
userspace? like the way USB reports?
Any suggestions?
First patch adds a common mbhc driver and the second one wcd934x specific driver
changes along with sdm845 soundcard related changes.
Common wcd-mbhc-v2 driver should be reusable across multiple codecs like
WCD9335, WCD934x, WCD937x and WCD938x.
Most of the work is derived from downstream Qualcomm kernels.
Credits to various Qualcomm authors from Patrick Lai's team who have
contributed to this code.
Changes since v2:
- switched to EXPORT_SYMBOL_GPL from EXPORT_SYMBOL
- converted one of the if else to switch case.
Srinivas Kandagatla (4):
ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection
ASoC: codecs: wcd: add multi button Headset detection support
ASoC: codecs: wcd934x: add mbhc support
ASoC: qcom: sdm845: add jack support for WCD934x
WCD934x has Multi Button Headset Control hardware to support Headset insertion,
type detection, 8 headset buttons detection, Over Current detection and Impedence
measurements.
This patch adds support for this feature via common mbhc layer.
ASoC: codecs: wcd: add multi button Headset detection support
Most new Qualcomm WCD codecs support MBHC(Multi Button Headset Control) via ADC.
This patchset adds support to Common parts of this MBHC support so that
WCD codecs need not duplicate them. To do that codec exposes set of
register fields and callbacks to this common driver to control it.
ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection
Add bindings required for Multi Button Headset detection.
WCD934x support Headsets with upto 8 buttons including, impedance measurement
on both L/R Headset speakers and cross connection detection.
Yufen Yu [Mon, 24 May 2021 09:35:21 +0000 (05:35 -0400)]
ASoC: img: Fix PM reference leak in img_i2s_in_probe()
pm_runtime_get_sync will increment pm usage counter even it failed.
Forgetting to putting operation will result in reference leak here.
Fix it by replacing it with pm_runtime_resume_and_get to keep usage
counter balanced.
Colin Ian King [Thu, 3 Jun 2021 11:03:15 +0000 (12:03 +0100)]
ASoC: rsnd: check for zero node count
Most callers of_get_child_count() check that "nr" is non-zero so it
causes a static checker warning when we don't do that here. This
does not cause a problem or a crash, but having zero SSUIes does not
make sense either so let's add a check.
set priv->adg before rsnd_adg_get_clkin/out() to be more simple code.
Nothing is changed, but is preparation for
next "ASoC: rsnd: adg: use more simple method for null_clk" patch
ASoC: rsnd: adg: supply __printf(x, y) formatting for dbg_msg()
Fixes the following W=1 kernel build warning(s):
sound/soc/sh/rcar/adg.c: In function 'dbg_msg':
sound/soc/sh/rcar/adg.c:594:2: warning: function 'dbg_msg' might \
be a candidate for 'gnu_printf' format attribute\
[-Wsuggest-attribute=format]
Chris Morgan [Tue, 1 Jun 2021 21:44:24 +0000 (16:44 -0500)]
ASoC: rk817: fix a warning in rk817_probe()
The return value of snd_soc_component_write() is stored but not
evaluated and this results in a warning when W=1 is set. Stop storing
the return value to be consistent with all other calls of
snd_soc_component_write() and to remove the warning.
Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Chris Morgan <[email protected]> Signed-off-by: Lee Jones <[email protected]>
Mark Brown [Tue, 1 Jun 2021 17:10:41 +0000 (18:10 +0100)]
Merge series "ASoC: Constify snd_compress_ops" from Rikard Falkeborn <[email protected]>:
The only use of the static and global snd_compress_ops structs is to
assign their address to the compress_ops field in the
snd_soc_component_driver struct which is a pointer to const. Make them
const to allow the compiler to put them in read-only memory.
ASoC: SOF: Intel: hda: don't print ROM status if cl_dsp_init() fails
cl_dsp_init() dumps the ROM status if it fails after max
attempts before powering off the DSP. Remove the duplicate
log to print the ROM status and error in
hda_dsp_cl_boot_firmware(). These values are invalid anyway
as the DSP is already powered off.
The only usage of sof_probe_compressed_ops is to assign its address to
the compress_ops field in the snd_soc_component_driver struct, which is
a pointer to const. The assignment is done in sound/soc/sof/pcm.c. Make
it const to allow the compiler to put it in read-only memory.
The snd_compress_ops structs are only stored in the compress_ops field
of a snd_soc_component_driver struct, so make it const to allow the
compiler to put it in read-only memory.
The snd_compress_ops structs are only stored in the compress_ops field
of a snd_soc_component_driver struct, so make it const to allow the
compiler to put it in read-only memory.
The snd_compress_ops structs are only stored in the compress_ops field
of a snd_soc_component_driver struct, so make it const to allow the
compiler to put it in read-only memory.
The snd_compress_ops structs are only stored in the compress_ops field
of a snd_soc_component_driver struct, so make it const to allow the
compiler to put it in read-only memory.
In general Renesas SoC's SSI/SRC are all enabled, but some SoC is not.
H2 E2
SRC0 <=
SRC1 SRC1
SRC2 SRC2
... ...
Renesas Sound driver is assuming that *all* modules are
enabled, and thus it is using *data array* to access each modules.
Because of it, we have been using "status = disabled" at DT,
and using *full size* array but avoiding disabled module.
This patch adds "char *name" to rsnd_dma_request_channel().
It is not yet used so far, but is preparation for
next "ASoC: rsnd: adjust disabled module" patch
This patch adds "char *name" to rsnd_parse_connect_common().
It is not yet used so far, but is preparation for
next "ASoC: rsnd: adjust disabled module" patch
Chris Morgan [Wed, 19 May 2021 20:37:53 +0000 (15:37 -0500)]
dt-bindings: Add Rockchip rk817 audio CODEC support
Create dt-binding documentation to document rk817 codec.
New property name of rockchip,mic-in-differential added to control if
the microphone is in differential mode or not.
Chris Morgan [Wed, 19 May 2021 20:37:52 +0000 (15:37 -0500)]
ASoC: Add Rockchip rk817 audio CODEC support
Add support for the Rockchip rk817 audio codec integrated into the
rk817 PMIC. This is based on the sources provided by Rockchip from
their BSP kernel.
Renesas Sound uses many modules (SSI/SSIU/SRC/CTU/MIX/DVC/DMA),
and supports complex connections/path.
Thus each modules needs to save its status to correctly control it.
This status is updated when by .trigger, and .hw_params/.hw_free.
Renesas Sound is protecting modules by using lock when .trigger,
but it was not enough to protecting each modules "status" if it was
used from many paths.
1) .hw_params/.hw_free update status
2) another doesn't update status, but overwrites by same value
This patch do
1) protects .hw_params/.hw_free by lock
2) do nothing if no status update
Without this patch, protected mod->status (= .trigger) might be
overwrote by non protected mod->status (= .hw_params / .hw_free),
and in such case, CTU/MIX/DVC/SSIU/SSI which are used from
many paths might get damage.
If above issue happens, Renesas Sound will be hung (= silence)
and never be recoverd.
I could reproduce this issue by continue playing very short sound
with loop very long term (3-4 hours) through 2 inputs (= MIXer).
For updating rsnd_status_update(), this patch removes rsnd_dai_call()
debug message. Because we already have debugfs support, and is not
good match to new code.
Current rsnd is using dev_dbg() if irq error happen,
but it makes debug very difficult if some strange things happen.
This patch uses dev_info() for it, and rename the macro name.
ASoC: rsnd: indicate unknown error at rsnd_dai_call()
Current rsnd_dai_call() doesn't indicate error message,
thus it is very difficult to know the issue
when strange things happen.
This patch indicates error for it.
ASoC: rsnd: call unregister for null_hw when removed
commit d6956a7dde6fb ("ASoC: rsnd: add null CLOCKIN support")
added null_clk, but it is using local static valuable.
It will be leaked if rsnd driver was removed.
This patch moves it to priv, and call unregister when removing.
Wei Yongjun [Mon, 24 May 2021 13:35:53 +0000 (13:35 +0000)]
ASoC: imx-card: Make some symbols static
The sparse tool complains as follows:
sound/soc/fsl/imx-card.c:121:27: warning:
symbol 'ak4458_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:138:31: warning:
symbol 'ak4458_tdm_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:149:27: warning:
symbol 'ak4497_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:166:27: warning:
symbol 'ak5558_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:180:31: warning:
symbol 'ak5558_tdm_fs_mul' was not declared. Should it be static?
Those symbols are not used outside of imx-card.c, so marks
them static.
The 16 Bits, 2 channels, 48K sample rate use case needs
to configure a safer pll_divout during the start of PLL
After 800us from the start of PLL the correct pll_divout
can be set
Stephan Gerhold [Thu, 13 May 2021 10:41:29 +0000 (12:41 +0200)]
ASoC: codecs: Add driver for NXP/Goodix TFA989x (TFA1) amplifiers
NXP's TFA98xx (now part of Goodix) are fairly popular speaker amplifiers
used in many smartphones and tablets. Most of them are sold as "smart
amplifiers" with built-in "CoolFlux DSP" that is used for volume control,
plus a "sophisticated speaker-boost and protection algorithm".
Unfortunately, they are also almost entirely undocumented. The short
datasheets (e.g. [1] for TFA9897) describe the available features,
but do not provide any information about the registers or how to use
the "CoolFlux DSP".
The amplifiers are most often configured through proprietary userspace
libraries. There are also some (rather complex) kernel drivers (e.g. [2])
but even those rely on obscure firmware blobs for configuration (so-called
"containers"). They seem to contain different "profiles" with tuned speaker
settings, sample rates and volume steps (which would be better exposed
as separate ALSA mixers).
The format of the firmware files seems to have changed a lot over the time,
so it's not even possible to simply re-use the firmware originally provided
by the vendor.
Overall, it seems close to impossible to develop a proper mainline driver
for these amplifiers that could make proper use of the built-in DSP.
This commit implements a compromise: At least the TFA1 family of the
TFA98xx amplifiers (usually called TFA989x) provide a way to *bypass*
the DSP using a special register sequence. The register sequence can be
found in similar variations in the kernel drivers from lots of vendors
e.g. in [3] and was probably mainly used for factory testing.
With the DSP bypassed, the amplifier acts mostly like a dumb standard
speaker amplifier, without (hardware) volume control. However, the setup
is much simpler and it works without any obscure firmware.
This driver implements the DSP bypass combined with chip-specific
initialization sequences adapted from [2]. Only TFA9895 is supported in
this initial commit. Except for the lack of volume control I can not hear
any difference with or without the DSP, it works just fine.
This driver allows the speaker to work on mainline Linux running on the
Samsung Galaxy A3/A5 (2015) [TFA9895] and Alcatel Idol 3 [TFA9897].
TFA9897 support will be added in separate patch set later.
Stephan Gerhold [Thu, 13 May 2021 10:41:28 +0000 (12:41 +0200)]
ASoC: dt-bindings: codecs: Add bindings for nxp, tfa989x
NXP/Goodix TFA989X (TFA1) amplifiers are controlled via an I2C bus.
Add simple device tree bindings that describe how to set them up
in the device tree.
Right now only nxp,tfa9895 is supported but this will be extended
to at least nxp,tfa9897 in the near future.
projects / linux.git / log