On i.MX9x platforms, the REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID,
REG_MICFIL_PARAM are added, but they are not existed on i.MX8x
platforms.
Use the existed micfil->soc->use_verid to distinguish the access
permission for these platforms.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1721897694-6088-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the i.MX9x platforms, the mask of FIFO watermark
is 0x1F, on i.MX8x platforms, the mask of FIFO watermark
is 0X7. So use the mask 0x1F for all platforms to make them
compatible.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1721897694-6088-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The static snd_soc_dai_link_components cause conflict for multiple
instances of this generic driver. For example, when there is
wm8962 and SPDIF case enabled together, the contaminated
snd_soc_dai_link_components will cause another device probe fail.
Fixes: 6d174cc4f2 ("ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1721877773-5229-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The imx-spdif machine driver creates audio card to directly use an
S/PDIF device. However, it doesn't support interacting with an ASRC.
fsl-asoc-card already has the support to create audio card which can
use the ASRC.
Merge the S/PDIF support from imx-spdif into driver fsl-asoc-card
to extend the support of S/PDIF audio card with the use of ASRC devices.
fsl-asoc-card uses slightly different DT properties than imx-spdif:
* the "spdif-controller" property from imx-spdif is named "audio-cpu" in
fsl-asoc-card.
* fsl-asoc-card uses codecs explicitly declared in DT
with "audio-codec".
With an SPDIF, codec drivers spdif_transmitter and
spdif_receiver should be used.
Driver imx-spdif used instead the dummy codec and a pair of
boolean properties, "spdif-in" and "spdif-out".
To keep backward compatibility, support for "spdif-controller",
"spdif-in" and "spdif-out" is also added to fsl-asoc-card.
However, it is recommended to use the new properties if possible.
It is better to declare transmitter and/or receiver in DT
than using the dummy codec.
DTs using compatible "fsl,imx-audio-spdif" are still compatible, and
fsl-asoc-card will behave the same as imx-spdif
for these DTs.
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240627083104.123357-5-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Adapt the driver to work with configurations using two codecs or more.
Modify fsl_asoc_card_probe() to handle use cases where 2 codecs are
given in the device tree.
This will be needed to add support for the SPDIF.
Use cases using one codec will ignore any given codecs other than the
first.
Co-developed-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240627083104.123357-4-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a second dai link component for codecs that will be used for use
cases with 2 codecs.
It is needed for future integration of the SPDIF support, which will
use spdif_receiver and spdif_transmitter drivers.
To prevent deferring in use cases using only one codec, also set
by default the number of codecs to 1 for the relevant dai links.
Co-developed-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240627083104.123357-3-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for dai links using multiple codecs for multi-codec
use cases.
Co-developed-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com>
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240627083104.123357-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current fsl_qmc_audio works in interleaved mode. The audio samples
are interleaved and all data are sent to (received from) one QMC
channel.
Using several QMC channels, non interleaved mode can be easily
supported. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.
In terms of constraints and settings, the two modes are slightly
different:
- Interleaved mode:
- The sample size should fit in the number of time-slots available
for the QMC channel.
- The number of audio channels should fit in the number of
time-slots (taking into account the sample size) available for the
QMC channel.
- Non-interleaved mode:
- The number of audio channels is the number of available QMC
channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC
channel.
Add support for the non-interleaved mode allowing multiple QMC channel
per DAI. The DAI switches in non-interleaved mode when more that one QMC
channel is available.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-11-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Constraints are set by qmc_dai_startup(). These constraints are specific
to the interleaved mode.
With the future introduction of support for non-interleaved mode, a new
set of constraints will be set. To make the code clear and keep
qmc_dai_startup() simple, extract the current interleaved mode
constraints settings to a specific function.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-7-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Submitting data to QMC channels is done in several places: transfer
completions and DAI start. The operation done is simple and consist in
one function call.
With the future introduction of support for non-interleaved mode,
submitting data will be more complex.
To avoid copy/paste of code in several places, introduce
qmc_audio_pcm_{read,write}_submit() whose goal is to handle this
data submission.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-6-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current QMC audio driver uses only one QMC channel per DAI. The
context used by QMC channel transfer (read and write) completion
routines does not contains any QMC channel and the only one available
per DAI is used to schedule the next transfer.
This works pretty well with only one QMC channel per DAI.
The future support for non-inlerleave mode will use several QMC channel
per DAI. In that case, QMC channel transfer completion routines need to
identify the QMC channel related to the completion.
In order to fill this lack, even if identifying the current QMC channel
among several QMC channels is not needed for the current code, add one
indirection level and introduce the qmc_dai_chan data structrure.
This structure contains the QMC channel involved in the completion and
refererences to the runtime context (capture and playback) used by the
DAI.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-5-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver mixes some internal values for channel DMA buffer handling
and PCM pointer handling. In the currently supported interleaved mode,
this mix does not lead to any issues but in order to prepare the
support for the non-interleaved mode, having them clearly separated will
ease the support and avoid additional computation to convert values used
in channel DMA buffer management in values usable for PCM pointer.
Use a specific set of variable for PCM pointer handling and an other set
for channel DMA buffer.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-4-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
./scripts/checkpatch.pl --strict --codespell detected several issues
when running on the fsl_qmc_audio.c file:
- CHECK: spaces preferred around that '*' (ctx:VxV)
- CHECK: Alignment should match open parenthesis
- CHECK: Comparison to NULL could be written "!prtd"
- CHECK: spaces preferred around that '/' (ctx:VxV)
- CHECK: Lines should not end with a '('
- CHECK: Please don't use multiple blank lines
Some of them are present several times.
Fix all of these issues without any functional changes.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-3-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the current flow all interrupts are disabled in runtime suspend
phase. However interrupts enablement only exists in fsl_xcvr_prepare().
After resume fsl_xcvr_prepare() may not be called so it will cause all
interrupts still disabled even if resume from suspend. Interrupts
should be explictily enabled after resume.
Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After
resume from suspend DPATH should be reset otherwise there'll be channel
swap issue.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240628094354.780720-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
bitfield.h is not explicitly included but it is required for FIELD_PREP
to be expanded by the preprocessor. If it is not implicitly included,
there will be a compiler error (as seen with ARCH=hexagon allmodconfig):
sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP);
| ^
sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8'
42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH'
65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH'
70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */
| ^
Include bitfield.h explicitly, so that FIELD_PREP is always expanded,
clearing up the compiler error.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that ret may be used uninitialized in
lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with
CONFIG_WERROR):
sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized]
326 | "failed to init register map: %d\n", ret);
| ^~~
sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning
310 | int ret;
| ^
| = 0
1 error generated.
One solution would be a small refactoring of the second parameter in
dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in
the if statement. However, a nicer solution for debugging purposes,
which is the point of this statement, would be to use the '%pe'
specifier to symbolically print the error pointer value. Do so, which
eliminates the uninitialized use of ret, clearing up the warning.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-uninitialized-ret-v1-1-985d86189739@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add compatible string and specific soc data to support rpmsg sound card
on i.MX95 platform.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240626071202.7149-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There will be three devices for this sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output. then capture device and playback device can be configured
with different master/slave mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As audmix requires playback and capture stream in different
master/slave mode, so separate playback and capture stream to
different DAI. There are three DAIs required, two DAIs for playback
one DAI for capture.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
+-------+ +-----------+
| SAI1 | --TX--> | |
| | <--RX-- | |
+-------+ | |
| AUDIOMIX |
+-------+ | |
| SAI2 | --TX--> | |
+-------+ +-----------+
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately. but only
defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx()
ops function for current case, in the future, the other ops
function for TX and RX can be defined if required.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With ARCH=arm, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/fsl/imx-pcm-fiq.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240616-md-arm-sound-soc-fsl-v2-1-228772e81a54@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
of_gpio.h is deprecated and subject to remove. The drivers in question
don't use it, simply remove the unused header.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20240605221446.2624964-3-andriy.shevchenko@linux.intel.com
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
On i.MX95, the XCVR uses a new PLL in the PHY, which is
General Purpose (GP) PLL. Add GP PLL configuration support
in the driver and add the 'pll_ver' flag to distinguish
different PLL on different platforms.
The XCVR also use PHY but limited for SPDIF only case
Add 'use_phy' flag to distinguish these platforms.
When there are 'pll8k' and 'pll11k' clock existing, the clock
source of 'phy_clk' can be changed for different sample rate
requirement.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/1716972002-2315-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/fsl/imx-pcm-dma.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240602-md-snd-fsl-imx-pcm-dma-v1-1-e7efc33c6bf3@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
There are two MQS instances on the i.MX95 platform.
The definition of bit positions in the control register are
different. In order to support these MQS modules, define
two compatible strings to distinguish them.
There are two MQS instances on the i.MX95 platform.
The definition of bit positions in the control register are
different. In order to support these MQS modules, define
two compatible strings to distinguish them.
Define different soc data according to compatible strings
On i.MX95 one instance in nect-mix is supported by this
commit, another instance in always-on-mix is not supported,
which depends on System Manager function readiness.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://msgid.link/r/1716347305-18457-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace SET_RUNTIME_PM_OPS() with its modern alternative RUNTIME_PM_OPS().
The combined usage of pm_ptr() and RUNTIME_PM_OPS() allows the
compiler to evaluate if the runtime suspend/resume() functions are used
at build time or are simply dead code.
This allows removing the __maybe_unused notation from the runtime
suspend/resume() functions.
Signed-off-by: Fabio Estevam <festevam@denx.de>
Link: https://msgid.link/r/20240507173735.3456334-3-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace SET_RUNTIME_PM_OPS() with its modern alternative RUNTIME_PM_OPS().
The combined usage of pm_ptr() and RUNTIME_PM_OPS() allows the
compiler to evaluate if the runtime suspend/resume() functions are used
at build time or are simply dead code.
This allows removing the __maybe_unused notation from the runtime
suspend/resume() functions.
Signed-off-by: Fabio Estevam <festevam@denx.de>
Link: https://msgid.link/r/20240507173735.3456334-2-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace SET_RUNTIME_PM_OPS() with its modern alternative RUNTIME_PM_OPS().
The combined usage of pm_ptr() and RUNTIME_PM_OPS() allows the
compiler to evaluate if the runtime suspend/resume() functions are used
at build time or are simply dead code.
This allows removing the __maybe_unused notation from the runtime
suspend/resume() functions.
Signed-off-by: Fabio Estevam <festevam@denx.de>
Link: https://msgid.link/r/20240507173735.3456334-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-11-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Static 'struct snd_pcm_hardware' is not modified by few drivers and its
copy is passed to the core, so it can be made const for increased code
safety.
Merge series from Chancel Liu <chancel.liu@nxp.com>:
echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware
(A) echo start > /sys/class/remoteproc/remoteproc0/state
(B) echo stop > /sys/class/remoteproc/remoteproc0/state
The rpmsg sound card is registered in (A) and unregistered in (B).
After "start", imx-audio-rpmsg registers devices for ASoC platform driver
and machine driver. Then sound card is registered. After "stop",
imx-audio-rpmsg unregisters devices for ASoC platform driver and machine
driver. Then sound card is unregistered.
The gpio field is not used anymore, remove the leftover.
This also fixes the compilation error after the ...
Fixes: 9855f05e55 ("ASoC: fsl: imx-es8328: Switch to using gpiod API")
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240325191341.3977321-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to support register and unregister rpmsg sound card through
remoteproc platform device for card to probe is registered in
imx-audio-rpmsg. ASoC machine driver no longer can get DT node of ASoC
CPU DAI device through parent device.
ASoC machine driver can get DT node of ASoC CPU DAI device with rpmsg
channel name acquired from platform specific data.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-6-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each rpmsg sound card sits on one rpmsg channel. Register CPU DAI with
name of rpmsg channel so that ASoC machine driver can easily link CPU
DAI with rpmsg channel name.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-5-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Let imx-audio-rpmsg register platform device for card. So that card
register and unregister can be controlled by rpmsg driver's register
and unregister.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-4-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This rpmsg driver registers device for ASoC platform driver. To align
with platform driver use rpmsg channel name to create device.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-3-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine driver uses rpmsg channel name to link this platform component.
However if the component is re-registerd card will not find this new
created component in snd_soc_try_rebind_card().
Explicitly register this component with rpmsg channel name so that
card can always find this component.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-2-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
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Merge tag 'asoc-v6.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.9
This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
snd_soc_card_get_kcontrol() must be holding a read lock on
card->controls_rwsem while walking the controls list.
Compare with snd_ctl_find_numid().
The existing function is renamed snd_soc_card_get_kcontrol_locked()
so that it can be called from contexts that are already holding
card->controls_rwsem (for example, control get/put functions).
There are few direct or indirect callers of
snd_soc_card_get_kcontrol(), and most are safe. Three require
changes, which have been included in this patch:
codecs/cs35l45.c:
cs35l45_activate_ctl() is called from a control put() function so
is changed to call snd_soc_card_get_kcontrol_locked().
codecs/cs35l56.c:
cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from
control get()/put() functions so is changed to call
snd_soc_card_get_kcontrol_locked().
fsl/fsl_xcvr.c:
fsl_xcvr_activate_ctl() is called from three places, one of which
already holds card->controls_rwsem:
1. fsl_xcvr_mode_put(), a control put function, which will
already be holding card->controls_rwsem.
2. fsl_xcvr_startup(), a DAI startup function.
3. fsl_xcvr_shutdown(), a DAI shutdown function.
To fix this, fsl_xcvr_activate_ctl() has been changed to call
snd_soc_card_get_kcontrol_locked() so that it is safe to call
directly from fsl_xcvr_mode_put().
The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been
changed to take a read lock on card->controls_rsem() around calls
to fsl_xcvr_activate_ctl(). While this is not very elegant, it
keeps the change small, to avoid this patch creating a large
collateral churn in fsl/fsl_xcvr.c.
Analysis of other callers of snd_soc_card_get_kcontrol() is that
they do not need any changes, they are not holding card->controls_rwsem
when they call snd_soc_card_get_kcontrol().
Direct callers of snd_soc_card_get_kcontrol():
fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function
fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler
Indirect callers via soc_component_notify_control():
codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler
codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler
codecs/ak4118.c: ak4118_irq_handler() - IRQ handler
codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used
Indirect callers via snd_soc_limit_volume():
qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function
ti/rx51.c: rx51_aic34_init() - DAI init function
I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c
and ak4118.c changes.
Backport note:
The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added
since the Fixes commit so won't all be present on older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 209c6cdfd2 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card")
Link: https://lore.kernel.org/r/20240221123710.690224-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
