idle_bias_on was set because cs42l42 has a "VMID" type pseudo-midrail
supply (named FILT+), and these typically take a long time to charge.
But the driver never enabled pm_runtime so it would never have powered-
down the cs42l42 anyway.
In fact, FILT+ can charge to operating voltage within 12.5 milliseconds
of enabling HP or ADC. This time is already covered by the startup
delay of the HP/ADC.
The datasheet warning about FILT+ taking up to 1 second to charge only
applies in the special cases that either the PLL is started or
DETECT_MODE set to non-zero while both HP and ADC are off. The driver
never does either of these.
Removing idle_bias_on allows the Soundwire host controller to suspend
if there isn't a snd_soc_jack handler registered.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-8-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for using CS42L42 as a SoundWire device.
SoundWire-specifics are kept separate from the I2S implementation as
much as possible, aiming to limit the risk of breaking the I2C+I2S
support.
There are some important differences in the silicon behaviour between
I2S and SoundWire mode that are reflected in the implementation:
- ASP (I2S) most not be used in SoundWire mode because the two interfaces
share pins.
- The SoundWire capture (record) port only supports 1 channel. It does
not have left-to-right duplication like the ASP.
- DP2 can only be prepared if the HP has powered-up. DP1 can only be
prepared if the ADC has powered-up. (This ordering restriction does
not exist for ASPs.) The SoundWire core port-prepare step is
triggered by the DAI-link prepare(). This happens before the
codec DAI prepare() or the DAPM sequence so these cannot be used
to enable HP/ADC. Instead the HP/ADC enable/disable are done during
the port_prep callback.
- The SRCs are an integral part of the audio chain but in silicon their
power control is linked to the ASP. There is no equivalent power link
to SoundWire DPs so the driver must take "manual" control of SRC power.
- The SoundWire control registers occupy the lower part of the SoundWire
address space so cs42l42 registers are offset by 0x8000 (non-paged) in
SoundWire mode.
- Register addresses are 8-bit paged in I2C mode but 16-bit unpaged in
SoundWire.
- Special procedures are needed on register read/writes to (a) ensure
that the previous internal bus transaction has completed, and
(b) handle delayed read results, when the read value could not be
returned within the SoundWire read command.
There are also some differences in driver implementation between I2S
and SoundWire operation:
- CS42L42 I2S does not runtime_suspend, but runtime_suspend/resume support
has been added into the driver in SoundWire mode as the most convenient
way to power-up the bus manager and to handle the unattach_request
condition, though the CS42L42 chip does not itself suspend or resume.
- Intel SoundWire host controllers have a low-power clock-stop mode that
requires resetting all peripherals when resuming. This means that the
interrupt registers will be reset in between the interrupt being
generated and the interrupt being handled, and since the interrupt
status is debounced, these values may not be accurate immediately,
and may cause spurious unplug events before settling.
- As in I2S mode, the PLL is only used while audio is active because
of clocking quirks in the silicon. For SoundWire the cs42l42_pll_config()
is deferred until the DAI prepare(), to allow the cs42l42_bus_config()
callback to set the SCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-7-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The chosen clocking configuration must give an internal MCLK (MCLKint)
that is an integer multiple of the sample rate.
On I2S each of the supported bit clock frequencies can only be generated
from one sample rate group (either the 44100 or the 48000) so the code
could use only the bitclock to look up a PLL config.
The relationship between sample rate and bitclock frequency is more
complex on Soundwire and so it is possible to set a frame shape to
generate a bitclock from the "wrong" group. For example 2*147 with a
48000 sample rate would give a bitclock of 14112000 which on I2S
could only be derived from a 44100 sample rate.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-4-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, port_prep callback only has commands for PRE_PREP, PREP,
and POST_PREP, which doesn't directly say whether this is for a
prepare or deprepare call. Extend the command list enum to say
whether the call is for prepare or deprepare aswell.
Also remove SDW_OPS_PORT_PREP from sdw_port_prep_ops as this is unused,
and update this enum to be simpler and more consistent with enum
sdw_clk_stop_type.
Note: Currently, the only users of SDW_OPS_PORT_POST_PREP are codec
drivers sound/soc/codecs/wsa881x.c and sound/soc/codecs/wsa883x.c, both
of which seem to assume that POST_PREP only occurs after a prepare,
even though it would also have occurred after a deprepare. Since it
doesn't make sense to mark the port prepared after a deprepare, changing
the enum to separate PORT_DEPREP from PORT_PREP should make the check
for PORT_PREP in those drivers be more logical.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-By: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20230127165111.3010960-2-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
Following is series of fixes and cleanups for core topology code. Few
patches fixing various problems all around and few fixing function
names.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Audio-Graph-Card and Simple-Audio-Card are similar Card
and are sharing same utils. Thus we can also sharing same schema.
This patch-set fixup some Renesas's "make dtbs_check".
Set the reserved bit 7 in the ANALOG_CTRL_REG for the TAS5720A-Q1 device,
when probing.
The datasheet mentions that the bit should be 1 during reset/powerup.
The device did not initialize before setting this value to 1. So, this
could be a quirk of this device. Or it could be a quirk with the board on
which it was tested.
That is why this patch is separate from the patch that adds support for the
TAS5720A-Q1 device.
Signed-off-by: Steffen Aschbacher <steffen.aschbacher@stihl.de>
Signed-off-by: Alexandru Ardelean <alex@shruggie.ro>
Link: https://lore.kernel.org/r/20230128082744.41849-3-alex@shruggie.ro
Signed-off-by: Mark Brown <broonie@kernel.org>
This change adds support the TAS5720A-Q1 audio codec, in the same driver as
tas5720.
Functionally, this driver is pretty similar to it's TAS5720x variant.
The first 3 registers are the same, so the main control and device
identification can happen with these registers.
The next registers differ.
This variant offers control (in the registers) for 2 speakers, which is
implemented here (in a basic manner).
Signed-off-by: Steffen Aschbacher <steffen.aschbacher@stihl.de>
Signed-off-by: Alexandru Ardelean <alex@shruggie.ro>
Link: https://lore.kernel.org/r/20230128082744.41849-2-alex@shruggie.ro
Signed-off-by: Mark Brown <broonie@kernel.org>
This is to be re-used in tas5720_mute() (which is part of the dai_ops) and
also in the tas5720_fault_check_work() hook.
The benefit here isn't too great (now).
It's only when we add support for a new device with a slightly different
regmap that this becomes more useful.
Signed-off-by: Alexandru Ardelean <alex@shruggie.ro>
Link: https://lore.kernel.org/r/20230128082744.41849-1-alex@shruggie.ro
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of using MIXer with Simple Audio Card, it needs below DT.
simple-audio-card,dai-link@1 {
cpu@0 {
...
};
cpu@1 {
...
};
...
};
This case, it requires "reg = <xxx>" which needs #address-cells/#size-cells,
but simple-audio-card.yaml is missing these. This patch adds it.
Without this patch, we will get below warning.
${LINUX}/arch/arm64/boot/dts/renesas/r8a77950-ulcb.dtb: sound: simple-audio-card,dai-link@0: '#address-cells', '#size-cells' do not match any of the regexes: '^codec(@[0-9a-f]+)?', '^cpu(@[0-9a-f]+)?', 'pinctrl-[0-9]+'
From schema: ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.yaml
Link: https://lore.kernel.org/r/167344317928.394453.14105689826645262807.robh@kernel.org
Reported-by: Rob Herring <robh@kernel.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87cz757rdi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current renesas,rsnd is requesting #sound-dai-cells, but it is
needed in case of it is using "simple-card", but not needed in case of
"audio-graph". We will get below warning without this patch.
This patch fiup it.
${LINUX}/arch/arm64/boot/dts/renesas/r8a77950-salvator-x.dtb: sound@ec500000: '#sound-dai-cells' is a required property
From schema: ${LINUX}/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87edrl7rf4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
renesas,rsnd.yaml is possible to use ports/port/endpoint if it is using
Audio Graph Card/Card2 for sound. The schema is defined under
audio-graph-port.yaml.
rsnd driver needs "playback/capture" property under endpoint, but it is not
defined in audio-graph-port.yaml. This patch adds missing "playback/capture"
properties under endpoint.
Without this patch, we will get below warning
${LINUX}/arch/arm64/boot/dts/renesas/r8a77950-salvator-x.dtb: sound@ec500000: ports:port@0:endpoint: Unevaluated properties are not allowed ('playback', 'capture' were unexpected)
From schema: ${LINUX}/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h6wh7rfj.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4613 is possible to use Of-graph (Audio-Graph-Card) style,
but we need to indicate it. Otherwise we will get below warning.
This patch add it.
${LINUX}/arch/arm64/boot/dts/renesas/r8a77950-salvator-x.dtb: codec@10: 'port' does not match any of the regexes: '^asahi-kasei,in[1-2]-single-end$', '^asahi-kasei,out[1-6]-single-end$', 'pinctrl-[0-9]+'
From schema: ${LINUX}/Documentation/devicetree/bindings/sound/ak4613.yaml
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ilgx7rfp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-port is missing "mclk-fs" on ports/port,
it is used not only endpoint. It is already defined on simple-card.
This patch fixup it.
Without this patch, we will get below warning.
${LINUX}/arch/arm64/boot/dts/renesas/r8a77951-ulcb-kf.dtb: audio-codec@44: ports: 'mclk-fs' does not match any of the regexes: '^port@[0-9a-f]+$', 'pinctrl-[0-9]+'
From schema: ${LINUX}/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o7qp7rgp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Walking the dram->cs array was seen as accesses beyond the first array
item by the compiler. Instead, use the array index directly. This allows
for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen
with GCC 13 with -fstrict-flex-arrays:
../sound/soc/kirkwood/kirkwood-dma.c: In function
'kirkwood_dma_conf_mbus_windows.constprop':
../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=]
90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) {
| ~~^~~~~~
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Kees Cook <keescook@chromium.org>
Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The following series will enable multi-stream support for playback and capture
streams.
Currently only a single PCM can be connected to a DAI, with the multi-stream
support it is possible to connect multiple PCMs to a single DAI.
To achieve this we need to make sure that DAIs/AIF are only set up once since
other stream could be connected to it later.
We also need to introduce reference or use counting for widgets to make sure
that they are not going to be destroyed while other streams are still using
them.
With the multi-stream support we also need to extend our current locking scheme
which worked well for simple paths.
Merge series from Astrid Rost <astrid.rost@axis.com>:
Add a generic way to create jack inputs for auxiliary jack detection
drivers (e.g. via i2c, spi), which are not part of any real codec.
The simple-card can be used as combining card driver to add the jacks,
no new one is required.
Create a jack (for input-events) for jack devices in the auxiliary
device list (aux_devs). A device which returns a valid value on
get_jack_type counts as jack device; set_jack is required
to add the jack to the device.
Add a generic way to create jack inputs for auxiliary jack detection
drivers (e.g. via i2c, spi), which are not part of any real codec.
The simple-card can be used as combining card driver to add the jacks,
no new one is required.
Create a jack (for input-events) for jack devices in the auxiliary
device list (aux_devs). A device which returns a valid value on
get_jack_type counts as jack device; set_jack is required
to add the jack to the device.
Signed-off-by: Astrid Rost <astrid.rost@axis.com>
Link: https://lore.kernel.org/r/20230123135913.2720991-3-astrid.rost@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
