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Merge series from Luca Ceresoli <luca.ceresoli@bootlin.com>:
This series adds a driver for the internal audio codec of the Rockchip
RK3308 SoC, along with some related patches. This codec is internally
connected to the I2S peripherals on the same chip, and it has some
peculiarities arising from that interconnection.
For proper bidirectional operation with the internal codec at any possible
combination of sampling rates, the I2S peripheral needs two clock sources
(tx and rx), while connection with an external codec commonly needs only
one.
Since v5.16 there is a driver for the I2S in
sound/soc/rockchip/rockchip_i2s_tdm.c, but in some cases it does not
configure correctly the clocks, resulting in an unnecessarily inaccurate
rate. Patch 1 fixes this.
Patches 2-4 add the codec driver along with the bindings and a new helper
macro.
Patches 5-7 add to the SoC DT file two I2S controllers (those which are
internally connected to the internal codec) and the codec itself and enable
the driver in the ARM64 defconfig.
Luca
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
---
Changes in v4:
- several cleanups in the codec probe function
- Link to v3: https://lore.kernel.org/r/20240221-rk3308-audio-codec-v3-0-dfa34abfcef6@bootlin.com
Changes in v3:
- Add the I2S clock fix patch and remove a previous fix which is now superseded
- Codec driver: fix silent playback until a given amplitude of sigital
value, seen at >= 96 kHz rate
- various other changes, listed per-patch
- Link to v2: https://lore.kernel.org/r/20231219-rk3308-audio-codec-v2-0-c70d06021946@bootlin.com
Changes in v2:
- largely rewrote the codec driver to use DAPM and lots of improvements
and cleanups
- removed the RK3308 audio card and related patches
- various other changes, listed per-patch
- Link to v1: https://lore.kernel.org/all/20220907142124.2532620-1-luca.ceresoli@bootlin.com/
---
Luca Ceresoli (7):
ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates
ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec
ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro
ASoC: codecs: Add RK3308 internal audio codec driver
arm64: defconfig: enable Rockchip RK3308 internal audio codec driver
arm64: dts: rockchip: add i2s_8ch_2 and i2s_8ch_3
arm64: dts: rockchip: add the internal audio codec
.../bindings/sound/rockchip,rk3308-codec.yaml | 98 +++
MAINTAINERS | 7 +
arch/arm64/boot/dts/rockchip/rk3308.dtsi | 56 ++
arch/arm64/configs/defconfig | 1 +
include/sound/soc.h | 12 +
sound/soc/codecs/Kconfig | 11 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rk3308_codec.c | 974 +++++++++++++++++++++
sound/soc/codecs/rk3308_codec.h | 579 ++++++++++++
sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +-------
10 files changed, 1746 insertions(+), 346 deletions(-)
---
base-commit: dfda120c512b3edca1436f770924e91b14f93a98
change-id: 20231219-rk3308-audio-codec-a5558ba8949d
Best regards,
--
Luca Ceresoli <luca.ceresoli@bootlin.com>
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ASoC: Merge up release
In order to apply additional fixes that depend on the fixes merged for
v6.8 merge up the final release.
When pcm_runtime is adding platform components it will scan all
registered components. In case of DPCM FE/BE some DAI links will
configure dummy platform. However both dummy codec and dummy platform
are using "snd-soc-dummy" as component->name. Dummy codec should be
skipped when adding platforms otherwise there'll be overflow and UBSAN
complains.
Reported-by: Zhipeng Wang <zhipeng.wang_1@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240305065606.3778642-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The sample rates set by the rockchip_i2s_tdm driver in master mode are
inaccurate up to 5% in several cases, due to the driver logic to configure
clocks and a nasty interaction with the Common Clock Framework.
To understand what happens, here is the relevant section of the clock tree
(slightly simplified), along with the names used in the driver:
vpll0 _OR_ vpll1 "mclk_root"
clk_i2s2_8ch_tx_src "mclk_parent"
clk_i2s2_8ch_tx_mux
clk_i2s2_8ch_tx "mclk" or "mclk_tx"
This is what happens when playing back e.g. at 192 kHz using
audio-graph-card (when recording the same applies, only s/tx/rx/):
0. at probe, rockchip_i2s_tdm_set_sysclk() stores the passed frequency in
i2s_tdm->mclk_tx_freq (*) which is 50176000, and that is never modified
afterwards
1. when playback is started, rockchip_i2s_tdm_hw_params() is called and
does the following two calls
2. rockchip_i2s_tdm_calibrate_mclk():
2a. selects mclk_root0 (vpll0) as a parent for mclk_parent
(mclk_tx_src), which is OK because the vpll0 rate is a good for
192000 (and sumbultiple) rates
2b. sets the mclk_root frequency based on ppm calibration computations
2c. sets mclk_tx_src to 49152000 (= 256 * 192000), which is also OK as
it is a multiple of the required bit clock
3. rockchip_i2s_tdm_set_mclk()
3a. calls clk_set_rate() to set the rate of mclk_tx (clk_i2s2_8ch_tx)
to the value of i2s_tdm->mclk_tx_freq (*), i.e. 50176000 which is
not a multiple of the sampling frequency -- this is not OK
3a1. clk_set_rate() reacts by reparenting clk_i2s2_8ch_tx_src to
vpll1 -- this is not OK because the default vpll1 rate can be
divided to get 44.1 kHz and related rates, not 192 kHz
The result is that the driver does a lot of ad-hoc decisions about clocks
and ends up in using the wrong parent at an unoptimal rate.
Step 0 is one part of the problem: unless the card driver calls set_sysclk
at each stream start, whatever rate is set in mclk_tx_freq during boot will
be taken and used until reboot. Moreover the driver does not care if its
value is not a multiple of any audio frequency.
Another part of the problem is that the whole reparenting and clock rate
setting logic is conflicting with the CCF algorithms to achieve largely the
same goal: selecting the best parent and setting the closest clock
rate. And it turns out that only calling once clk_set_rate() on
clk_i2s2_8ch_tx picks the correct vpll and sets the correct rate.
The fix is based on removing the custom logic in the driver to select the
parent and set the various clocks, and just let the Clock Framework do it
all. As a side effect, the set_sysclk() op becomes useless because we now
let the CCF compute the appropriate value for the sampling rate. It also
implies that the whole calibration logic is now dead code and so it is
removed along with the "PCM Clock Compensation in PPM" kcontrol, which has
always been broken anyway. The handling of the 4 optional clocks also
becomes dead code and is removed.
The actual rates have been tested playing 30 seconds of audio at various
sampling rates before and after this change using sox:
time play -r <sample_rate> -n synth 30 sine 950 gain -3
The time reported in the table below is the 'real' value reported by the
'time' command in the above command line.
rate before after
--------- ------ ------
8000 Hz 30.60s 30.63s
11025 Hz 30.45s 30.51s
16000 Hz 30.47s 30.50s
22050 Hz 30.78s 30.41s
32000 Hz 31.02s 30.43s
44100 Hz 30.78s 30.41s
48000 Hz 29.81s 30.45s
88200 Hz 30.78s 30.41s
96000 Hz 29.79s 30.42s
176400 Hz 27.40s 30.41s
192000 Hz 29.79s 30.42s
While the tests are running the clock tree confirms that:
* without the patch, vpll1 is always used and clk_i2s2_8ch_tx always
produces 50176000 Hz, which cannot be divided for most audio rates
except the slowest ones, generating inaccurate rates
* with the patch:
- for 192000 Hz vpll0 is used
- for 176400 Hz vpll1 is used
- clk_i2s2_8ch_tx always produces (256 * <rate>) Hz
Tested on the RK3308 using the internal audio codec.
Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-1-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using __exit for the remove function results in the remove callback
being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets
unbound (e.g. using sysfs or hotplug), the driver is just removed
without the cleanup being performed. This results in resource leaks. Fix
it by compiling in the remove callback unconditionally.
This also fixes a W=1 modpost warning:
WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text)
(which only happens with SND_SOC_TLV320ADC3XXX=m).
Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver")
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://msgid.link/r/20240310143852.397212-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Some more driver specific fixes for v6.8, plus one new x86 platform
quirk. All good fixes to have if you have systems that use the relevant
hardware.
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Merge tag 'asoc-fix-v6.8-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
Some more driver specific fixes for v6.8, plus one new x86 platform
quirk. All good fixes to have if you have systems that use the relevant
hardware.
PM constants for PCI devices are defined with bitwise annotation.
When used as is, sparse complains about that:
.../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer
.../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer
Force them to be u32 in the driver.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240307163734.3852754-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't use mic1_src and mic2_src.so we delete these two members.
We changed the default value of interrupt-clk for headphone detection
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240307051222.24010-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
intel-mid.h is providing some core parts of the South Complex PM,
which are usually are not used by individual drivers. In particular,
this driver doesn't use it, so simply remove the unused header.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240305160723.1363534-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP EliteBook using ALC236 codec which using 0x02 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240304134033.773348-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name
for dlc (snd_soc_dai_link_component). In this function call
dlc->dai_name is parsed via snd_soc_dai_name_get() (B).
(A) int snd_soc_get_dlc(...)
{
...
(B) dlc->dai_name = snd_soc_dai_name_get(dai);
...
}
(B) has a priority to return dai->name as dlc->dai_name. In most cases
card can probe successfully. However it has an issue that ASoC tries to
rebind card. Here is a simplified flow for example:
| a) Card probes successfully at first
| b) One of the component bound to this card is removed for some
| reason the component->dev is released
| c) That component is re-registered
v d) ASoC calls snd_soc_try_rebind_card()
a) points dlc->dai_name to dai->name. b) releases all resource of the
old DAI. c) creates new DAI structure. In result d) can not use
dlc->dai_name to add new created DAI.
So it's reasonable that prefer to return dai->driver->name in
snd_soc_dai_name_get() because dai->driver is a pre-defined global
variable. Also update snd_soc_is_matching_dai() for alignment.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a KUnit test for the cs-amp-lib library. This has test cases
for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs().
A KUNIT_STATIC_STUB_REDIRECT() has been added to
cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the
KUnit test can redirect these to test harness functions.
Much of the testing involves invoking the same function with different
parameters, i.e. the number of amps and the amp index within the array.
This uses parameterization rather than looping. The idea is to avoid
looping over configurations within one test case as that has a higher
chance of having a bug that doesn't actually test all the expected cases.
Having the test run exactly one configuration, and then tear-down, is less
prone to accidentally skipped configurations.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240304143705.26362-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function.
Signed-off-by: Al Raj Hassain <alrajhassain@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Timing select registers for SRC and CMD are by default
referring to the corresponding SSI word select.
The calculation rule from HW spec skips SSI8, which has
no clock connection.
>From section 43.2.18 CMD Output Timing Select Register (CMDOUT_TIMSEL),
of R-Car Series, 3rd Generation Hardware User’s Manual Rev.2.20:
CMD0_OUT_DIVCLK_ Output Timing
SEL [4:0] Signal Select
B'0 0110: ssi_ws0
B'0 0111: ssi_ws1
B'0 1000: ssi_ws2
B'0 1001: ssi_ws3
B'0 1010: ssi_ws4
B'0 1011: ssi_ws5
B'0 1100: ssi_ws6
B'0 1101: ssi_ws7
<GAP>
B'0 1110: ssi_ws9
B'0 1111: Setting prohibited
Fix the erroneous prohibited setting of timsel value 1111 (0xf) for SSI9
by using timsel value 1110 (0xe) instead. This is possible because SSI8
is not connected as shown by <GAP> in the table above.
[21.695055] rcar_sound ec500000.sound: b adg[0]-CMDOUT_TIMSEL (32):00000f00/00000f1f
Correct the timsel assignment.
Fixes: 629509c5bc478c ("ASoC: rsnd: add Gen2 SRC and DMAEngine support")
Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Andreas Pape <Andreas.Pape4@bosch.com>
Signed-off-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
Tested-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
[erosca: massage commit description]
Signed-off-by: Eugeniu Rosca <eugeniu.rosca@bosch.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/20240301085003.3057-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Whilst this laptop contains _DSD inside the BIOS, there is an error in
this configuration. Override the _DSD in the BIOS with the correct
configuration for this laptop.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In azx_probe_codecs function, when bus->codec_mask is becomes to 0(no codecs),
execute azx_init_chip, bus->codec_mask will be initialized to a value again,
this causes snd_hda_codec_new function to run, the process is as follows:
-->snd_hda_codec_new
-->snd_hda_codec_device_init
-->snd_hdac_device_init---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_SUBSYSTEM_ID) 2s
---snd_hdac_read_parm(...AC_PAR_REV_ID) 2s
---snd_hdac_read_parm(...AC_PAR_NODE_COUNT) 2s
when no codecs, read communication is error, each command will be polled for
2 second, a total of 10s, it is easy to some problem.
like this:
2 [ 14.833404][ 6] [ T164] hda 0006:00: Codec #0 probe error; disabling it...
3 [ 14.844178][ 6] [ T164] hda 0006:00: codec_mask = 0x1
4 [ 14.880532][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
5 [ 15.891988][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
6 [ 16.978090][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0001
7 [ 18.140895][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0002
8 [ 19.135516][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0004
10 [ 19.900086][ 6] [ T164] hda 0006:00: no codecs initialized
11 [ 45.573398][ 2] [ C2] watchdog: BUG: soft lockup - CPU#2 stuck for 22s! [kworker/2:0:25]
Here, when bus->codec_mask is 0, use a direct break to avoid execute snd_hda_codec_new function.
Signed-off-by: songxiebing <songxiebing@kylinos.cn>
Link: https://lore.kernel.org/r/20240301011841.7247-1-soxiebing@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset Mic will no show at resume back.
This patch will fix this issue.
Fixes: d7f32791a9fc ("ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4713d48a372e47f98bba0c6120fd8254@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bios version can differ depending if it is a dual-boot variant of the tablet.
Therefore another DMI match is required.
Signed-off-by: Alban Boyé <alban.boye@protonmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix a typo in the shift value used in madera_set_fll_clks.
Fixes: 3863857dd5ca3 ("ASoC: madera: Enable clocks for input pins when used for the FLL")
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240229114637.352098-1-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few small fixes, some driver specific and one slightly larger one
from Richard which adds a new core helper and updates a small clutch of
drivers to deal with the fact that they were using a helper which
requires that the lock for the list of controls without holding that
lock. We also have some quirks for new AMD based Lenovo systems.
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Merge tag 'asoc-fix-v6.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
A few small fixes, some driver specific and one slightly larger one
from Richard which adds a new core helper and updates a small clutch of
drivers to deal with the fact that they were using a helper which
requires that the lock for the list of controls without holding that
lock. We also have some quirks for new AMD based Lenovo systems.
Like many other models, the Lenovo 21J2 (ThinkBook 16 G5+ APO)
needs a quirk entry for the internal microphone to function.
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240228073914.232204-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo 21J2 (ThinkBook 16 G5+ APO) has this new variant,
as detected with lspci:
64:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD]
ACP/ACP3X/ACP6x Audio Coprocessor (rev 63)
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240228073914.232204-1-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
Cast u8 values to u32 when using them to build a 32-bit unsigned value
that is then stored in a u64. This avoids the possibility of a bad sign
extension where the u8 is implicitly extended to an int, thus changing it
from an unsigned to a signed value.
Whether this is a real problem is debatable, but it does no harm to
ensure that the u8 are cast to a suitable type for shifting.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration")
Link: https://msgid.link/r/20240227100042.99-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It had pop noise from Headphone port when system reboot state.
If NID 58h Index 0x0 to fill default value, it will reduce pop noise.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/7493e207919a4fb3a0599324fd010e3e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Jerome Brunet <jbrunet@baylibre.com>:
This are various fixes and clean up gathered while working on Amlogic audio
support. These help better handle higher and unusual clock configuration
for TDM, SPDIF or PDM.
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
The patchset may not cover all codecs found in the codecs/ directory -
noticed a possible improvement and grepped for similar pattern across C
files found in the directory. Those addressed here seem pretty
straightforward.
Most of clk_xxx() functions do check if provided clk-pointer is
non-NULL. These do not check if the pointer is an error-pointer.
Providing such to a clk_xxx() results in a panic.
By utilizing _optional() variant of devm_clk_get() the driver code is
both simplified and more robust. There is no need to remember about
IS_ERR(clk) checks each time mclk is accessed.
The rate of the stream does not matter for the fifos of the axg family.
Fifos will just push or pull data to/from the DDR according to consumption
or production of the downstream element, which is the DPCM backend.
Drop the rate list and allow continuous rates. The lower and upper rate are
set according what is known to work with the different backends
This allows the PDM input backend to also use continuous rates.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-6-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use maximum width between 2 edges to setup spdifin thresholds
and detect the input sample rate. This comes from Amlogic SDK and
seems to be marginally more reliable than minimum width.
This is done to align with a future eARC support.
No issue was reported with minimum width so far, this is considered
to be an update so no Fixes tag is set.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC stopped using CBS_CFS and CBM_CFM a few years ago but the traces in
the amlogic tdm interface driver did not follow.
Update this to match the new format names
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.
The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.
Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.
For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.
Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
By default, when mclk-fs is not provided, the tdm-interface driver
requests an MCLK that is 4x the bit clock, SCLK.
However there is no justification for this:
* If the codec needs MCLK for its operation, mclk-fs is expected to be set
according to the codec requirements.
* If the codec does not need MCLK the minimum is 2 * SCLK, because this is
minimum the divider between SCLK and MCLK can do.
Multiplying by 4 may cause problems because the PLL limit may be reached
sooner than it should, so use 2x instead.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Starting with Qualcomm SM8350 SoC, so Low Power Audio SubSystem (LPASS)
block version v9.2, the register responsible for TX SMIC MUXn muxes is
different. In earlier LPASS versions this mux had bit fields for
analogue (ADCn) and digital (SWR_DMICn) MICs. Choice of ADCn was
selecting the analogue path in CDC_TX_TOP_CSR_SWR_DMICn_CTL register.
With LPASS v9.2 and newer, the bit fields are integrated into just
SWR_MICn and there is no distinction for analogue or digital MIC in the
register.
Fix support for LPASS v9.2+:
1. Add new set of widgets and audio routes for LPASS v9.2.
2. Do not choose analogue or digital in CDC_TX_TOP_CSR_SWR_DMICn_CTL
based on value of the mux.
3. Replace all the input widgets (TX SWR_ADCn, TX SWR_DMICn) with TX
SWR_INPUTn ones.
The change is not backwards compatible with older DTBs and existing
mixer settings, therefore it does not change handling of older platforms
with working micrphones (SC8280xp) but only the ones with issues
(SM8450, SM8550) which need the fix.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240226115925.53953-3-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
TX macro codec differs slightly between different Qualcomm Low Power
Audio SubSystem (LPASS) block versions. In LPASS version 9.2 the
register responsible for TX SMIC MUXn muxes is different, thus to
properly support it, the driver needs to register different widgets per
different LPASS version.
Prepare for supporting this register difference by refactoring existing
code:
1. Move few widgets (TX SMIC MUXn, TX SWR_ADCn, TX SWR_DMICn) out of
common 'tx_macro_dapm_widgets[]' array to a new per-variant specific
array 'tx_macro_dapm_widgets_v9[]'.
2. Move also related audio routes into new array.
3. Store pointers to these variant-specific arrays in new variant-data
structure 'tx_macro_data'.
4. Add variant-specific widgets and routes in component probe, instead
of driver probe.
The change should have no real impact, except re-shuffling code and
registering some widgets and audio routes in component probe, instead of
driver probe.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240226115925.53953-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
Factory calibration of the speakers stores the calibration information
into an EFI variable.
This set of patches adds support for applying speaker calibration
data from that EFI variable.
The HDA patch (#5) depends on the ASoC patches #2 and #3
If there are factory calibration settings in EFI, extract the
settings and write them to the firmware calibration controls.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If there are factory calibration settings in EFI, extract the
settings and write them to the firmware calibration controls.
This must be done after any firmware or coefficients have been
downloaded to the amp.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds some helper functions and data for applying amp calibration.
1. cs35l56_read_silicon_uid() to get the silicon ID that is used to
search for the correct calibration data entry.
2. Add the registers for the silicon ID to the readable registers.
3. cs35l56_get_calibration() wrapper around
cs_amp_get_efi_calibration_data()
4. cs35l56_calibration_controls() table of the firmware controls
for calibration data.
5. Added members to struct cs35l56_base to store the calibration
data.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Create a new library for code that is used by multiple Cirrus Logic
amps. This initially implements extracting amp calibration data
from EFI and writing it to firmware controls.
During factory calibration of built-in speakers the firmware
calibration constants are stored in an EFI file. The file contains
an array of calibration constants for each of the speakers.
cs_amp_get_calibration_data() searches for an entry matching the
requested UID stamp, otherwise by array index. If the data is found in
EFI the constants for that speaker are copied back to the caller.
If EFI is not enabled, the cs_amp_get_calibration_data() implementation
will compile to simply return -ENOENT and the linker can drop the code.
The code to write calibration controls uses cs_dsp. Building of cs_dsp
is not forced. Instead, the code will compile away the calls to
cs_dsp if cs_dsp is not reachable.
This strategy of conditional code allows cs-amp-lib to be shared by
multiple drivers without forcing inclusion of other modules that might
be unnecessary.
The calls to efi.get_variable() and cs_dsp are in small wrapper
functions. This is so that a KUNIT_STATIC_STUB_REDIRECT can be added in
a future patch to redirect these calls to replacement functions for
KUnit testing.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
