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: 081068fd64 ("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: e9a3b57efd ("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>
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>
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We found that using 0x45 as the default value for interrupt-clk
would cause a headset detection error.So we took 0x00 as the default
value for interrupt-clk and passed the test.
We removed mic1-src and mic2-src, which were not used.
The kerneldoc for struct cs_dsp refers to a fw_file_name member but
there's no such member.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240307105516.40250-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize debugfs_root to -ENODEV so that if the client never sets a
valid debugfs root the debugfs files will not be created.
A NULL pointer passed to any of the debugfs_create_*() functions means
"create in the root of debugfs". It doesn't mean "ignore".
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240307105353.40067-1-rf@opensource.cirrus.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>
Convert the RT1015 Mono Class D Audio Amplifier to DT schema.
Signed-off-by: Javier García <javier.gar.tab@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240304142315.14522-1-javier.gar.tab@gmail.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>
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>
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: e1830f66f6 ("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>
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: d60e4f1e4b ("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: d60e4f1e4b ("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>
Separate the functionality of wm_adsp_event() into two exported
functions wm_adsp_start() and wm_adsp_stop().
This allows the codec driver to start and stop the DSP outside of a
DAPM widget.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_is_matching_dai() checks DAI name, which is paired function
with snd_soc_dai_name_get().
It checks dlc->dai_name and dai->name (A) or dai->driver_name (B) or
dai->component->name (C)
static int snd_soc_is_matching_dai(...)
{
...
if (strcmp(dlc->dai_name, dai->name) == 0)
~~~~~~~~~~~~~ ^^^^^^^^^(A)
if (...
strcmp(dai->driver->name, dlc->dai_name) == 0)
(B)^^^^^^^^^^^^^^^^ ~~~~~~~~~~~~~
if (...
strcmp(dlc->dai_name, dai->component->name) == 0)
~~~~~~~~~~~~~ ^^^^^^^^^^^^^^^^^^(C)
...
}
But (B) part order is different with (A) and (C) (= ^^^^ and ~~~~).
This is not a big deal, but confusable to read. Fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87wmqxjbcg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To align with AMD SoundWire manager driver license, update license as
GPL-2.0-only for Pink Sardine ACP PCI driver and corresponding child
drivers.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240222102656.631144-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
rt5640_set_dai_sysclk() is an example of that - clk_set_rate() is not
guarded by IS_ERR().
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240221152516.852353-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On some boards with multiple WSA8840/WSA8845 speakers, the reset
(shutdown) GPIO is shared between two speakers. Use the reset
controller framework and its "reset-gpio" driver to handle this case.
This allows bring-up and proper handling of all WSA884x speakers on
X1E80100-CRD board.
Cc: Bartosz Golaszewski <brgl@bgdev.pl>
Cc: Sean Anderson <sean.anderson@seco.com>
Reviewed-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240129115216.96479-7-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On newer Qualcomm platforms, like X1E80100-CRD, the WSA884x speakers
share SD_N GPIOs between two speakers, thus a coordinated assertion is
needed. Linux supports handling shared GPIO lines through "reset-gpios"
property, thus allow specifying either powerdown or reset GPIOs (these
are the same).
Cc: Bartosz Golaszewski <brgl@bgdev.pl>
Cc: Sean Anderson <sean.anderson@seco.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240129115216.96479-6-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
The avs-driver continues to be utilized on more recent Intel machines.
As TGL-based (cAVS 2.5) e.g.: RPL, inherit most of the functionality
from previous platforms:
SKL <- APL <- CNL <- ICL <- TGL
rather than putting everything into a single file, the platform-specific
bits are split into cnl/icl/tgl.c files instead. Makes the division clear
and code easier to maintain.
Layout of the patchset:
First are two changes combined together address the sound-clipping
problem, present when only one stream is running - specifically one
CAPTURE stream.
Follow up is naming-scheme adjustment for some of the existing functions
what improves code incohesiveness. As existing IPC/IRQ code operates
solely on cAVS 1.5 architecture, it needs no abstraction. The situation
changes when newer platforms come into the picture. Thus the next two
patches abstract the existing IPC/IRQ handlers so that majority of the
common code can be re-used.
The ICCMAX change stands out a bit - the AudioDSP firmware loading
procedure differs on ICL-based platforms (and onwards) and having a
separate commit makes the situation clear to the developers who are
going to support the solution from LTS perspective. For that reason
I decided not to merge it into the commit introducing the icl.c file.
Update board selection with tables specifying supported I2S
configurations. DMIC/HDAudio board selection require no update as
dmic/hdaudio machine boards are generic and not tied to any specific
codec.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For ICL+ platforms to avoid DMI/OPIO L1 entry during the base firmware
load procedure, HW recommends to set LTRP_GB to 95us and start an
additional CAPTURE stream in the background.
Once the load completes, original LTRP_GB value is restored and the
additional stream is released.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-10-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Define handlers specific to cAVS 2.5 platforms, that is TGL, ADL, RPL
and all other variants based on this very version of AudioDSP
architecture. Most operations are inherited from their predecessors with
the major difference being AudioDSP cores management - firmware handlers
that on its own so there is no need to interfere.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-9-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Define handlers specific to cAVS 2.0 platforms, that is ICL, JSL and all
other variants based on this very version of AudioDSP architecture. Most
operations are inherited from their predecessors with the major
difference being firmware-logging functionality - IPC request as well as
debug memory windows layout have changed.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-8-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
