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The get_stream_position has been replaced by get_dai_frame_counter, it
should not be set to allow it to be dropped from core code.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch to the new callback to retrieve the DAI (link) frame counter.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-9-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add implementation for reading the LDP (Linear DMA Position) to be used as
get_host_byte_counter().
The LDP is counting the number of bytes moved between the DSP and host
memory.
Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting
the frames on the link side of the DSP.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For delay calculation we need two information:
Number of bytes transferred between the DSP and host memory (ALSA buffer)
Number of frames transferred between the DSP and external device
(link/codec/DMIC/etc).
The reason for the different units (bytes vs frames) on host and dai side
is that the format on the dai side is decided by the firmware and might
not be the same as on the host side, thus the expectation is that the
counter reflects the number of frames.
The kernel know the host side format and in there we have access to the
DMA position which is in bytes.
In a simplified way, the DSP caused delay is the difference between the
two counters.
The existing get_stream_position callback is defined to retrieve the frame
counter on the DAI side but it's name is too generic to be intuitive and
makes it hard to define a callback for the host side.
This patch introduces a new set of callbacks to replace the
get_stream_position and define the host side equivalent:
get_dai_frame_counter
get_host_byte_counter
Subsequent patches will remove the old callback.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related
defines since it can only work on platforms which have 19 streams because
of the use of 0x948 as base offset for the LLP registers.
The generic hda_dsp_get_stream_hda_link_position() takes the number of
streams into consideration when reading the LLP registers for the stream
and can handle different HDA configurations.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the Linear Link Position is not available in firmware SRAM window we
use the host accessible position registers to read it.
The address of the PPLCLLPL/U registers depend on the number of streams
(playback+capture).
At probe time the pplc_addr is calculated for each stream and we can use
it to read the LLP without the need of address re-calculation.
Set the get_stream_position callback in sof_hda_common_ops for all
platforms:
The callback is used for IPC4 delay calculations only but the register is
a generic HDA register, not tied to any specific IPC version.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the PCM have the dsp_max_burst_size_in_ms set then place a constraint
to limit the minimum buffer time to avoid xruns caused by DMA bursts
spinning on the ALSA buffer.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When setting up the pcm widget, save the DSP buffer size (in ms) for
platform code to place a constraint on playback.
On playback the DMA will fill the buffer on start and if the period
size is smaller it will immediately overrun.
On capture the DMA will move data in 1ms bursts.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The dsp_max_burst_size_in_ms can be used to save the length of the maximum
burst size in ms the host DMA will use.
Platform code can place constraint using this to avoid user space
requesting too small ALSA buffer which will result xruns.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A bunch of fixes that came in during the merge window, probably the most
substantial thing is the DPCM locking fix for compressed audio which has
been lurking for a while.
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Merge tag 'asoc-fix-v6.9-merge-window' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A bunch of fixes that came in during the merge window, probably the most
substantial thing is the DPCM locking fix for compressed audio which has
been lurking for a while.
We find mising DPCM locking inside soc_compr_set_params_fe
before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare()
which cause lockdep assert for DPCM lock not held in
__soc_pcm_hw_params() and __soc_pcm_prepare()
Signed-off-by: Shalini Manjunatha <quic_c_shalma@quicinc.com>
Link: https://msgid.link/r/d985beeafdd32316eb45f20811eb7926da7a796e.1709720380.git.quic_c_shalma@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series restores audio support on Valve's Steam Deck OLED model, which
broke after the recent introduction of ACP/PSP communication for IRAM/DRAM fence
register programming.
The signed_fw_image member of struct sof_amd_acp_desc is used to enable
signed firmware support in the driver via the acp_sof_quirk_table.
In preparation to support additional use cases of the quirk table (i.e.
adding new flags), move signed_fw_image to a new struct acp_quirk_entry
and update all references to it accordingly.
No functional changes intended.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://msgid.link/r/20240220201623.438944-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 316a784839b21b122e1761cdca54677bb19a47fa,
that enabled Yellow Carp (YC) driver for PCI revision id 0x63.
Mukunda Vijendar [1] points out that revision 0x63 is Pink
Sardine platform, not Yellow Carp. The YC driver should not
be enabled for this platform. This patch prevents the YC
driver from being incorrectly enabled.
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240313015853.3573242-3-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit ed00a6945dc32462c2d3744a3518d2316da66fcc,
which added a quirk entry to enable the Yellow Carp (YC)
driver for the Lenovo 21J2 laptop.
Although the microphone functioned with the YC driver, it
resulted in incorrect driver usage. The Lenovo 21J2 is not a
Yellow Carp platform, but a Pink Sardine platform, which
already has an upstreamed driver.
The microphone on the Lenovo 21J2 operates correctly with the
CONFIG_SND_SOC_AMD_PS flag enabled and does not require the
quirk entry. So this patch removes the quirk entry.
Thanks to Mukunda Vijendar [1] for pointing this out.
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Link: https://msgid.link/r/20240313015853.3573242-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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
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>
The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has
I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire
control and audio.
The hardware differences between L54, L56 and L57 do not affect the
driver control interface so they can all be handled by the same driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-2-rf@opensource.cirrus.com>
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>
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>
HDMI codecs which are present and functional from audio perspective lack
i915 support on drm side what results in -ENODEV during the probing
sequence. There is no reason to perform recovery procedure e.g.: reset
the HDAudio controller if this is the case.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-4-cezary.rojewski@intel.com
If i915 does not support given platform but the hardware i.e.: HDAudio
codec is still there, the codec-probing procedure will succeed for such
device but the follow up initialization will always end up with -ENODEV.
While bus could filter out address '2' which Intel's HDMI/DP codecs
always enumerate on, more robust approach is to check for i915 presence
before registering display codecs.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-3-cezary.rojewski@intel.com
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>
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>
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>
