When a wmfw file has not been loaded the firmware control descriptions
necessary to write a stored calibration are not present. In this case
print a more descriptive error message.
The message is logged at info level because it is not fatal, and does
not necessarily imply that anything is broken.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240325144450.293630-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SoCs with ACE architecture are tailored to use s2idle instead deep (S3)
suspend state and the IMR content is lost when the system is forced to
enter even to S3.
When waking up from S3 state the IMR boot will fail as the content is lost.
Set the skip_imr_boot flag to make sure that we don't try IMR in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240322112504.4192-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During pause/reset or stop/start the LLP counter is not reset, which will
result broken delay reporting.
Read the LLP value on STOP/PAUSE trigger and use it in LLP reading to
normalize the LLP from the register.
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/20240321130814.4412-18-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch improves the delay calculation by relying on the
LLP (Linear Link Position) on the DAI side and the
LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA
position as LPIB, but with a linear count instead of a position in the
ALSA ring buffer. The LDP values are provided in bytes and must be
converted to frames. The difference in units means that the host counter
will wrap earlier than the LLP. We need to wrap the LLP at the same
boundary as the host counter.
The ASoC framework relies on separate pointer and delay callback.
Measurement errors can be reduced by processing all the counter values in
the pointer callback. The delay value is stored, and will be reported to
higher levels in the delay callback.
For playback, the firmware provides a stream_start offset to handle
mixing/pause usages, where the DAI might have started earlier than the
PCM device. The delay calculation must be special-cased when the link
counter has not reached the start offset value, i.e. no valid audio has
left 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/20240321130814.4412-16-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The IPC specific pointer callback can be used when additional or custom
handling is needed during the pointer calculation, like executing a delay
calculation at the same time to minimize drift between the reported pointer
and the calculated delay.
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/20240321130814.4412-15-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the
stream will be restarted (resume or start) in which case we need to update
the offset from the firmware.
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/20240321130814.4412-14-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SNDRV_PCM_TRIGGER_PAUSE_PUSH does not need to be a separate case, it
can be handled along with STOP and SUSPEND
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/20240321130814.4412-13-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The sof_ipc4_timestamp_info is only used by ipc4-pcm.c internally, it
should not be in a generic header implying that it might be used elsewhere.
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/20240321130814.4412-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The get_stream_position has been replaced by get_dai_frame_counter and all
related code can be dropped form the core.
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/20240321130814.4412-11-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
MediaTek sound card drivers are checking whether a DAI link is present
and used on a board to assign the correct parameters and this is done
by checking the codec DAI names at probe time.
If no real codec is present, assign the dummy codec to the DAI link
to avoid NULL pointer during string comparison.
Fixes: 4302187d95 ("ASoC: mediatek: common: add soundcard driver common code")
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://msgid.link/r/20240313110147.1267793-5-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Both the enumerations for UL/DL rates, delay data and the functions
adda_{dl,ul}_rate_transform were duplicated for each MediaTek SoC
dai-adda driver: move the common bits to a new mtk-dai-adda-common
file and its header.
While at it, also add the "mtk_" prefix to the exported functions.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://msgid.link/r/20240313110147.1267793-4-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify the probe function by switching error prints to return
dev_err_probe(), lowering the lines count; while at it, also
beautify some messages and change some others' level from warn
to error.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://msgid.link/r/20240313110147.1267793-3-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch from pm_runtime_enable() to devm_pm_runtime_enable(), allowing
to remove all gotos from the probe function.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://msgid.link/r/20240313110147.1267793-2-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never
selected by another driver.
So replace the extra complexity with a normal Kconfig dependency in
SND_SOC_AMD_SOUNDWIRE.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240322112018.3063344-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment we cannot instantiate two dmaengine_pcms with the same
parent device, as the components will be named the same, leading to
conflicts.
Add 'name' field to the snd_dmaengine_pcm_config, and use that (if
defined) as the component name instead of deriving the component name
from the device.
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ideasonboard.com>
Link: https://msgid.link/r/20240319-xilinx-dp-audio-v2-1-92d6d3a7ca7e@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
REG_SUPPLY mutes the DAC when switching between
HDMI and speaker, so remove it to fix the mute issues
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240320083012.4282-3-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For lower power consumption during hibernation, the configuration of
es8326_suspend and es8326_remove will be adjusted.
Adding es8326_i2c_shutdown and es8326_i2c_remove to cover different
situations
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240320083012.4282-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to support register and unregister rpmsg sound card through
remoteproc platform device for card to probe is registered in
imx-audio-rpmsg. ASoC machine driver no longer can get DT node of ASoC
CPU DAI device through parent device.
ASoC machine driver can get DT node of ASoC CPU DAI device with rpmsg
channel name acquired from platform specific data.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-6-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each rpmsg sound card sits on one rpmsg channel. Register CPU DAI with
name of rpmsg channel so that ASoC machine driver can easily link CPU
DAI with rpmsg channel name.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-5-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Let imx-audio-rpmsg register platform device for card. So that card
register and unregister can be controlled by rpmsg driver's register
and unregister.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-4-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This rpmsg driver registers device for ASoC platform driver. To align
with platform driver use rpmsg channel name to create device.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-3-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine driver uses rpmsg channel name to link this platform component.
However if the component is re-registerd card will not find this new
created component in snd_soc_try_rebind_card().
Explicitly register this component with rpmsg channel name so that
card can always find this component.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-2-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The card-name suffix and the DP-widgets are an unintended copy-paste
from skl_nau88215_ssm4567.c. Both are redundant.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A copy-paste from intel/boards/skl_nau88l25_ssm4567.c made the avs's
equivalent disable route checks as well. Such behavior is not desired.
Fixes: 69ea14efe9 ("ASoC: Intel: avs: Add ssm4567 machine board")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
One of the framework responsibilities is to ensure that the enumerated
DPCMs are valid i.e.: a valid BE is connected to a valid FE DAI. While
the are checks in soc-core.c and soc-pcm.c that verify this, a component
driver may attempt to workaround this by loading an invalid graph
through the topology file.
Be strict and fail topology loading when invalid graph is encountered.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-3-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology files that are propagated to the world and utilized by the
skylake-driver carry shortcomings in their SectionGraphs.
Since commit daa480bde6 ("ASoC: soc-core: tidyup for
snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe
failures for Intel boards have been partially addressed by commit
a22ae72b86 ("ASoC: soc-core: disable route checks for legacy devices")
and its follow up but only skl_nau88l25_ssm4567.c is patched. Fix the
problem for the rest of the boards.
Link: https://lore.kernel.org/all/20200309192744.18380-1-pierre-louis.bossart@linux.intel.com/
Fixes: daa480bde6 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-2-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the jack detection functionality in the A64 variant,
which uses a pair of IRQs; and microphone accessory (button) detection,
which uses an ADC with an IRQ trigger.
IRQs will only be triggered if the JACKDETEN, HMICBIASEN, and MICADCEN
bits are set appropriately in the analog codec component
(sun50i-codec-analog), but there is no direct software dependency
between the two components.
Setup ADC so that it samples with period of 16ms, disable smoothing
and enable MDATA threshold (should be below idle voltage/HMIC_DATA
value). Also enable HMIC_N, which makes sure we get HMIC_N samples
after HMIC_DATA crosses the threshold.
This allows us to perform steady state detection of HMIC_DATA, by
comparing current and previous ADC samples, to detect end of the
transient when the user de-presses the button. Otherwise ADC could
sample anywhere within the transient, and the driver may mis-issue
key-press events for other buttons attached to the resistor ladder.
[Ondrej: Almost complete rewrite of the patch, change to use set_jack
API. Better de-bounce, fix mic button handling, better interrupt
processing.]
Signed-off-by: Arnaud Ferraris <arnaud.ferraris@collabora.com>
[Samuel: Decouple from analog codec, fixes]
Co-developed-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Co-developed-by: Ondrej Jirman <megi@xff.cz>
Signed-off-by: Ondrej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-5-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds the necessary setup to enable jack detection on startup
as well as the callback function enabling the microphone ADC when
headset bias is enabled. The microphone ADC is also disabled in suspend.
Signed-off-by: Arnaud Ferraris <arnaud.ferraris@collabora.com>
[Samuel: Moved MICADCEN setup to HBIAS event, added bias hooks]
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-4-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
For codec variants that have a bus clock, that clock must be running to
receive interrupts. Since jack and mic accessory detection should work
even when no audio is playing, that means the bus clock should be
enabled any time the system is on.
Accomplish that by tying the bus clock to the runtime PM state, which is
then tied to the bias level not being OFF. Since the codec sets
idle_bias_on, bias will generally never be OFF. However, we can set
suspend_bias_off to maintain the power savings of gating the bus clock
during suspend, when we don't expect jack/accessory detection to work.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-3-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
With idle_bias_on and suspend_bias_off, there are bias level transitions
that match the suspend/resume callbacks. However, there are also
transitions during probe (OFF => STANDBY) and removal (STANDBY => OFF).
By using the set_bias_level hook, the driver can have one copy of code
that would otherwise be duplicated between the probe/resume and
suspend/remove hooks.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-2-megi@xff.cz
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>
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: 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>
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: 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>
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>
wm_adsp_write_ctl() must hold the pwr_lock mutex when calling
cs_dsp_get_ctl().
This was previously partially fixed by commit 781118bc2f
("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
but this only put locking around the call to cs_dsp_coeff_write_ctrl(),
missing the call to cs_dsp_get_ctl().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 781118bc2f ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
Link: https://msgid.link/r/20240307110227.41421-1-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>
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: 629509c5bc ("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: 3863857dd5 ("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: 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.
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_card_get_kcontrol() must be holding a read lock on
card->controls_rwsem while walking the controls list.
Compare with snd_ctl_find_numid().
The existing function is renamed snd_soc_card_get_kcontrol_locked()
so that it can be called from contexts that are already holding
card->controls_rwsem (for example, control get/put functions).
There are few direct or indirect callers of
snd_soc_card_get_kcontrol(), and most are safe. Three require
changes, which have been included in this patch:
codecs/cs35l45.c:
cs35l45_activate_ctl() is called from a control put() function so
is changed to call snd_soc_card_get_kcontrol_locked().
codecs/cs35l56.c:
cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from
control get()/put() functions so is changed to call
snd_soc_card_get_kcontrol_locked().
fsl/fsl_xcvr.c:
fsl_xcvr_activate_ctl() is called from three places, one of which
already holds card->controls_rwsem:
1. fsl_xcvr_mode_put(), a control put function, which will
already be holding card->controls_rwsem.
2. fsl_xcvr_startup(), a DAI startup function.
3. fsl_xcvr_shutdown(), a DAI shutdown function.
To fix this, fsl_xcvr_activate_ctl() has been changed to call
snd_soc_card_get_kcontrol_locked() so that it is safe to call
directly from fsl_xcvr_mode_put().
The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been
changed to take a read lock on card->controls_rsem() around calls
to fsl_xcvr_activate_ctl(). While this is not very elegant, it
keeps the change small, to avoid this patch creating a large
collateral churn in fsl/fsl_xcvr.c.
Analysis of other callers of snd_soc_card_get_kcontrol() is that
they do not need any changes, they are not holding card->controls_rwsem
when they call snd_soc_card_get_kcontrol().
Direct callers of snd_soc_card_get_kcontrol():
fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function
fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler
Indirect callers via soc_component_notify_control():
codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler
codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler
codecs/ak4118.c: ak4118_irq_handler() - IRQ handler
codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used
Indirect callers via snd_soc_limit_volume():
qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function
ti/rx51.c: rx51_aic34_init() - DAI init function
I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c
and ak4118.c changes.
Backport note:
The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added
since the Fixes commit so won't all be present on older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 209c6cdfd2 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card")
Link: https://lore.kernel.org/r/20240221123710.690224-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
In the case where __lpass_get_dmactl_handle is called and the driver
id dai_id is invalid the pointer dmactl is not being assigned a value,
and dmactl contains a garbage value since it has not been initialized
and so the null check may not work. Fix this to initialize dmactl to
NULL. One could argue that modern compilers will set this to zero, but
it is useful to keep this initialized as per the same way in functions
__lpass_platform_codec_intf_init and lpass_cdc_dma_daiops_hw_params.
Cleans up clang scan build warning:
sound/soc/qcom/lpass-cdc-dma.c:275:7: warning: Branch condition
evaluates to a garbage value [core.uninitialized.Branch]
Fixes: b81af585ea ("ASoC: qcom: Add lpass CPU driver for codec dma control")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://msgid.link/r/20240221134804.3475989-1-colin.i.king@gmail.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>
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>
Define handlers specific to cAVS 1.8 platforms, that is CNL, CFL, CML
and all other variants based on this very version of AudioDSP
architecture. Most operations are inherited from their predecessors.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Servicing IPCs on CNL platforms and onward differs from the existing
one. To make room for these, relocate SKL-based platforms specific code
into the skl.c file leaving only the genering irq_handler in the common
code.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Servicing IPCs on CNL platforms and onward differs from the existing
one. To make room for these, enrich platform descriptor with fields
representing crucial IPC registers and utilize them throughout the code.
While cleaning up device descriptors, reduce the number of code lines by
assigning 'min_fw_version' within a single line.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix members that are platform-specific with 'avs_' to improve code
cohesiveness and reduce the chance for naming-conflics with other
drivers.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To avoid sound clipping when there just one, single CAPTURE stream
ongoing, disable L1SEN before it is started. Any PLAYBACK stream or
additional CAPTURE allows L1SEN to be re-enabled.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Code loading is not the only procedure that manipulates L1SEN. Update
existing mechanism so the stream starting procedure can interfere with
L1SEN without causing any trouble to its other users.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pointer 'data' being initialized with a value that is never read, it
is being re-assigned inside a while-loop. The initialization is redundant
and can be removed.
Cleans up clang scan build warning
sound/soc/codecs/tas2781-fmwlib.c:1534:17: warning: Value stored to
'data' during its initialization is never read [deadcode.DeadStores]
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://msgid.link/r/20240216142219.2109050-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The soundwire-amd driver has a bit of a layering violation requiring
the SOF driver to directly call into its exported symbols rather than
through an abstraction.
The SND_SOC_SOF_AMD_SOUNDWIRE Kconfig symbol tries to deal with the
dependency by selecting SOUNDWIRE_AMD in a complicated set of conditions,
but gets it wrong for a configuration involving SND_SOC_SOF_AMD_COMMON=y,
SND_SOC_SOF_AMD_ACP63=m, and SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE=m
SOUNDWIRE_AMD=m, which results in a link failure:
ld.lld: error: undefined symbol: sdw_amd_get_slave_info
>>> referenced by acp-common.c
ld.lld: error: undefined symbol: amd_sdw_scan_controller
ld.lld: error: undefined symbol: sdw_amd_probe
ld.lld: error: undefined symbol: sdw_amd_exit
>>> referenced by acp.c
>>> sound/soc/sof/amd/acp.o:(amd_sof_acp_remove) in archive vmlinux.a
In essence, the SND_SOC_SOF_AMD_COMMON option cannot be built-in when
trying to link against a modular SOUNDWIRE_AMD driver.
Since CONFIG_SOUNDWIRE_AMD is a user-visible option, it really should
never be selected by another driver in the first place, so replace the
extra complexity with a normal Kconfig dependency in SND_SOC_SOF_AMD_SOUNDWIRE,
plus a top-level check that forbids any of the AMD SOF drivers from being
built-in with CONFIG_SOUNDWIRE_AMD=m.
In normal configs, they should all either be built-in or all loadable
modules anyway, so this simplification does not limit any real usecases.
Fixes: d948218424 ("ASoC: SOF: amd: add code for invoking soundwire manager helper functions")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://msgid.link/r/20240219093900.644574-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add machine select logic for SoundWire interface and create a machine
device node based on ACP PDM/SoundWire configuration.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240214104014.1144668-5-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Consider the below scenario, When ACP and SoundWire managers are in
D3 state and SoundWire manager power off mode is selected and acp and
SoundWire manager instances are in runtime suspended state.
In this case, for the ACP PME wake event, the ACP PCI driver should resume
SoundWire manager devices based on wake enable status set.
Add code for handling ACP PME wake event for runtime suspend scenario
when SoundWire power off mode is selected.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240214104014.1144668-4-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The earlier acp_reset flag is set to true in two instances as mentioned
below.
1. When active SoundWire manager instances power mode is set to
Power off mode when SoundWire configuration is selected.
2. For other acp configurations
As code being refactored and common function being used for scanning
SoundWire controller, acp_reset flag update logic is dropped.
Instead of it, check the SoundWire manager instance enable state, based on
it update sdw_en_stat flag which will be used to apply ACP init/de-init
sequence during suspend/resume callbacks based on flag set value when
SoundWire configuration is selected.
For other acp configurations, acp init/de-init will be called by default.
Refactor existing pm ops logic for SoundWire configuration and use
sdw_en_stat flag for invoking acp init/de-init sequence.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240214104014.1144668-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Refactor ACP child platform device creation code based on acp config.
Use common SoundWire manager functions for device probe and exit
sequences.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240214104014.1144668-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify pointer to of_phandle_args in few function arguments, for code
safety and self-documenting code.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240216145448.224185-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver must write 0 to HALO_STATE before sending the SYSTEM_RESET
command to the firmware.
HALO_STATE is in DSP memory, which is preserved across a soft reset.
The SYSTEM_RESET command does not change the value of HALO_STATE.
There is period of time while the CS35L56 is resetting, before the
firmware has started to boot, where a read of HALO_STATE will return
the value it had before the SYSTEM_RESET. If the driver does not
clear HALO_STATE, this would return BOOT_DONE status even though the
firmware has not booted.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240216140535.1434933-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull the latest 6.8 stuff into devel branch for further development.
Fixed the trivial merge conflict for HD-audio Realtek stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch complains about "head->full_size - head->header_size" can
underflow. To some extent, we're always going to have to trust the
firmware a bit. However, it's easy enough to add a check for negatives,
and let's add a upper bounds check as well.
Fixes: d2458baa79 ("ASoC: SOF: ipc3-loader: Implement firmware parsing and loading")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://msgid.link/r/5593d147-058c-4de3-a6f5-540ecb96f6f8@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Jerome Brunet <jbrunet@baylibre.com>:
This patchset fixes 2 -Wcast-function-type-strict warning in amlogic
audio drivers with clang 16.
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
For both ChainDMA and DSPless mode the requirement is that the link must
be serviced by HD-DMA.
On pre Lunar Lake platforms this was only valid for HDAudio links but with
Lunar Lake all link types now serviced by HD-DMA.
This allows us to enable ChainDMA and DSPless mode for SoundWire links as
well.
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/t9015.c:274:4: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
274 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 33901f5b9b ("ASoC: meson: add t9015 internal DAC driver")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/aiu.c:243:12: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
243 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 6ae9ca9ce9 ("ASoC: meson: aiu: add i2s and spdif support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ACPI in some SoundWire laptops has a spk-id-gpios property but
it points to the wrong Device node. This patch adds a workaround to
try to get the GPIO directly from the correct Device node.
If the attempt to get the GPIOs from the property fails, the workaround
looks for the SDCA node "AF01", which is where the GpioIo resource is
defined. If this exists, a spk-id-gpios mapping is added to that node
and then the GPIO is got from that node using the property.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240209111840.1543630-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When updating all bits in AMIC control registers (mask 0xff), use more
obvious snd_soc_component_write(). Replace also hard-coded value 0x00
with a define.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240202154134.66967-4-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Downstream driver configures DMIC clock rate through the divider
register but only parts of this code ended up in the upstream driver: we
always write the same value 0, so DIV2. Same default value is used also
for the AMIC rate control.
Let's make it obvious and drop unneeded parts of the code.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240202154134.66967-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
New versions of VA Macro has soundwire integrated, so handle the soundwire npl
clock correctly in the codec driver.
Introduce has_npl_clk and handle the sm8550 case separately because
it has soundwire integrated but doesn't have an npl clock.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://msgid.link/r/20240203-topic-sm8x50-upstream-va-macro-npl-v2-1-f2db82ae3359@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Hi,
Align the IPC4 firmware path/name and the topology path to the documentation:
default_fw_path: intel/sof-ipc4/{platform_name}
default_lib_path: intel/sof-ipc4-lib/{platform_name}
default_tplg_path: intel/sof-ipc4-tplg
default_fw_filename: sof-{platform_name}.ri
Tiger Lake and Lunar Lake support is not yet available via the official
firmware release, the paths can be changed now to avoid misalignment in the
future.
Regards,
Peter
---
Peter Ujfalusi (2):
ASoC: SOF: Intel: pci-tgl: Change the default paths and firmware names
ASoC: SOF: Intel: pci-lnl: Change the topology path to
intel/sof-ipc4-tplg
sound/soc/sof/intel/pci-lnl.c | 2 +-
sound/soc/sof/intel/pci-tgl.c | 64 +++++++++++++++++------------------
2 files changed, 33 insertions(+), 33 deletions(-)
--
2.43.0
Set ipc4_copier->data.gtw_cfg.config_length dynamically based on
blob->alh_cfg.device_count to align with the other OS.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213123007.29956-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the retain context is enabled we will unconditionally increment the
device's pm use count on each exception and when the drivers are unloaded
we do not correct this (as we don't know how many times we 'prevented
d3 entry').
Introduce a flag to make sure that we do not increment the use count more
than once and on module unload decrement the use count if needed to
balance it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240213114729.7055-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For SoundWire/ALH, we need to have a dai configured, but we don't want
to send a DMA_TLV to firmware. Add additional code branches.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-16-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When DSPless mode is selected the DMIC/SSP offload status should not be
changed since the DSP is not in use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-15-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This mode is only supported starting with LunarLake (ACE_2_0).
DMIC and SSP remain supported with the DSP only for now, since they
need a DAI configuration that is provided to firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-14-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have the dai_type we can remove any dependencies on
copiers.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-13-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Starting with LunarLake, the dspless mode can handle SoundWire/ALH,
DMIC and SSPs, so we need to identify the dai type from topology.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SoundWire integration is different from previous platforms, with
no dependencies on the DSP enablement. We can start the SoundWire
links in the probe instead of waiting for the post_fw_run stage.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-11-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For dspless mode, we need to allocate and store an 'sdai'
structure. The existing code allocate the data on the stack and does
not set the widget->private pointer.
This minor change should not have any impact on existing DAIs, even
when the DSP is used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code forces a parameter to be NULL but that parameter is
not used yet. Remove the special case in preparation for additional
changes.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-9-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have a 'is_chain_dma_supported' callback we can use it to
double-check possible disconnects between a topology file enabling
chain-dma for a DAI and the hardware/firmware capabilities.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-8-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code uses (stream_tag - 1) for the host and link dma id.
This is correct for playback, but for capture this results in an
invalid dma_type being used. The firmware assumes that the dma_id for
capture is always larger than DAI_NUM_HDA_OUT
This patch adds the offset for num_playback_streams, filled on Intel
platforms with the value extracted from the hardware capabilities.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CHAIN_DMA IPC needs the number of playback streams as a start
offset for the dma_id of a capture stream.
This offset can be retrieved on Intel platforms from the GCAP
information, and stored in the sof_ipc4_fw_data structure.
One could argue that the fields added are not really dependent on any
firmware definitions but rather on hardware capabilities, but they are
required for the IPC CHAIN_DMA definitions so adding them in
ipc4_fw_data isn't completely silly.
The CHAIN_DMA IPC is currently only functional on Intel HDaudio DMAs,
and gated by the snd_sof_is_chain_dma_supported() helper.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the existing callbacks and mix/match of HDaudio and SoundWire
support.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reuse existing function to get the interface mask and expose it to the
SOF core with a callback - the main user is the IPC4 topology so only
HDaudio platforms provide this callback.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
IPC4 introduced a 'chain-dma' mode when host and link DMA are
connected by firmware without using a regular pipeline or the ability
to add intermediate connections. This mode is not available on all
platforms and all links, so add a platform-specific callback to help
the SOF ipc4-topology core handle different hardware+firmware
configurations.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The chain_dma mode is currently only handled for HDaudio, but can be
used for orther DAIs starting with LunarLake. Move the chain_dma
handling earlier.
Error detection for the chain_dma case for older platforms is handled
at a different level.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213101247.28887-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 74ad8ed651 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
introduced a new allocation before the upper bounds check in
do_rx_work. As a result A DSP can cause bad allocations if spewing
garbage.
Fixes: 74ad8ed651 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
Reported-by: Tim Van Patten <timvp@google.com>
Cc: stable@vger.kernel.org
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213123834.4827-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the system is suspended while audio is active, the
sof_ipc4_pcm_hw_free() is invoked to reset the pipelines since during
suspend the DSP is turned off, streams will be re-started after resume.
If the firmware crashes during while audio is running (or when we reset
the stream before suspend) then the sof_ipc4_set_multi_pipeline_state()
will fail with IPC error and the state change is interrupted.
This will cause misalignment between the kernel and firmware state on next
DSP boot resulting errors returned by firmware for IPC messages, eventually
failing the audio resume.
On stream close the errors are ignored so the kernel state will be
corrected on the next DSP boot, so the second boot after the DSP panic.
If sof_ipc4_trigger_pipelines() is called from sof_ipc4_pcm_hw_free() then
state parameter is SOF_IPC4_PIPE_RESET and only in this case.
Treat a forced pipeline reset similarly to how we treat a pcm_free by
ignoring error on state sending to allow the kernel's state to be
consistent with the state the firmware will have after the next boot.
Link: https://github.com/thesofproject/sof/issues/8721
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240213115233.15716-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
clang-16 points out a mismatch in function types that was hidden
by a typecast:
sound/soc/qcom/qdsp6/q6apm-dai.c:355:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
355 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler, prtd, graph_id);
sound/soc/qcom/qdsp6/q6apm-dai.c:499:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
499 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id);
The only difference here is the 'payload' argument, which is not even
used in this function, so just fix its type and remove the cast.
Fixes: 88b60bf047 ("ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://msgid.link/r/20240213101105.459402-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The firmware release which going to introduce support for Lunar Lake will
use the documented default topology directory for IPC4:
intel/sof-ipc4-tplg
Change the default path accordingly before sof-bin (sof-firmware) release
includes Lunar Lake firmware and topologies.
Link: https://github.com/thesofproject/sof-docs/blob/master/getting_started/intel_debug/introduction.rst#2-topology-file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Mengdong Lin <mengdong.lin@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240213080418.21256-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The currently used paths and firmware name reflects the reference firmware
convention:
default_fw_path: intel/avs/{platform_name}
default_lib_path: intel/avs-lib/{platform_name}
default_tplg_path: intel/avs-tplg
default_fw_filename: dsp_basefw.bin
The SOF supports building the firmware for cAVS2.5 platforms using IPC4 and
it is the preferred IPC4 implementation to be used on these devices.
Change the paths and firmware names to reflect this:
default_fw_path: intel/sof-ipc4/{platform_name}
default_lib_path: intel/sof-ipc4-lib/{platform_name}
default_tplg_path: intel/sof-ipc4-tplg
default_fw_filename: sof-{platform_name}.ri
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@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/20240213080418.21256-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch corrects a common misspelling of "request" as "reguest" found
in error log across multiple files within sound/soc/codecs.
Signed-off-by: Yinchuan Guo <guoych37@mail2.sysu.edu.cn>
Link: https://msgid.link/r/20240212144247.43744-1-guoych37@mail2.sysu.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Like many other models, the Lenovo 82UU (Yoga Slim 7 Pro 14ARH7)
needs a quirk entry for the internal microphone to function.
Signed-off-by: Attila Tőkés <attitokes@gmail.com>
Link: https://msgid.link/r/20240210193638.144028-1-attitokes@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We describe tplg_file_name and drv_name using snd_sof_of_mach
array and select correct machine description based on dts compatible
string.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240212125258.420265-1-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MeeGoPad T8 uses the standard rt5645 jd_mode=3 setting for jack-detect,
but the used jack connector outputs an inverted jack-detect signal.
Add a DMI quirk for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20240211212736.179605-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS()
for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs.
Just a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-14-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Hans de Goede <hdegoede@redhat.com>:
While testing 6.8 on a Bay Trail device with a ALC5640 codec
I noticed a regression in 6.8 which causes a NULL pointer deref
in probe().
All BYT/CHT Intel machine drivers are affected. Patch 1/2 of
this series fixes all of them.
Patch 2/2 adds some small cleanups to cht_bsw_rt5645.c for
issues which I noticed while working on 1/2.
A recent change in acp_irq_thread() was meant to address a potential race
condition while trying to acquire the hardware semaphore responsible for
the synchronization between firmware and host IPC interrupts.
This resulted in an improper use of the IPC spinlock, causing normal
kernel memory allocations (which may sleep) inside atomic contexts:
1707255557.133976 kernel: BUG: sleeping function called from invalid context at include/linux/sched/mm.h:315
...
1707255557.134757 kernel: sof_ipc3_rx_msg+0x70/0x130 [snd_sof]
1707255557.134793 kernel: acp_sof_ipc_irq_thread+0x1e0/0x550 [snd_sof_amd_acp]
1707255557.134855 kernel: acp_irq_thread+0xa3/0x130 [snd_sof_amd_acp]
1707255557.134904 kernel: ? irq_thread+0xb5/0x1e0
1707255557.134947 kernel: ? __pfx_irq_thread_fn+0x10/0x10
1707255557.134985 kernel: irq_thread_fn+0x23/0x60
Moreover, there are attempts to lock a mutex from the same atomic
context:
1707255557.136357 kernel: =============================
1707255557.136393 kernel: [ BUG: Invalid wait context ]
1707255557.136413 kernel: 6.8.0-rc3-next-20240206-audio-next #9 Tainted: G W
1707255557.136432 kernel: -----------------------------
1707255557.136451 kernel: irq/66-AudioDSP/502 is trying to lock:
1707255557.136470 kernel: ffff965152f26af8 (&sb->s_type->i_mutex_key#2){+.+.}-{3:3}, at: start_creating.part.0+0x5f/0x180
...
1707255557.137429 kernel: start_creating.part.0+0x5f/0x180
1707255557.137457 kernel: __debugfs_create_file+0x61/0x210
1707255557.137475 kernel: snd_sof_debugfs_io_item+0x75/0xc0 [snd_sof]
1707255557.137494 kernel: sof_ipc3_do_rx_work+0x7cf/0x9f0 [snd_sof]
1707255557.137513 kernel: sof_ipc3_rx_msg+0xb3/0x130 [snd_sof]
1707255557.137532 kernel: acp_sof_ipc_irq_thread+0x1e0/0x550 [snd_sof_amd_acp]
1707255557.137551 kernel: acp_irq_thread+0xa3/0x130 [snd_sof_amd_acp]
Fix the issues by reducing the lock scope in acp_irq_thread(), so that
it guards only the hardware semaphore acquiring attempt. Additionally,
restore the initial locking in acp_sof_ipc_irq_thread() to synchronize
the handling of immediate replies from DSP core.
Fixes: 802134c8c2 ("ASoC: SOF: amd: Refactor spinlock_irq(&sdev->ipc_lock) sequence in irq_handler")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20240208234315.2182048-1-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a path in rt5645_jack_detect_work(), where rt5645->jd_mutex
is left locked forever. That may lead to deadlock
when rt5645_jack_detect_work() is called for the second time.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: cdba4301ad ("ASoC: rt5650: add mutex to avoid the jack detection failure")
Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Link: https://lore.kernel.org/r/1707645514-21196-1-git-send-email-khoroshilov@ispras.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
4 fixes / cleanups to the rt5645 mc driver's codec_name handling:
1. In the for loop looking for the dai_index for the codec, replace
card->dai_link[i] with cht_dailink[i]. The for loop already uses
ARRAY_SIZE(cht_dailink) as bound and card->dai_link is just a pointer to
cht_dailink using card->dai_link only obfuscates that cht_dailink is being
modified directly rather then say a copy of cht_dailink. Using
cht_dailink[i] also makes the code consistent with other machine drivers.
2. Don't set cht_dailink[dai_index].codecs->name in the for loop,
this immediately gets overridden using acpi_dev_name(adev) directly
below the loop.
3. Add a missing break to the loop.
4. Remove the now no longer used (only set, never read) codec_name field
from struct cht_mc_private.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20240210134400.24913-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 13f58267cd ("ASoC: soc.h: don't create dummy Component
via COMP_DUMMY()") dummy snd_soc_dai_link.codecs entries no longer
have a name set.
This means that when looking for the codec dai_link the machine
driver can no longer unconditionally run strcmp() on
snd_soc_dai_link.codecs[0].name since this may now be NULL.
Add a check for snd_soc_dai_link.codecs[0].name being NULL to all
BYT/CHT machine drivers to avoid NULL pointer dereferences in
their probe() methods.
Fixes: 13f58267cd ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()")
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20240210134400.24913-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver never uses the IRQ1_CFG register so there's no need to provide
a default value. It's set as a readable register only for debugging
through the regmap registers file.
A system-specific firmware could overwrite this register with a non-default
value. Therefore the driver can't hardcode what the initial value actually
is. As the register is only for debugging the value can be left unknown
until someone wants to read it through debugfs.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240209145700.1555950-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The soundwire links do not have their IDs as consecutive numbers, thus
the last link might have lower be_id than the previous one and this
leads to id collision with non SDW links.
For example,
create dai link SDW0-Playback-SimpleJack, id 0
create dai link SDW0-Capture-SmartMic, id 4
create dai link SDW0-Capture-SimpleJack, id 1
create dai link SDW2-Playback-SmartAmp, id 2
create dai link SDW2-Capture-SmartAmp, id 3
create dai link iDisp1, id 4
create dai link iDisp2, id 5
create dai link iDisp3, id 6
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Co-developed-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-25-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds match table for rt722 multiple
function codec on link 0.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-24-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds RT712 support for LNL.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-23-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Recent commits remove a lot of init functions remove their function
prototypes as well.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-22-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As this function is now used in sof_board_helpers it requires a build
stub for the case SSP_COMMON is not built in.
Fixes: ba0c7c3287 ("ASoC: Intel: board_helpers: support amp link initialization")
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some codec .init callbacks are empty after removing dai_links->init =
xxx_rtd_init;. Remove those callbacks.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we set sdw dai link .init callback in the codec_info_list's
dais.init function. This works fine if all codecs in the dai link are
the same. However, we need to do all the .init stuff for all different
codecs in the dai link if not all codecs in the dai link are the same.
Use a common dai link .init callback to call the new rtd_init callback
in sof_sdw_dai_info{} to do rtd_init for each dai.
Some codec init callback will become empty after this change. They will
be removed in the follow up patch.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use helper to get codec dai by name.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we assume the codecs in a dai link are all the same.
So that we get codec dai with snd_soc_rtd_to_codec(rtd, 0) in dai_links
->init callback. However, a link can include different codecs.
For example, a 4 speakers link can consist of rt712 and rt1316.
Therefore, we need to select the codec dai by name in the dai link.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2 amps can be in the same or different dai links. To handle this, the
existing code implements different spk_init functions to add dapm routes
for different amps. However, sof_sdw.c doesn't support non-aggregated
amp any more since it used pre-defined BE id.
With that assumption, combine the spk_init functions together.
This is a preparation of putting different types amps in a single dai
link.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use intel_board module to generate DAI link array and update num_links
field in snd_soc_card structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add an new field link_order_overwrite to sof_card_private structure to
support machine drivers which DAI link order is different from the
order used in sof_rt5682 (i.e. GLK boards or no-codec boards).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some dmi quirks are duplicated since codec and amplifier type are
removed from board quirk. Remove redundant quirks.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many board configs are duplicated since codec and amplifier type are
removed from board quirk. Introduce "mtl_rt5682_def" board to reduce
the number of mtl board configs.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many board configs are duplicated since codec and amplifier type are
removed from board quirk. Introduce "rpl_rt5682_def" board to reduce
the number of rpl board configs.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many board configs are duplicated since codec and amplifier type are
removed from board quirk. Introduce "adl_rt5682_def" board to reduce
the number of adl board configs.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many board configs are duplicated since codec and amplifier type are
removed from board quirk. Introduce "tgl_rt5682_def" board to reduce
the number of tgl board configs.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many board configs are duplicated since codec and amplifier type are
removed from board quirk. Introduce "jsl_rt5682_def" board to reduce
the number of jsl board configs.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240208165545.93811-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
In some cases it may be desirable to provide default initial
configuration for FW module using payload. To facilitate such solution
extend topology to contain initial config information, parse it and then
send data to FW if present.
Amadeusz Sławiński (3):
ASoC: Intel: avs: UAPI: Add tokens for initial config feature
ASoC: Intel: avs: Add topology parsing support for initial config
ASoC: Intel: avs: Send initial config to module if present
include/uapi/sound/intel/avs/tokens.h | 9 ++
sound/soc/intel/avs/path.c | 33 ++++++
sound/soc/intel/avs/topology.c | 164 +++++++++++++++++++++++++-
sound/soc/intel/avs/topology.h | 13 ++
4 files changed, 217 insertions(+), 2 deletions(-)
--
2.34.1
With the change in the widget free logic to power down the cores only
when the scheduler widgets are freed, we need to ensure that the
scheduler widget is freed only after all the widgets associated with the
scheduler are freed. This is to ensure that the secondary core that the
scheduler is scheduled to run on is kept powered on until all widgets
that need them are in use. While this works well for dynamic pipelines,
in the case of static pipelines the current logic does not take this into
account and frees all widgets in the order they occur in the
widget_list. So, modify this to ensure that the scheduler widgets are freed
only after all other types of widgets in the widget_list are freed.
Link: https://github.com/thesofproject/linux/issues/4807
Fixes: 31ed8da1c8 ("ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240208133432.1688-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rewrite the handling of ASP1 TX mixer mux initialization to prevent a
deadlock during component_remove().
The firmware can overwrite the ASP1 TX mixer registers with
system-specific settings. This is mainly for hardware that uses the
ASP as a chip-to-chip link controlled by the firmware. Because of this
the driver cannot know the starting state of the ASP1 mixer muxes until
the firmware has been downloaded and rebooted.
The original workaround for this was to queue a work function from the
dsp_work() job. This work then read the register values (populating the
regmap cache the first time around) and then called
snd_soc_dapm_mux_update_power(). The problem with this is that it was
ultimately triggered by cs35l56_component_probe() queueing dsp_work,
which meant that it would be running in parallel with the rest of the
ASoC component and card initialization. To prevent accessing DAPM before
it was fully initialized the work function took the card mutex. But this
would deadlock if cs35l56_component_remove() was called before the work job
had completed, because ASoC calls component_remove() with the card mutex
held.
This new version removes the work function. Instead the regmap cache and
DAPM mux widgets are initialized the first time any of the associated ALSA
controls is read or written.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 07f7d6e7a1 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://lore.kernel.org/r/20240208123742.1278104-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case TDM is set in topology on SSP0, parser will overwrite vindex
value, because it only checks if port is set. Fix this by checking whole
field value.
Fixes: e6d50e474e ("ASoC: Intel: avs: Improve topology parsing of dynamic strings")
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240207112624.2132821-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ should be disabled whilst entering and exiting system suspend to
avoid the IRQ handler being called whilst the PM runtime is disabled.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240206113850.719888-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
'config SND_PXA2XX_AC97' is already present in sound/arm/Kconfig with
a prompt.
Commit 734c2d4bb7 ("[ALSA] ASoC pxa2xx build support") redundantly
added the second one to sound/soc/pxa/Kconfig.
Remove it.
Signed-off-by: Masahiro Yamada <masahiroy@kernel.org>
Link: https://lore.kernel.org/r/20240204091424.38306-1-masahiroy@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_WCD939X has an optional dependency on TYPEC, so the newly added
SND_SOC_WCD939X_SDW option that selects it needs the same dependency, otherwise
it can fail randconfig builds like:
WARNING: unmet direct dependencies detected for SND_SOC_WCD939X
Depends on [m]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_WCD939X_SDW [=y] && (SOUNDWIRE [=y] || !SOUNDWIRE [=y]) && (TYPEC [=m]
|| !TYPEC [=m])
Selected by [y]:
- SND_SOC_WCD939X_SDW [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SOUNDWIRE [=y]
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_soc_codec_remove':
wcd939x.c:(.text+0x1950): undefined reference to `wcd_clsh_ctrl_free'
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_ear_dac_event':
wcd939x.c:(.text+0x35d8): undefined reference to `wcd_clsh_ctrl_set_state'
arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_enable_hphr_pa':
wcd939x.c:(.text+0x39b0): undefined reference to `wcd_clsh_ctrl_set_state'
arm-linux-gnueabi-ld: wcd939x.c:(.text+0x39dc): undefined reference to `wcd_clsh_set_hph_mode'
arm-linux-gnueabi-ld: wcd939x.c:(.text+0x3bc0): undefined reference to `wcd_clsh_ctrl_set_state'
Fixes: be2af391ce ("ASoC: codecs: Add WCD939x Soundwire devices driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://lore.kernel.org/r/20240204212207.3158914-2-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd-amd-sdw-acpi.ko module is under CONFIG_SND_SOC_AMD_ACP_COMMON but
selected from SoF, which causes build failures in some randconfig builds
that enable SOF but not ACP:
WARNING: unmet direct dependencies detected for SND_AMD_SOUNDWIRE_ACPI
Depends on [n]: SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP_COMMON [=n] && ACPI [=y]
Selected by [m]:
- SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE [=m] && SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_SOC_SOF_TOPLEVEL [=y] && SND_SOC_SOF_AMD_TOPLEVEL [=m] && ACPI [=y]
ERROR: modpost: "amd_sdw_scan_controller" [sound/soc/sof/amd/snd-sof-amd-acp.ko] undefined!
Change the Makefile and Kconfig to allow it to get built regardless
of CONFIG_SND_SOC_AMD_ACP_COMMON.
Fixes: d948218424 ("ASoC: SOF: amd: add code for invoking soundwire manager helper functions")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240204212207.3158914-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Synchronise the headphone ilimit work functions when removing the
component. These can only trigger whilst the headphone is enabled which
shouldn't be possible once the component is removed but the works rely
on the stashed component pointer so they should be shut down before the
code moves on from component remove.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240202140619.1068560-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the component is removed the stashed component pointer in the
CODECs private struct should also be cleared to prevent use of a stale
pointer.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240202140619.1068560-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The tascodec_init() of the snd-soc-tas2781-comlib module is called from
snd-soc-tas2781-i2c and snd-hda-scodec-tas2781-i2c modules. It calls
request_firmware_nowait() with parameter THIS_MODULE and a cont/callback
from the latter modules.
The latter modules can be removed while their callbacks are running,
resulting in a general protection failure.
Add module parameter to tascodec_init() so request_firmware_nowait() can
be called with the module of the callback.
Fixes: ef3bcde75d ("ASoC: tas2781: Add tas2781 driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/118dad922cef50525e5aab09badef2fa0eb796e5.1707076603.git.soyer@irl.hu
Signed-off-by: Mark Brown <broonie@kernel.org>
Recent changes modified operation-order in the probe() function without
updating its error path accordingly. If snd_hdac_i915_init() exists with
status EPROBE_DEFER the error path must cleanup allocated IRQs before
leaving the scope.
Fixes: 2dddc514b6 ("ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work.")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240202114901.1002127-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It looks like the "!" character was added accidentally. The
regmap_update_bits_check() function is normally going to succeed. This
means the rest of the function is unreachable and we don't handle the
situation where "changed" is true correctly.
Fixes: 07f7d6e7a1 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/0c254c07-d1c0-4a5c-a22b-7e135cab032c@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
These patches fixe various things that were undocumented, unknown or
uncertain when the original driver code was written. And also a few
things that were just bugs.
If the "spk-id-gpios" property is present it points to GPIOs whose
value must be used to select the correct bin file to match the
speakers.
Some manufacturers use multiple sources of speakers, which need
different tunings for best performance. On these models the type of
speaker fitted is indicated by the values of one or more GPIOs. The
number formed by the GPIOs identifies the tuning required.
The speaker ID must be used in combination with the subsystem ID
(either from PCI SSID or cirrus,firmware-uid property), because the
GPIOs can only indicate variants of a specific model.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 1a1c3d794e ("ASoC: cs35l56: Use PCI SSID as the firmware UID")
Link: https://msgid.link/r/20240129162737.497-14-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Check during initialization whether the firmware is already patched.
If so, include the firmware version in the wm_adsp fwf_name string.
If the firmware has already been patched by the BIOS the driver
can only replace it if it has control of hard RESET.
If the driver cannot replace the firmware, it can still load a wmfw
(for ALSA control definitions) and/or a bin (for additional tunings).
But these must match the version of firmware that is running on the
CS35L56.
The firmware is pre-patched if FIRMWARE_MISSING == 0.
Including the firmware version in the fwf_name string will
qualify the firmware file name:
Normal (unpatched or replaceable firmware):
cs35l56-rev-dsp1-misc[-system_name].[wmfw|bin]
Preloaded firmware:
cs35l56-rev[-s]-VVVVVV-dsp1-misc[-system_name].[wmfw|bin]
Where:
[-s] is an optional -s added into the name for a secured CS35L56
VVVVVV is the 24-bit firmware version in hexadecimal.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 608f1b0dbd ("ASoC: cs35l56: Move DSP part string generation so that it is done only once")
Link: https://msgid.link/r/20240129162737.497-13-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Put the silicon revision and secured flag in the wm_adsp fwf_name
string instead of including them in the part string.
This changes the format of the firmware name string from
cs35l56[s]-rev-misc[-system_name]
to
cs35l56-rev[-s]-misc[-system_name]
No firmware files have been published, so this doesn't cause a
compatibility break.
Silicon revision and secured flag are included in the firmware
filename to pick a firmware compatible with the part. These strings
were being added to the part string, but that is a misuse of the
string. The correct place for these is the fwf_name string, which
is specifically intended to select between multiple firmware files
for the same part.
Backport note:
This won't apply to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 608f1b0dbd ("ASoC: cs35l56: Move DSP part string generation so that it is done only once")
Link: https://msgid.link/r/20240129162737.497-12-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Defer initializing the state of the ASP1 mixer registers until
the firmware has been downloaded and rebooted.
On a SoundWire system the ASP is free for use as a chip-to-chip
interconnect. This can be either for the firmware on multiple
CS35L56 to share reference audio; or as a bridge to another
device. If it is a firmware interconnect it is owned by the
firmware and the Linux driver should avoid writing the registers.
However, if it is a bridge then Linux may take over and handle
it as a normal codec-to-codec link. Even if the ASP is used
as a firmware-firmware interconnect it is useful to have
ALSA controls for the ASP mixer. They are at least useful for
debugging.
CS35L56 is designed for SDCA and a generic SDCA driver would
know nothing about these chip-specific registers. So if the
ASP is being used on a SoundWire system the firmware sets up the
ASP mixer registers. This means that we can't assume the default
state of these registers. But we don't know the initial state
that the firmware set them to until after the firmware has been
downloaded and booted, which can take several seconds when
downloading multiple amps.
DAPM normally reads the initial state of mux registers during
probe() but this would mean blocking probe() for several seconds
until the firmware has initialized them. To avoid this, the
mixer muxes are set SND_SOC_NOPM to prevent DAPM trying to read
the register state. Custom get/set callbacks are implemented for
ALSA control access, and these can safely block waiting for the
firmware download.
After the firmware download has completed, the state of the
mux registers is known so a work job is queued to call
snd_soc_dapm_mux_update_power() on each of the mux widgets.
Backport note:
This won't apply cleanly to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-11-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Patch the SDW TX mixer registers to silicon defaults.
CS35L56 is designed for SDCA and a generic SDCA driver would
know nothing about these chip-specific registers. So the
firmware sets up the SDW TX mixer registers to whatever audio
is relevant on a specific system.
This means that the driver cannot assume the initial values
of these registers. But Linux has ALSA controls to configure
routing, so the registers can be patched to silicon default and
the ALSA controls used to select what audio to feed back to the
host capture path.
Backport note:
This won't apply to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-9-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a dummy SUPPLY widget connected to the ASP that forces the
chip registers to match the regmap cache when the ASP is
powered-up.
On a SoundWire system the ASP is free for use as a chip-to-chip
interconnect. This can be either for the firmware on multiple
CS35L56 to share reference audio; or as a bridge to another
device. If it is a firmware interconnect it is owned by the
firmware and the Linux driver should avoid writing the registers.
However. If it is a bridge then Linux may take over and handle
it as a normal codec-to-codec link.
CS35L56 is designed for SDCA and a generic SDCA driver would
know nothing about these chip-specific registers. So if the
ASP is being used on a SoundWire system the firmware sets up the
ASP registers. This means that we can't assume the default
state of the ASP registers. But we don't know the initial state
that the firmware set them to until after the firmware has been
downloaded and booted, which can take several seconds when
downloading multiple amps.
To avoid blocking probe() for several seconds waiting for the
firmware, the silicon defaults are assumed. This allows the machine
driver to setup the ASP configuration during probe() without being
blocked. If the ASP is hooked up and used, the SUPPLY widget
ensures that the chip registers match what was configured in the
regmap cache.
If the machine driver does not hook up the ASP, it is assumed that
it won't call any functions to configure the ASP DAI. Therefore
the regmap cache will be clean for these registers so a
regcache_sync() will not overwrite the chip registers. If the
DAI is not hooked up, the dummy SUPPLY widget will not be
invoked so it will never force-overwrite the chip registers.
Backport note:
This won't apply cleanly to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-8-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the check of fw_patched from cs35l56_is_fw_reload_needed().
Also remove the redundant check for control of the reset GPIO.
The fw_patched flag is set when cs35l56_dsp_work() has completed its
steps to download firmware and power-up wm_adsp. There was a check in
cs35l56_is_fw_reload_needed() to make a quick exit of 'false' if
!fw_patched. The original idea was that the system might be suspended
before the driver has ever made any attempt to download firmware, and
in that case the driver doesn't need to return to a patched state
because it was never in a patched state.
This check of fw_patched is buggy because it prevented ever recovering
from a failed patch. If a previous attempt to patch and reboot the
silicon had failed it would leave fw_patched==false. This would mean
the driver never attempted another download even though the fault may
have been cleared (by a hard reset, for example).
It is also a redundant check because the calling code already makes
a quick exit if cs35l56_component_probe() has not been called, which
deals with the original intent of this check but in a safer way.
The check for reset GPIO is redundant: if the silicon was hard-reset
the FIRMWARE_MISSING flag will be 1. But this check created an
expectation that the suspend/resume code toggles reset. This can't
easily be protected against accidental code breakage. The only reason
for the check was to skip runtime-resuming the driver to read the
PROTECTION_STATUS register when it already knows it reset the silicon.
But in that case the driver will have to be runtime-resumed to do
the firmware download. So it created an assumption for no benefit.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240129162737.497-7-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the call to cs35l56_set_patch() earlier in cs35l56_init() so
that it only adds the register patch on first-time initialization.
The call was after the post_soft_reset label, so every time this
function was run to re-initialize the hardware after a reset it would
call regmap_register_patch() and add the same reg_sequence again.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 898673b905 ("ASoC: cs35l56: Move shared data into a common data structure")
Link: https://msgid.link/r/20240129162737.497-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cs35l56_component_remove() must call wm_adsp_power_down() and
wm_adsp2_component_remove().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs35l56->component pointer is used by the suspend-resume handling to
know whether the driver is fully instantiated. This is to prevent it
queuing dsp_work which would result in calling wm_adsp when the driver
is not an instantiated ASoC component. So this pointer must be cleared
by cs35l56_component_remove().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There's no need to overwrite fwf_name with a kstrdup() of the cs_dsp part
name. It is trivial to select either fwf_name or cs_dsp.part as the string
to use when building the filename in wm_adsp_request_firmware_file().
This leaves fwf_name entirely owned by the codec driver.
It also avoids problems with freeing the pointer. With the original code
fwf_name was either a pointer owned by the codec driver, or a kstrdup()
created by wm_adsp. This meant wm_adsp must free it if it set it, but not
if the codec driver set it. The code was handling this by using
devm_kstrdup().
But there is no absolute requirement that wm_adsp_common_init() must be
called from probe(), so this was a pseudo-memory leak - each new call to
wm_adsp_common_init() would allocate another block of memory but these
would only be freed if the owning codec driver was removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240129162737.497-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Check for the cases of system-specific bin file without a
wmfw before falling back to looking for a generic wmfw.
All system-specific options should be tried before falling
back to loading a generic wmfw/bin. With the original code,
the presence of a fallback generic wmfw on the filesystem
would prevent using a system-specific tuning with a ROM
firmware.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 0e7d82cbea ("ASoC: wm_adsp: Add support for loading bin files without wmfw")
Link: https://msgid.link/r/20240129162737.497-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Vijendar Mukunda <Vijendar.Mukunda@amd.com>:
This patch series is to redesign existing platform device creation logic
for SoundWire managers and Implement generic functions for SoundWire
manager probe, start and exit sequence which are common for both Legacy
(NO DSP enabled) and SOF stack, and add SoundWire Interface support for
AMD SOF stack (ACP 6.3 based platform).
The patch series was reviewed in
https://github.com/thesofproject/linux/pull/4699
Refactor acp driver pm ops to support SoundWire interface.
When SoundWire configuration is enabled, In case of ClockStopMode,
DSP soft reset should be applied and for rest of the scenarios
acp init/deinit sequence should be invoked.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240129055147.1493853-14-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add code for invoking Soundwire manager helper functions
when SoundWire configuration is selected.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240129055147.1493853-8-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement generic function for scanning SoundWire controller.
Same function will be used for legacy and sof stack for AMD platforms.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://msgid.link/r/20240129055147.1493853-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The devm_request_irq() call is done for "dma_rt" interrupt but the error
message printed "dma_tx" interrupt on failure, fix this by updating
dma_tx -> dma_rt in dev_err_probe() message. While at it aligned the code.
Signed-off-by: Lad Prabhakar <prabhakar.mahadev-lad.rj@bp.renesas.com>
Fixes: 38c042b59a ("ASoC: sh: rz-ssi: Update interrupt handling for half duplex channels")
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://msgid.link/r/20240130150822.327434-1-prabhakar.mahadev-lad.rj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The layout is configured as:
- Link0: CS42L43 Jack and mics (2ch)
- Link2: 2x CS35L56 Speaker (amps 3 and 4, right)
- Link3: 2x CS35L56 Speaker (amps 1 and 2, left)
Corresponding SOF topology:
https://github.com/thesofproject/sof/pull/8773
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Link: https://msgid.link/r/20240123113246.75539-1-mstrozek@opensource.cirrus.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previous commit that added support for Huawei MateBook D16 2021
with Ryzen 4600H (HVY-WXX9 M1010) was incomplete.
To activate support for this laptop, the DMI table in
acp3x-es83xx machine driver must also be updated.
Fixes: b5338b1b90 ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop")
Signed-off-by: Marian Postevca <posteuca@mutex.one>
Link: https://msgid.link/r/20240128172229.657142-1-posteuca@mutex.one
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Chen-Yu Tsai <wens@kernel.org>:
This series adds SPDIF controllers for the H616 and H618.
There's also a fix for SPDIF on H6: the controller also has a
receiver that was not correctly modeled.
The SPDIF hardware block found in the H616 SoC has the same layout as
the one found in the H6 SoC, except that it is missing the receiver
side.
Since the driver currently only supports the transmit function, support
for the H616 is identical to what is currently done for the H6.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://msgid.link/r/20240127163247.384439-4-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Use fls to calculate the pre-divider and input frequency for the PLL,
this is marginally faster than the previous loop.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240125103117.2622095-7-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Refactor the code in cs42l43_mask_to_slots() to use for_each_set_bit().
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240125103117.2622095-6-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
