commit 9d4f9f6a7b upstream.
On the Renesas RZ/G3S SMARC Carrier II board having a DA7212 codec (using
da7213 driver) connected to one SSIF-2 available on the Renesas RZ/G3S SoC
it has been discovered that using the runtime PM API for suspend/resume
(as will be proposed in the following commits) leads to the codec not
being propertly initialized after resume. This is because w/o
max_register populated to regmap_config the regcache_rbtree_sync()
breaks on base_reg > max condition and the regcache_sync_block() call is
skipped.
Fixes: ef5c2eba24 ("ASoC: codecs: Add da7213 codec")
Cc: stable@vger.kernel.org
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20241106081826.1211088-23-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1157733344 upstream.
An atomicity violation occurs when the validity of the variables
da7219->clk_src and da7219->mclk_rate is being assessed. Since the entire
assessment is not protected by a lock, the da7219 variable might still be
in flux during the assessment, rendering this check invalid.
To fix this issue, we recommend adding a lock before the block
if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) so that
the legitimacy check for da7219->clk_src and da7219->mclk_rate is
protected by the lock, ensuring the validity of the check.
This possible bug is found by an experimental static analysis tool
developed by our team. This tool analyzes the locking APIs
to extract function pairs that can be concurrently executed, and then
analyzes the instructions in the paired functions to identify possible
concurrency bugs including data races and atomicity violations.
Fixes: 6d817c0e9f ("ASoC: codecs: Add da7219 codec driver")
Cc: stable@vger.kernel.org
Signed-off-by: Qiu-ji Chen <chenqiuji666@gmail.com>
Link: https://patch.msgid.link/20240930101216.23723-1-chenqiuji666@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 2f2020327c ]
Following commit 13f58267cd ("ASoC: soc.h: don't create dummy
Component via COMP_DUMMY()"), COMP_DUMMY() became an array with zero
length, and only gets populated with the dummy struct after the card is
registered. Since the sound card driver's probe happens before the card
registration, accessing any of the members of a dummy component during
probe will result in undefined behavior.
This can be observed in the mt8188 and mt8195 machine sound drivers. By
omitting a dai link subnode in the sound card's node in the Devicetree,
the default uninitialized dummy codec is used, and when its dai_name
pointer gets passed to strcmp() it results in a null pointer dereference
and a kernel panic.
In addition to that, set_card_codec_info() in the generic helpers file,
mtk-soundcard-driver.c, will populate a dai link with a dummy codec when
a dai link node is present in DT but with no codec property.
The result is that at probe time, a dummy codec can either be
uninitialized with num_codecs = 0, or be an initialized dummy codec,
with num_codecs = 1 and dai_name = "snd-soc-dummy-dai". In order to
accommodate for both situations, check that num_codecs is not zero
before accessing the codecs' fields but still check for the codec's dai
name against "snd-soc-dummy-dai" as needed.
While at it, also drop the check that dai_name is not null in the mt8192
driver, introduced in commit 4d4e1b6319 ("ASoC: mediatek: mt8192:
Check existence of dai_name before dereferencing"), as it is actually
redundant given the preceding num_codecs != 0 check.
Fixes: 13f58267cd ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()")
Signed-off-by: Nícolas F. R. A. Prado <nfraprado@collabora.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Fei Shao <fshao@chromium.org>
Acked-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://patch.msgid.link/20241126-asoc-mtk-dummy-panic-v1-1-42d53e168d2e@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e038f43eda ]
devm_kasprintf() can return a NULL pointer on failure,but this
returned value in imx_audmix_probe() is not checked.
Add NULL check in imx_audmix_probe(), to handle kernel NULL
pointer dereference error.
Fixes: 05d996e113 ("ASoC: imx-audmix: Split capture device for audmix")
Signed-off-by: Charles Han <hanchunchao@inspur.com>
Link: https://patch.msgid.link/20241118084553.4195-1-hanchunchao@inspur.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7ce8e4d380 ]
Multi link aggregation is not supported for acp6.3 platform.
Below combinations are supported.
- one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI
- one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs
As Single cpu dai is going to be created, In create_sdw_dailink() function
cpu dai index won't be incremented.
Refactor cpu dai index logic to fix below smatch static checker warning.
sound/soc/amd/acp/acp-sdw-sof-mach.c:157 create_sdw_dailink()
warn: iterator 'i' not incremented.
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Closes: https://lore.kernel.org/all/a201e871-375e-43eb-960d-5c048956c2ff@amd.com/T/
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20241007085321.3991149-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cfd1054c65 ]
The DT bindings deprecated the "hp-det-gpio" and "mic-det-gpio"
properties in favor of "hp-det-gpios" and "mic-det-gpios", but the
driver was never updated to support the latter.
Even before, there existed users of "hp-det-gpios" and "mic-det-gpios".
While this may have been handled fine by the ASoC core, this was missed
by the Freescale-specific part.
Fixes: 4189b54220 ("ASoC: dt-bindings: fsl-asoc-card: convert to YAML")
Fixes: 40ba2eda0a ("arm64: dts: imx8mm-nitrogen-r2: add audio")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://patch.msgid.link/dbcb5bfea005a468ec6dc38374fe6d02bc693c22.1727438777.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 06df673d20 ]
The last parameter 1 means BIT(0), which should be the
correct BIT(X).
Fixes: 47a70e6fc9 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://patch.msgid.link/1727424031-19551-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The max9768 has a custom control for playback mute which unconditionally
returns 0 from the put() operation, rather than returning 1 on change to
ensure notifications are generated to userspace. Check to see if the value
has changed and return appropriately.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20241112-asoc-max9768-event-v1-1-ba5d50599787@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds a quirk to include the codec amplifier function for this
Dell SKU.
Note: In this SKU '0CF1', the RT722 codec amplifier is
excluded, and an external amplifier is used instead.
Signed-off-by: Deep Harsora <deep_harsora@dell.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20241111070618.5414-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio graph card doesn't mark its subnodes such as multi {}, dpcm {}
and c2c {} as not requiring any suppliers. This causes a hang as Linux
waits for these phantom suppliers to show up on boot.
Make it clear these nodes have no suppliers.
Example error message:
[ 15.208558] platform 2034000.i2s: deferred probe pending: platform: wait for supplier /sound/multi
[ 15.208584] platform sound: deferred probe pending: asoc-audio-graph-card2: parse error
Signed-off-by: John Watts <contact@jookia.org>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20241108-graph_dt_fix-v1-1-173e2f9603d6@jookia.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Write the size of the optional payload of SOF_IPC4_MOD_INIT_INSTANCE
message to extension param_size-bits.
The previous IPC4 version does not set these bits that should indicate
the size of the optional payload (struct sof_ipc4_probe_cfg). The old
firmware side component code works well without these bits, but when
the probes are converted to use the generic module API, this does not
work anymore.
Fixes: f562359306 ("ASoC: SOF: IPC4: probes: Implement IPC4 ops for probes client device")
Signed-off-by: Jyri Sarha <jyri.sarha@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20241107132840.17386-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch checks if div is less than or equal to zero (div <= 0). If
div is zero or negative, the function returns -EINVAL, ensuring the
division operation is safe to perform.
Signed-off-by: Luo Yifan <luoyifan@cmss.chinamobile.com>
Reviewed-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20241107015936.211902-1-luoyifan@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch checks if div is less than or equal to zero (div <= 0). If
div is zero or negative, the function returns -EINVAL, ensuring the
division operation (*prate / div) is safe to perform.
Signed-off-by: Luo Yifan <luoyifan@cmss.chinamobile.com>
Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Another model of Thinkpad E14 Gen 6 (21M4)
needs a quirk entry for the dmic to be detected.
Signed-off-by: Markus Petri <mp@mpetri.org>
Link: https://patch.msgid.link/20241107094020.1050935-1-mp@localhost
Signed-off-by: Mark Brown <broonie@kernel.org>
Xiaomi Book Pro 14 2022 (MIA2210-AD) requires a quirk entry for its
internal microphone to be enabled.
This is likely due to similar reasons as seen previously on Redmi Book
14/15 Pro 2022 models (since they likely came with similar firmware):
- commit dcff8b7ca9 ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022
into DMI table")
- commit c1dd6bf619 ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022
into DMI table")
A quirk would likely be needed for Xiaomi Book Pro 15 2022 models, too.
However, I do not have such device on hand so I will leave it for now.
Signed-off-by: Mingcong Bai <jeffbai@aosc.io>
Link: https://patch.msgid.link/20241106024052.15748-1-jeffbai@aosc.io
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of error when requesting ctrl_chan DMA channel, ctrl_chan is not
null. So the release of the dma channel leads to the following issue:
[ 4.879000] st,stm32-spdifrx 500d0000.audio-controller:
dma_request_slave_channel error -19
[ 4.888975] Unable to handle kernel NULL pointer dereference
at virtual address 000000000000003d
[...]
[ 5.096577] Call trace:
[ 5.099099] dma_release_channel+0x24/0x100
[ 5.103235] stm32_spdifrx_remove+0x24/0x60 [snd_soc_stm32_spdifrx]
[ 5.109494] stm32_spdifrx_probe+0x320/0x4c4 [snd_soc_stm32_spdifrx]
To avoid this issue, release channel only if the pointer is valid.
Fixes: 794df9448e ("ASoC: stm32: spdifrx: manage rebind issue")
Signed-off-by: Amelie Delaunay <amelie.delaunay@foss.st.com>
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20241105140242.527279-1-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On a system with wcd937x, rxmacro and Qualcomm audio DSP, which is pretty
common set of devices on Qualcomm platforms, and due to the order of how
DAPM widgets are powered on (they are sorted), there is a small time window
when wcd937x chip is online and expects the flow of incoming data but
rxmacro is not yet online. When wcd937x is programmed to receive data
via AUX port then its AUX PDM watchdog is enabled in
wcd937x_codec_enable_aux_pa(). If due to some reasons the rxmacro and
soundwire machinery are delayed to start streaming data, then there is
a chance for this AUX PDM watchdog to reset the wcd937x codec. Such event
is not logged as a message and only wcd937x IRQ counter is increased
however there could be a lot of other reasons for that IRQ.
There is a similar opportunity for such delay during DAPM widgets power
down sequence.
If wcd937x codec reset happens on the start of the playback, then there
will be no sound and if such reset happens at the end of a playback then
it may generate additional clicks and pops noises.
On qrb4210 RB2 board without any debugging bits the wcd937x resets are
sometimes observed at the end of a playback though not always.
With some debugging messages or with some tracing enabled the AUX PDM
watchdog resets the wcd937x codec at the start of a playback and there
is no sound output at all.
In this patch:
- TIMEOUT_SEL bit in PDM_WD_CTL2 register is set to increase the watchdog
reset delay to 100ms which eliminates the AUX PDM watchdog IRQs on
qrb4210 RB2 board completely and decreases the number of unwanted clicks
noises;
- HOLD_OFF bit postpones triggering such watchdog IRQ till wcd937x codec
reset which usually happens at the end of a playback. This allows to
actually output some sound in case of debugging.
Cc: Adam Skladowski <a39.skl@gmail.com>
Cc: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Cc: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20241022033132.787416-3-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The wcd937x supports also AUX input but the control that sets correct
soundwire port for this is missing. This control is required for audio
playback, for instance, on qrb4210 RB2 board as well as on other
SoCs.
Reported-by: Adam Skladowski <a39.skl@gmail.com>
Reported-by: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Suggested-by: Adam Skladowski <a39.skl@gmail.com>
Suggested-by: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20241022033132.787416-2-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The widgets array in the snd_soc_dapm_widget_list has a __counted_by
attribute attached to it, which points to the num_widgets variable. This
attribute is used in bounds checking, and if it is not set before the
array is filled, then the bounds sanitizer will issue a warning or a
kernel panic if CONFIG_UBSAN_TRAP is set.
This patch sets the size of the widgets list calculated with
list_for_each as the initial value for num_widgets as it is used for
allocating memory for the array. It is updated with the actual number of
added elements after the array is filled.
Signed-off-by: Aleksei Vetrov <vvvvvv@google.com>
Fixes: 80e698e2df ("ASoC: soc-dapm: Annotate struct snd_soc_dapm_widget_list with __counted_by")
Link: https://patch.msgid.link/20241028-soc-dapm-bounds-checker-fix-v1-1-262b0394e89e@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If devm_gpiod_get_optional() fails, we need to disable previously enabled
regulators, as done in the other error handling path of the function.
Also, gpiod_set_value_cansleep(, 1) needs to be called to undo a
potential gpiod_set_value_cansleep(, 0).
If the "reset" gpio is not defined, this additional call is just a no-op.
This behavior is the same as the one already in the .remove() function.
Fixes: 11b9cd748e ("ASoC: cs42l51: add reset management")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/a5e5f4b9fb03f46abd2c93ed94b5c395972ce0d1.1729975570.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Some old Bay Trail tablets which shipped with Android as factory OS
have the SST/LPE audio engine described by an ACPI device with a
HID (Hardware-ID) of LPE0F28 instead of 80860F28.
Add support for this. Note this uses a new sst_res_info for just
the LPE0F28 case because it has a different layout for the IO-mem ACPI
resources then the 80860F28.
An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet,
which has been distributed to schools in the Spanish Andalucía region.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241025090221.52198-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Vexia Edu Atla 10 tablet mostly uses the BYTCR tablet defaults,
but as happens on more models it is using IN1 instead of IN3 for
its internal mic and JD_SRC_JD2_IN4N instead of JD_SRC_JD1_IN4P
for jack-detection.
Add a DMI quirk for this to fix the internal-mic and jack-detection.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241024211615.79518-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On some x86 Bay Trail tablets which shipped with Android as factory OS,
the DSDT is so broken that the codec needs to be manually instantatiated
by the special x86-android-tablets.ko "fixup" driver for cases like this.
This means that the codec-dev cannot be retrieved through its ACPI fwnode,
add support to the bytcr_rt5640 machine driver for such manually
instantiated rt5640 i2c_clients.
An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet,
which has been distributed to schools in the Spanish Andalucía region.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241024211615.79518-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 15c7fab0e0 ("ASoC: qcom: Move Soundwire runtime stream alloc to
soundcards") moved the allocation of Soundwire stream runtime from the
Qualcomm Soundwire driver to each individual machine sound card driver,
except that it forgot to update SC7280 card.
Just like for other Qualcomm sound cards using Soundwire, the card
driver should allocate and release the runtime. Otherwise sound
playback will result in a NULL pointer dereference or other effect of
uninitialized memory accesses (which was confirmed on SDM845 having
similar issue).
Cc: stable@vger.kernel.org
Cc: Alexey Klimov <alexey.klimov@linaro.org>
Cc: Steev Klimaszewski <steev@kali.org>
Fixes: 15c7fab0e0 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards")
Link: https://lore.kernel.org/r/20241010054109.16938-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20241012101108.129476-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On some platforms, for example i.MX93, there is only one
audio PLL source, so some sample rate can't be supported.
If the PLL source is used for 8kHz series rates, then 11kHz
series rates can't be supported.
So add constraints according to the frequency of available
clock sources, then alsa-lib will help to convert the
unsupported rate for the driver.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1728884313-6778-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
Clean up the link DMA for playback during stop for IPC4 is required to
reset the DMA read/write pointers when the stream is prepared and
restarted after a call to snd_pcm_drain()/snd_pcm_drop().
The change is mainly on ASoC. We may go via ASoC tree with Vinod's
Acked-by tag
Ranjani Sridharan (4):
ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs
ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's
soundwire: intel_ace2x: Send PDI stream number during prepare
ASoC: SOF: Intel: hda: Always clean up link DMA during stop
drivers/soundwire/intel_ace2x.c | 19 +++++-----------
sound/soc/sof/intel/hda-dai-ops.c | 23 +++++++++----------
sound/soc/sof/intel/hda-dai.c | 37 +++++++++++++++++++++++++++----
sound/soc/sof/ipc4-topology.c | 15 +++++++++++--
4 files changed, 62 insertions(+), 32 deletions(-)
--
2.43.0
On i.MX8MM the register of volume control has positive and negative
values. It is different from other platforms like i.MX8MP and i.MX93
which only have positive values. Add a volume_sx flag to use SX_TLV
volume control for this kind of platform. Use common TLV volume control
for other platforms.
Fixes: cdfa92eb90 ("ASoC: fsl_micfil: Correct the number of steps on SX controls")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://patch.msgid.link/20241017071507.2577786-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Turns out some registers of pre-2.5 version of rxmacro codecs are not
located at the expected offsets but 0xc further away in memory. So far
the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and
CDC_RX_RX2_RX_PATH_DSM_CTL.
CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address
0x540 for RX2 but it should be 0x54C and it also overwrites
CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540.
The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n).
Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version
and above this offset will be equal to 0.
With such change the corresponding RXn() macros will generate the same
values for 2.5 codec version for all RX paths and the same old values
for pre-2.5 version for RX0 and RX1. However for the latter case with
RX2 path it will also add rxn_reg_stride2 on top.
While at this, also remove specific if-check for INTERP_AUX from
rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check
was used to handle such special offset for AUX interpolator but since
CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)
macros will generate the correst addresses of dsm register, they are no
longer needed.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20241016221049.1145101-1-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new match table entry on Lunarlake for the TM2 laptops
with rt713 and rt1318.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20241016030703.13669-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is required to reset the DMA read/write pointers when the stream is
prepared and restarted after a call to snd_pcm_drain()/snd_pcm_drop().
Also, now that the stream is reset during stop, do not save LLP registers
in the case of STOP/suspend to avoid erroneous delay reporting.
Link: https://github.com/thesofproject/sof/issues/9502
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-5-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When a PCM is restarted after a snd_pcm_drain/snd_pcm_drop(), the prepare
callback will be invoked and the hw_params will be set again. For the
HDA DAI's, the hw_params function handles this case already but not for
the non-HDA DAI's. So, add the check for link_prepared to verify if the
hw_params should be done again or not. Additionally, for SDW DAI's reset
the PCMSyCM registers as would be done in the case of a start after a
hw_free.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For aggregated DAIs, the node ID is set to the group_id during the DAI
widget's ipc_prepare op. With the current logic, setting the dai_index
for node_id in the dai_config is redundant as it will be overwritten
with the group_id anyway. Removing it will also prevent any accidental
clearing/resetting of the group_id for aggregated DAIs due to the
dai_config calls could that happen before the allocated group_id is
freed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SDM845 sound card driver uses qcom_snd_sdw_startup() from the common
Soundwire module, so select it to fix build failures:
ERROR: modpost: "qcom_snd_sdw_startup" [sound/soc/qcom/snd-soc-sdm845.ko] undefined!
Fixes: d0e806b0cc ("ASoC: qcom: sdm845: add missing soundwire runtime stream alloc")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20241012100957.129103-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Irq handler need to be executed as fast as possible, so
the log in irq handler is better to use dev_dbg which needs
to be enabled when debugging.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Link: https://patch.msgid.link/1728622433-2873-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the HiHope boards, we have a single port with a single endpoint defined
as below:
....
rsnd_port: port {
rsnd_endpoint: endpoint {
remote-endpoint = <&dw_hdmi0_snd_in>;
dai-format = "i2s";
bitclock-master = <&rsnd_endpoint>;
frame-master = <&rsnd_endpoint>;
playback = <&ssi2>;
};
};
....
With commit 547b02f74e ("ASoC: rsnd: enable multi Component support for
Audio Graph Card/Card2"), support for multiple ports was added. This caused
probe failures on HiHope boards, as the endpoint could not be retrieved due
to incorrect device node pointers being used.
This patch fixes the issue by updating the `rsnd_dai_of_node()` and
`rsnd_dai_probe()` functions to use the correct device node pointers based
on the port names ('port' or 'ports'). It ensures that the endpoint is
properly parsed for both single and multi-port configurations, restoring
compatibility with HiHope boards.
Fixes: 547b02f74e ("ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2")
Signed-off-by: Lad Prabhakar <prabhakar.mahadev-lad.rj@bp.renesas.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20241010141432.716868-1-prabhakar.mahadev-lad.rj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The variable slot_found is being initialized to zero and inside
a for-loop is being checked if it's reached MAX_NUM_CH, however,
this is currently impossible since slot_found is never changed.
In a previous loop a similar coding pattern is used and slot_found
is being incremented. It appears the increment of slot_found is
missing from the loop, so fix the code by adding in the increment.
Fixes: 6a8e1d46f0 ("ASoC: max98388: add amplifier driver")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://patch.msgid.link/20241010182032.776280-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Added proper error handling for register value check that
return -EPERM when register value does not meet expected condition
Signed-off-by: Zhu Jun <zhujun2@cmss.chinamobile.com>
Link: https://patch.msgid.link/20241009073938.7472-1-zhujun2@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
