The DAI ID is a hardware port ID and is not necessarily the index of
the corresponding entry in variant->dai_driver.
When configuring a DAI with LPAIF_I2SCTL_MODE_QUAD01, the probe code
currently uses dai_id to index variant->dai_driver. This is incorrect
for platforms where DAI IDs are sparse. For example, the IPQ806x MI2S
DAI has ID 4 while it is the only entry in the DAI driver table. If
that DAI is configured for QUAD01 and probe reaches this branch, the
code writes beyond that single entry via dai_driver[4] instead of
updating the DAI being processed.
Use the loop index i when updating the current DAI's playback channel
constraints, while retaining dai_id for indexing the hardware-port
specific playback SD-line mode array.
This fixes the incorrect DAI table access for platforms where the DAI
ID does not match its position in the driver table.
Fixes: c223f41c1a ("ASoC: qcom: Add four speaker support on MI2S secondary")
Cc: stable@vger.kernel.org
Signed-off-by: Zimeng Li <me@lizi.moe>
Assisted-by: LLM-assisted source analysis
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260927010911.51980-1-me@lizi.moe
Signed-off-by: Mark Brown <broonie@kernel.org>
This is another run of the Coccinelle script for converting kmalloc()
family of allocations to kmalloc_obj() via the existing rules in
scripts/coccinelle/api/kmalloc_objs.cocci
This catches both the set of kmalloc() uses added since the first
kmalloc_obj() conversions in v7.0 and adds a large group missed in the
first pass due to Coccinelle not interacting well with the cleanup.h
scoped_...() family of macros[1]. I worked around this with spatch's
"--macro-file" argument to a file with all the scoped_...() macros mapped
to Coccinelle's YACFE_ITERATOR[2] as that was the closest viable control
flow indicator I could find.
Build tested allmodconfig on x86, arm64, arm, loongarch, mips, powerpc,
riscv, and s390 with no new warnings.
Link: https://lore.kernel.org/lkml/202609021314.8A9C0B8@keescook/ [1]
Link: https://github.com/coccinelle/coccinelle/blob/master/standard.h [2]
Signed-off-by: Kees Cook <kees+treewide@kernel.org>
Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com> says:
Address review findings for the recently merged AudioReach TDM backend
support.
The first patch fixes a typo in the newly added QAIF clock ID names by
renaming the LAPSS-prefixed identifiers to LPASS and updating all
corresponding in-tree users.
The remaining patches address issues identified in the AudioReach TDM
configuration and machine-driver paths:
- Zero-initialize AudioReach module configuration structures so
unspecified slot parameters correctly fall back to topology defaults.
- Distinguish between missing TDM configuration and malformed TDM slot
properties by using separate return codes, preventing valid DT
configuration errors from being silently ignored.
- Fix sc8280xp TDM hw_params error handling by properly distinguishing
optional and invalid TDM slot configurations and propagating errors
accordingly.
All issues were reported by Sashiko.
Link:
https://sashiko.dev/#/patchset/20260804070307.117119-1-prasad.kumpatla@oss.qualcomm.com
Link: https://patch.msgid.link/20260818084655.3240284-1-prasad.kumpatla@oss.qualcomm.com
Treat -ENOENT from TDM slot parsing as the optional "configuration not
present" case and continue to propagate real configuration errors.
Also ignore -ENOTSUPP from optional DAI format and codec sysclk
callbacks, apply codec_dai_fmt to codec DAIs on TDM links, and use a
signed bclk_freq variable so errors from snd_soc_tdm_params_to_bclk()
are handled correctly.
Reported-by: Sashiko <sashiko-bot@kernel.org>
Link: https://sashiko.dev/#/patchset/20260804070307.117119-1-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260818084655.3240284-5-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
q6apm_dai_prepare() and q6apm_dai_compr_set_params() build an
AudioReach module config on the stack and populate only the fields
needed for the stream type.
Audio IF media-format setup treats zero slot_mask, nslots_per_frame and
slot_width as a request to use the topology defaults. Leave those fields
uninitialized and random stack data can override the topology values sent
to the DSP.
Initialize the whole config structure so omitted fields reliably retain
their zero/default meaning.
Reported-by: Sashiko <sashiko-bot@kernel.org>
Link: https://sashiko.dev/#/patchset/20260804070307.117119-1-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260818084655.3240284-3-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
q6apm_graph_start() increments start_count even when APM_CMD_GRAPH_START
fails, leaving the graph counted as running while the DSP never started
it. A later start - a retried prepare, or a resume after a failed start -
then finds a non-zero count, skips the command and returns success with
no data flowing.
Count the graph only once the DSP has accepted the start. The count then
stays at zero for a graph that never started, so also stop decrementing
below zero in q6apm_graph_stop(): the compressed free path stops
unconditionally, and a negative count would make the next start skip the
command in the same way.
Fixes: 5477518b8a ("ASoC: qdsp6: audioreach: add q6apm support")
Assisted-by: Claude:claude-opus-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
Link: https://patch.msgid.link/20260726211226.94059-1-jorijnvdgraaf@catcrafts.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai <tiwai@suse.de> says:
here is a revised series of trivial patches to use the auto-cleanup
(via __free(firmware)) for firmware management.
Now compile warnings due to the mixture with goto have been addressed
for rt1320-sdw and wm2000 drivers. Others remain identical, just took
a few review Acks.
Link: https://patch.msgid.link/20260806140006.1412298-1-tiwai@suse.de
Simplify the code to manage the firmware loading with __free(firmware)
auto-cleanup.
Only the code refactoring, no functional changes.
Cc: Srinivas Kandagatla <srini@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260806140006.1412298-29-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Continuous high-amplitude noise could occur in the DA7213 microphone
capture path after a Bluetooth out-of-range/reset event followed by
reconnection. The noise was present in both the raw ALSA capture and
PipeWire input, confirming that it originated before Bluetooth
encoding.
The codec already obtains and enables MCLK through its DT clock and
bias-level handling. However, the machine driver did not explicitly
configure the codec sysclk during hw_params(). Enable
codec_sysclk_set for QCS615 so that the DA7213 clock source and rate
are configured before the codec power-up sequence.
Verified on QCS615 Talos with repeated Bluetooth disconnect and
reconnect cycles. The noise was no longer reproducible.
Signed-off-by: Le Qi <le.qi@oss.qualcomm.com>
Link: https://patch.msgid.link/20260810031032.2001053-1-le.qi@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add TDM backend handling to the sc8280xp machine driver.
Use the common QCOM DAI-node TDM helper to parse the standard
DAI TDM slot properties from backend CPU and codec endpoints. Reuse the
parsed configuration when programming DAIs so hw_params does not need a
second DT traversal.
Derive the LPASS backend bit clock from the runtime TDM parameters and
request it through the backend child-clock path using LPAIF_MI2S_BCLK.
Program codec sysclk in hw_params so codec PLL setup happens before the
stream is triggered.
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260804070307.117119-8-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add common helpers to parse standard dai-tdm-slot-* properties from the
CPU and codec child nodes of a backend DAI link and apply the result to
the active DAIs.
QCOM machine drivers already use qcom_snd_parse_of() to build links from
DT, but they lacked a shared helper to translate endpoint TDM properties
into snd_soc_dai_set_tdm_slot() calls. Boards therefore had to carry ad
hoc parsing or rely on non-standard DT properties.
The helpers parse endpoint masks, validate the shared slot count and
slot width, and program CPU and codec DAIs with the resulting slot
configuration. A cfg-based apply helper is provided for callers that
already parsed the DT data and want to avoid a second DT traversal.
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260804070307.117119-7-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the q6prm clock table entries and internal DSP clock IDs for LPASS
Audio IF backend clocks.
The public binding IDs map to q6prm DSP clock IDs starting at 0x500 for
Audio IF0 IBIT/EBIT. Add the internal definitions and register all Audio
IF IBIT and EBIT clocks so machine drivers can request them through the
APM clock controller.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260804070307.117119-5-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add TDM DAI operations to q6apm-lpass-dais so AudioReach TDM
backends can be configured through the normal ASoC hw_params and DAI
setup flow.
The TDM set_tdm_slot() callback validates the supported slot width and
slot count, stores the active slot mask in the AudioReach module
configuration, and leaves existing DMA, I2S and HDMI paths unchanged.
Reuse the existing LPASS child-clock handling for TDM nodes as well as
MI2S nodes, since TDM backends also request optional backend clocks
through the machine driver set_sysclk() path.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260804070307.117119-3-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add topology parsing and media-format programming for Audio IF source and
sink modules.
Add the Audio IF module IDs, the required topology tokens, and a dedicated
topology loader that stores the parsed interface configuration in the
AudioReach module state. Also add the Audio IF media-format path that
sends the interface configuration, hardware endpoint media format, and
frame-duration parameters for Audio IF modules.
This keeps the serial-interface configuration topology-driven while still
allowing the machine driver to provide runtime slot and media format
settings. The same Audio IF path can then be reused for TDM, PCM, and I2S
style backends.
The new UAPI tokens (AR_TKN_U16_MODULE_SYNC_SRC=262 through
AR_TKN_U8_MODULE_INV_EXT_BIT_CLK=276) are added, together with the value
defines used by the sync source, sync mode, data delay, interface mode,
bit clock type, and polarity tokens.
MODULE_ID_AUDIO_IF_SINK (0x0700117C) and MODULE_ID_AUDIO_IF_SOURCE
(0x0700117D) are introduced in this patch.
This Module is validated on Hawi and Shikra platforms.
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260804070307.117119-2-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> says:
This series adds sound support for the Monaco/Monza (VENTUNO-Q)
platform on the QCS8275 board, which uses a MAX98091 codec on an
LPI MI2S link.
Patch 2 (from Neil) registers the LPASS MCLK1..MCLK5 clock IDs in
q6prm-clocks.c. Without these entries the DT clock reference
"clocks = <&q6prmcc LPASS_CLK_ID_MCLK_3 ...>" that the MAX98091
node uses cannot resolve at probe, and the codec silently runs
without MCLK.
Note: this patch has been posted to the list more than once from Oct
2025 and does not appear to be actively pursued by anyone. I
am picking it up here because the rest of this series depends on
it to get sound working on QCS8275, and it makes little sense to
send the machine-driver changes without the clock IDs they rely
on.
With these patches now we have full audio working on Arduino VENTUNO-Q.
Link: https://patch.msgid.link/20260731162626.1588561-1-srinivas.kandagatla@oss.qualcomm.com
The driver-local structure was named 'snd_soc_common', which occupies
the generic snd_soc_ ASoC namespace even though it is specific to the
Qualcomm sc8280xp machine driver.
Rename the type to qcom_snd_soc_common so the identifier is properly
scoped to this driver, and rename the pointer field in
sc8280xp_snd_data from 'snd_soc_common_priv' to 'priv' so the shorter
name doesn't repeat the type name at every use.
No functional change.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260731162626.1588561-4-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing MCLK ids for the q6prm DSP interface.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260731162626.1588561-3-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Not all codecs implement the set_sysclk operation. When the board
enables codec_sysclk_set, snd_soc_dai_set_sysclk() on the codec DAI
can return -ENOTSUPP, which currently aborts hw_params and breaks
playback/capture on such boards even though the missing clock setup
is harmless.
Ignore -ENOTSUPP for the codec set_sysclk call.
Fixes: 766f3f79c3 ("ASoC: qcom: sc8280xp: enhance machine driver for board-specific config")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260731162626.1588561-2-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Richard Acayan <mailingradian@gmail.com> says:
This adds support for the SDM660 (formerly "SDM660 internal") sound card
and support for WCD codecs over internal MI2S (represented in APIs as
LPI MI2S).
Like on MSM8916 and MSM8953, some SDM660 and SDM670 devices connect to a
digital and analog codec. The connection to the digital codec is through
special "internal" MI2S ports. The digital and analog codecs are used on
the Xiaomi Redmi Note 7 for headset (playback + capture) and earpiece,
and also on the Google Pixel 3a for the headset.
This series does not include devicetree patches.
Link: https://patch.msgid.link/20260730174353.108023-1-mailingradian@gmail.com
Add the compatible for SDM660 and SDM670 devices, which can use the
support for WCD codecs connected to internal MI2S.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-13-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The LPI_MI2S_RX_0 and LPI_MI2S_TX_3 ports on SDM660 can be connected to
the digital and analog WCD codecs. They can be supported with the same
logic for other ports, but just need to be explicitly stated. Add
support for these ports.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-12-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On SDM660, LPI MI2S ports can be connected to a WCD codec which may
support headphones. Register the headphone jack on codecs connected to
the playback port, LPI_MI2S_RX_0.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-11-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the ASM-AFE routing for LPI MI2S ports which represent internal MI2S
ports on SDM660.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-10-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The bindings for LPI MI2S ports, originally exclusive to q6apm, can be
used for internal MI2S ports on q6afe. Add the internal MI2S ports found
on the SDM660 internal sound card using the LPI MI2S bindings.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-9-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The bindings for LPI MI2S ports, originally exclusive to q6apm, can be
used for internal MI2S ports on q6afe. Add the port mappings for
internal MI2S, found on the Snapdragon 660 internal sound card.
Signed-off-by: Richard Acayan <mailingradian@gmail.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730174353.108023-8-mailingradian@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com> says:
This series adds support for LPASS low-power resource (LPR) voting
through PRM and introduces LPASS LPI TLMM pinctrl support for newer
platforms such as Hawi.
On such platforms, LPASS requires LPR resource voting via PRM to keep
the subsystem active. This is handled by adding a new clock ID and
support for PARAM_ID_RSC_CPU_LPR in q6prm.
Additionally, a new LPASS LPI TLMM block is introduced, requiring a
dedicated DT binding and pinctrl driver.
Link: https://patch.msgid.link/20260724141708.2212057-1-prasad.kumpatla@oss.qualcomm.com
Add support for issuing LPASS low-power resource (LPR) votes through
the PRM interface.
Some platforms (e.g. Hawi) require the LPASS to be kept active via LPR
resource voting instead of the existing hardware core vote mechanism.
Handle this by introducing support for PARAM_ID_RSC_CPU_LPR when the
LPR vote clock ID is requested.
For LPR requests, use the appropriate parameter ID and payload format
to disable CPU subsystem sleep, ensuring that the LPASS register space
remains accessible.
Also add the corresponding clock mapping for LPASS_HW_LPR_VOTE.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Acked-by: Bartosz Golaszewski <bartosz.golaszewski@oss.qualcomm.com>
Link: https://patch.msgid.link/20260724141708.2212057-4-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Q6DSP_MAX_CLK_ID defines the upper bound of supported clock
identifiers in the qdsp6 LPASS clock driver.
Increase the maximum clock ID value to accommodate the LPASS
LPR vote clock identifier.
Signed-off-by: Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com>
Acked-by: Bartosz Golaszewski <bartosz.golaszewski@oss.qualcomm.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260724141708.2212057-3-prasad.kumpatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Neil Armstrong <neil.armstrong@linaro.org> says:
The Qualcom SM8650 based Ayaneo Pocket S2 gaming device has a set
of 2 WSA speakers connected on the WSA2 lines.
But the Audioreach DSP only handles WSA2 in pair with the WSA
interface by using the upper bits of the active_channels_mask
for WSA2 and the lower bits for WSA:
__________________________________________________
| Bits | 3 | 2 | 1 | 0 |
---------------------------------------------------
| Line | WSA2 Ch2 | WSA2 Ch1 | WSA Ch2 | WSA Ch1 |
---------------------------------------------------
Setting only the WSA2 upper bits is perfectly valid and
functional but the current Audioreach code builds the bitmask
from the channels count with:
active_channels_mask = (1 << num_channels) - 1;
In order to enable the WSA2 bits the channel count should be 4,
but the lower WSA bits are then also enabled and the DSP errors
out when trying to play on the disabled WSA interface.
A solution would've been to add a fake WSA2 topology element which
would be translated into the top bits only, but it's not clean and
add some special exceptions in the generic Audioreach code.
The solution suggested by Srinivas is to use the channel mapping to
set this bitmask.
This works but makes all the other calls using the channel mapping fail
because the DSP requires the channel_mapping table to start from index 0
and using num_channel length in order to apply the mapping on the
active_channels_mask bits in order.
So if you have an active_channels_mask like:
_______________________
| Bits | 3 | 2 | 1 | 0 |
-------------------------
| Mask | 1 | 1 | 0 | 0 |
------------------------
And the channel mapping:
_________________________________
| Channel | 3 | 2 | 1 | 0 |
-----------------------------------
| Mapping | FR | FL | NULL | NULL |
-----------------------------------
Will need to be converted to:
____________________
| Channel | 1 | 0 |
---------------------
| Mapping | FR | FL |
---------------------
And the DSP will map FR on WSA2 Ch2 and FL on WSA2 Ch1 as:
_________________________________
| Line | WSA2 Ch2 | WSA2 Ch1 |
----------------------------------
| Mapping | FR | FL |
----------------------------------
So we need to skip the empty channel mapping entries in all other
users of the channel_map to build valid channel_mapping tables.
This should not break any other usecases since the default channel
mapping always start from index 0, and will add flexibilty to allow
some special non linear mapping for other interfaces as well.
Finally we add a special prepare callback for the Ayaneo PS2 sound card
which sets the custom channel mapping for the WSA interface to make audio
work properly on the gaming device.
Link: https://patch.msgid.link/20260728-topic-sm8650-ayaneo-pocket-s2-wsa2-fix-v3-0-b29f44720178@linaro.org
The WSA Speakers are connected on the WSA2 interface, but the
WSA and WSA2 links are handled as a single dai and DSP interface, so
we need to specify the channel mapping of the Ayaneo Pocket S2 for the
WSA dai in order to have functional playback and avoid DSP errors.
Let's add a special entry for the Ayaneo Pocket S2 adding a prepare
callback in order to set the proper channel mapping.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://patch.msgid.link/20260728-topic-sm8650-ayaneo-pocket-s2-wsa2-fix-v3-3-b29f44720178@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The Qualcom SM8650 based Ayaneo Pocket S2 gaming device has a set
of 2 WSA speakers connected on the WSA2 lines.
But the Audioreach DSP only handles WSA2 in pair with the WSA
interface by using the upper bits of the active_channels_mask
for WSA2 and the lower bits for WSA:
/-------------------------------------------------\
| Bits | 3 | 2 | 1 | 0 |
|-------------------------------------------------|
| Line | WSA2 Ch2 | WSA2 Ch1 | WSA Ch2 | WSA Ch1 |
\-------------------------------------------------/
Setting only the WSA2 upper bits is perfectly valid and
functional but the current Audioreach code builds the bitmask
from the channels count with:
active_channels_mask = (1 << num_channels) - 1;
In order to enable the WSA2 bits the channel count should be 4,
but the lower WSA bits are then also enabled and the DSP errors
out when trying to play on the disabled WSA interface.
A solution would've been to add a fake WSA2 topology element which
would be translated into the top bits only, but it's not clean and
add some special exceptions in the generic Audioreach code.
The solution suggested by Srinivas is to use the channel mapping to
set this bitmask.
This works but makes all the other calls using the channel mapping fail
because the DSP requires the channel_mapping table to start from index 0
and using num_channel length in order to apply the mapping on the
active_channels_mask bits in order.
So we need to skip the empty channel mapping entries in all other
users of the channel_map to build valid channel_mapping tables.
This should not break any other usecases since the default channel
mapping always start from index 0, and will add flexibilty to allow
some special non linear mapping for other interfaces as well.
Suggested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://patch.msgid.link/20260728-topic-sm8650-ayaneo-pocket-s2-wsa2-fix-v3-1-b29f44720178@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
With this, SM8475 topologies can have their own firmware folder.
Signed-off-by: Esteban Urrutia <esteuwu@proton.me>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260729-sm8475-asoc-v1-2-3edad8aa7628@proton.me
Signed-off-by: Mark Brown <broonie@kernel.org>
The sc8280xp machine driver is currently written with a largely
SoC-centric view and assumes a uniform audio topology across all boards.
In practice, multiple products based on the same SoC use different board
designs and external audio components, which require board-specific
configuration to function correctly.
Several Qualcomm platforms integrate third-party audio codecs
or use different external audio paths. These designs often require
additional configuration such as explicit MI2S MCLK/BCLK settings for
audio to work.
This change enhances the sc8280xp machine driver to support board-specific
configuration such as allowing each board variant to provide its own DAPM
widgets and routes, reflecting the actual audio components and connectors
present and enabling MI2S MCLK programming for boards that use external
codecs requiring a stable master clock.
Tested-by: Neil Armstrong <neil.armstrong@linaro.org>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Link: https://patch.msgid.link/20260724182446.1484894-4-mohammad.rafi.shaik@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for MI2S clock control within q6apm-lpass DAIs, including
handling of MCLK, BCLK via the DAI .set_sysclk callback. Each MI2S port
now retrieves its clock handles from the device tree, allowing per-port
clock configuration and proper enable/disable during startup and shutdown.
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Link: https://patch.msgid.link/20260724182446.1484894-3-mohammad.rafi.shaik@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
storm_parse_of() can be processed without using *card.
No functional change, but is preparation for Card capsuling.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/87mrvli0hx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is not used. Let's remove.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@oss.qualcomm.com>
Link: https://patch.msgid.link/87o6g1i0i6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
qcom_snd_parse_of() already have dev. Let's use it.
No functional change, but is preparation for Card capsuling.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/87pl0hi0il.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the #include of <linux/mod_devicetable.h> by the more specific
<linux/device-id/*.h> where applicable. For most cases the include
can be dropped completely, only a few drivers need one or two headers
added.
Acked-by: Danilo Krummrich <dakr@kernel.org>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Bjorn Helgaas <bhelgaas@google.com>
Link: https://patch.msgid.link/1a3f2007c5c5dcf555c09a4035ce3ae8ef1b6c49.1782808461.git.u.kleine-koenig@baylibre.com
Signed-off-by: Uwe Kleine-König (The Capable Hub) <u.kleine-koenig@baylibre.com>
When q6apm_free_fragments() is called it frees rx_data.buf/tx_data.buf
and sets them to NULL under graph->lock. A late DSP buffer-done response
can race with this: graph_callback() passes the !graph->ar_graph guard
(not yet NULL), acquires the lock, but then dereferences a now-NULL buf
pointer to read buf[token].phys, crashing at virtual address 0x10.
Add a NULL check for buf inside the mutex-protected section in both the
write-done (DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2) and
read-done (DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2) handlers and bail
out cleanly if buffers have already been freed.
This problem is only shown up recently while apr bus was updated to
process the commands per service rather from single global queue.
Fixes: 5477518b8a ("ASoC: qdsp6: audioreach: add q6apm support")
Cc: Stable@vger.kernel.org
Assisted-by: Claude:claude-4-6-sonnet
Reported-by: Val Packett <val@packett.cool>
Closes: https://lore.kernel.org/all/133ced18-1aa9-475d-80d8-6120678bdde4@packett.cool/
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260616170257.9381-1-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cássio Gabriel <cassiogabrielcontato@gmail.com> says:
Some ASoC drivers still manually release child OF nodes
when leaving child-node iteration loops early.
Convert these focused cases to scoped OF node cleanup
so early returns and normal loop exits keep the same node
lifetime handling without explicit of_node_put() calls.
- Patch 1 updates qcom_snd_parse_of() to use
for_each_available_child_of_node_scoped() for link nodes and
__free(device_node) for temporary cpu/platform/codec child nodes.
- Patch 2 updates fsl_qmc_audio to use
for_each_available_child_of_node_scoped() for DAI child-node parsing.
- Patch 3 updates cygnus-ssp to use
for_each_available_child_of_node_scoped() for SSP child-node parsing.
Link: https://patch.msgid.link/20260608-asoc-of-node-scoped-cleanup-v1-0-9e3ac518dc2e@gmail.com
