priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
chip->flag variable assignment will be skipped when acp platform device
creation is skipped. In this case chip>flag value will not be set.
chip->flag variable should be assigned along with other structure
variables for 'chip' structure. Move chip->flag variable assignment
prior to acp platform device creation.
Fixes: 3a94c8ad0a ("ASoC: amd: acp: add code for scanning acp pdm controller")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP supports different pin configurations for I2S IO. Checking ACP pin
configuration value against specific value breaks the functionality for
other I2S pin configurations. This check is no longer required in i2s dai
driver probe call as i2s configuration check will be verified during acp
platform device creation sequence.
Remove i2s_mode check in acp_i2s_probe() function.
Fixes: b24484c18b ("ASoC: amd: acp: ACP code generic to support newer platforms")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
IRQ lookup functions such as those in ACPI can return error values when
an IRQ is not defined. The i2c core driver converts the error codes to a
value of 0 and the SPI bus driver passes them unaltered to client device
drivers.
The cs35l56 driver should only accept positive non-zero values as IRQ
numbers.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240617135338.82006-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
'struct mtk_adsp_ipc_ops' is not modified in these drivers.
Constifying this structure moves some data to a read-only section, so
increase overall security.
In order to do it, "struct mtk_adsp_ipc" also needs to be adjusted to this
new const qualifier.
On a x86_64, with allmodconfig:
Before:
======
text data bss dec hex filename
15533 2383 0 17916 45fc sound/soc/sof/mediatek/mt8195/mt8195.o
After:
=====
text data bss dec hex filename
15557 2367 0 17924 4604 sound/soc/sof/mediatek/mt8195/mt8195.o
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://msgid.link/r/a45d6b2b5ec040ea0fc78fca662c2dca3f13a49f.1718312321.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
This patchset adds support to reading codec version and also adds
support for v2.5 codec version in rx macro.
LPASS 2.5 and up versions have changes in some of the rx blocks which
are required to get headset functional correctly.
Tested this on SM8450, X13s and x1e80100 crd.
This changes also fixes issue with sm8450, sm8550, sm8660 and x1e80100.
LPASS Digital codec have changes in register layout across multiple
versions. Add a proper way read the codec version allowint all the lpass
macro drivers (tx, rx, wsa, va) to configure the registers correctly.
LPASS VA macro has the required registers to read the codec version.
Read the the version and make it available to other lpass codec macros
using the common helper functions.
Existing method of using LPASS IP version is not accurate as the same
the codec versioning is totally independent of LPASS IP block versions.
These helper functions should be able to provide a convient way to get
the codec version, and will help scale the drivers in right direction.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Link: https://lore.kernel.org/r/20240612-lpass-codec-v25-v4-1-f63d3676dbc4@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
There is an issue around with error handling and graph management with
the exising code, none of the error paths close the graph, which result in
leaving the loaded graph in dsp, however the driver thinks otherwise.
This can have a nasty side effect specially when we try to load the same
graph to dsp, dsp returns error which leaves the board with no sound and
requires restart.
Fix this by properly closing the graph when we hit errors between
open and close.
Fixes: 30ad723b93 ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://lore.kernel.org/r/20240613-q6apm-fixes-v1-1-d88953675ab3@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer
sources should be initialized to disconnected. There aren't currently
any available products using the ASP so this doesn't affect any
existing systems.
The cs35l56 does not have any fixed default for the mixer source
registers. When the cs35l56 boots, its firmware patches these registers
to setup a system-specific routing; this is so that Windows can use
generic SDCA drivers instead of needing knowledge of chip-specific
registers. The setup varies between end-products, which each have
customized firmware, and so the default register state varies between
end-products. It can also change if the firmware on an end-product is
upgraded - for example if a change was needed to the routing for Windows
use-cases. It must be emphasized that the settings applied by the
firmware are not internal magic tuning; they are statically implementing
use-case setup that on Linux would be done via ALSA controls.
The driver is currently syncing the mixer controls with whatever
initial state the firmware wrote to the registers, so that they report
the actual audio routing. But if the ASP DAI is hooked up this can create
a powered-up DAPM graph without anything intentionally setting up a path.
This can lead to parts of the audio system powering up unexpectedly.
For example when cs35l56 is connected to cs42l43 using a codec-codec link,
this can create a complete DAPM graph which then powers-up cs42l43. But
the cs42l43 can only be clocked from its SoundWire bus so this causes a
bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up
without a clock.
If the host is taking ownership of the ASP (either directly or as a
codec-to-codec link) there is no need to keep the mixer settings that the
firmware wrote. The driver has ALSA controls for setting these using
standard Linux mechanisms. So if the machine driver hooks up the ASP the
ASP mixers are initialized to "None" (no input). This prevents unintended
DAPM-graph power-ups, and means the initial state of the mixers is
always going to be None.
Since the initial state of the mixers can vary from system to system and
potentially between firmware upgrades, no use-case manager can currently
assume that cs35l56 has a known initial state. The firmware could just as
easily default them to "None" as to any input source. So defaulting them
to "None" in the driver is not increasing the entropy of the system.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With ARCH=x86, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/renoir/snd-acp3x-rn.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/yc/snd-soc-acp6x-mach.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-i2s.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-pdm.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-legacy-common.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-pci.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/ps/snd-soc-ps-mach.o
Add the missing invocations of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://lore.kernel.org/r/20240612-md-sound-soc-amd-v1-1-ad1de0409c11@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Jai Luthra <j-luthra@ti.com>:
This series fixes two patches:
1. Fix the dmaengine API usage by calling dmaengine_synchronize() after
dmaengine_terminate_async() when xrun events occur in application
2. Use the McASP AFIFO property from DT to refine the period size,
instead of hardcoding minimum to 64 samples
The minimum period size was enforced to 64 as older devices integrating
McASP with EDMA used an internal FIFO of 64 samples.
With UDMA based platforms this internal McASP FIFO is optional, as the
DMA engine internally does some buffering which is already accounted for
when registering the platform. So we should read the actual FIFO
configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to
64.
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Andy Shevchenko <andriy.shevchenko@linux.intel.com>:
Replace or drop the legacy header that is subject to remove.
Not all of them were compile-tested, the series might have
hidden compilation errors.
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
Originally reported here:
https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-2127892605
There is various level of failure there, first of all when topology
loads routes, it points directly into FW file, but it may be freed after
topology load. After fixing the above, when avs driver parses topology
it should allocate its own memory, as target strings can be shorter than
needed. Also clean up soc_tplg_dapm_graph_elems_load() a bit.
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
The acpi_get_local_address() helper assumes a 32-bit ADR is used. If
we want to use this helper for SoundWire/SDCA ASoC codecs, we need an
extension where the native 64-bits are used. This patchset suggests a
new helper, acpi_get_local_address() may be renamed if desired in a
folow-up patch.
The path of least resistance would be to merge this patchset in the
ASoC tree, since I have additional changes for ASoC/SDCA (SoundWire
Device Class) that depend on the new helper.
Pierre-Louis Bossart (3):
ACPI: utils: introduce acpi_get_local_u64_address()
soundwire: slave: simplify code with acpi_get_local_u64_address()
ALSA: hda: intel-sdw-acpi: use acpi_get_local_u64_address()
drivers/acpi/utils.c | 22 ++++++++++++++++------
drivers/soundwire/slave.c | 13 ++++---------
include/linux/acpi.h | 1 +
sound/hda/intel-sdw-acpi.c | 6 +++---
4 files changed, 24 insertions(+), 18 deletions(-)
--
2.43.0
Merge series from Mohammad Rafi Shaik <quic_mohs@quicinc.com>:
This patchset adds support for Qualcomm WCD9370/WCD9375 codec.
Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC
connected over SoundWire. This device has two SoundWire devices, RX and
TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH,
6 DMICs and MBHC.
For codec driver to be functional it would need both tx and rx Soundwire devices
to be up and this is taken care by using device component framework and device-links
are used to ensure proper pm dependencies. Ex tx does not enter suspend
before rx or codec is suspended.
This patchset along with other SoundWire patches on the list
have been tested on QCM6490 IDP device.
