Miglia Harmony Audio does neither support AV/C Stream Format Information
command nor AV/C Extended Stream Format Information command.
This commit adds a workaround for the case and uses the hard-coded formats.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218074128.95210-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit fixes the following warning when building virtio_snd driver.
"
*** CID 1583619: Uninitialized variables (UNINIT)
sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op()
288
289 break;
290 }
291
292 kfree(tlv);
293
vvv CID 1583619: Uninitialized variables (UNINIT)
vvv Using uninitialized value "rc".
294 return rc;
295 }
296
297 /**
298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type.
299 * @snd: VirtIO sound device.
"
This warning is caused by the absence of the "default" branch in the
switch-block, and is a false positive because the kernel calls
virtsnd_kctl_tlv_op() only with values for op_flag processed in
this block.
Also, this commit unifies the cleanup path for all possible control
paths in the callback function.
Signed-off-by: Anton Yakovlev <anton.yakovlev@opensynergy.com>
Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac@opensynergy.com>
Reported-by: coverity-bot <keescook+coverity-bot@chromium.org>
Addresses-Coverity-ID: 1583619 ("Uninitialized variables")
Fixes: d6568e3de4 ("ALSA: virtio: add support for audio controls")
Acked-by: Michael S. Tsirkin <mst@redhat.com>
Link: https://lore.kernel.org/r/20240216100643.688590-1-aiswarya.cyriac@opensynergy.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In modern C versions, 'bool' is a keyword that cannot be used as
a variable name, so change this instance use something else, and
change the type to bool instead.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240216130211.3828455-1-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The USB audio driver tries to retrieve MIDI jack name strings that can
be used for rawmidi substream names and sequencer port names, but its
checking is too strict: often the firmware provides the jack info for
unexpected directions, and then we miss the info although it's
present.
In this patch, the code to extract the jack info is changed to allow
both in and out directions in a single loop. That is, the former two
functions to obtain the descriptor pointers for jack in and out are
changed to a single function that returns iJack of the corresponding
jack ID, no matter which direction is used. It's a code
simplification at the same time as well as the fix.
Fixes: eb596e0fd1 ("ALSA: usb-audio: generate midi streaming substream names from jack names")
Link: https://lore.kernel.org/r/20240215153144.26047-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP mt645 G7 Thin Client uses an ALC236 codec and needs the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make the mute and
micmute LEDs work.
There are two variants of the USB-C PD chip on this device. Each uses
a different BIOS and board ID, hence the two entries.
Signed-off-by: Eniac Zhang <eniac-xw.zhang@hp.com>
Signed-off-by: Alexandru Gagniuc <alexandru.gagniuc@hp.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240215154922.778394-1-alexandru.gagniuc@hp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A relatively large set of fixes and quirk additions here but they're all
driver specific, people seem to be back into the swing of things after
the holidays. This is all driver specific and much of it fairly minor.
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Merge tag 'asoc-fix-v6.8-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
A relatively large set of fixes and quirk additions here but they're all
driver specific, people seem to be back into the swing of things after
the holidays. This is all driver specific and much of it fairly minor.
Since commit d492cc2573 ("driver core: device.h: make struct
bus_type a const *"), the driver core can properly handle constant
struct bus_type, move the snd_seq_bus_type variable to be a constant
structure as well, placing it into read-only memory which can not be
modified at runtime.
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Suggested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Ricardo B. Marliere <ricardo@marliere.net>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20240214-bus_cleanup-alsa-v1-2-8fedbb4afa94@marliere.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit d492cc2573 ("driver core: device.h: make struct
bus_type a const *"), the driver core can properly handle constant
struct bus_type, move the soundbus_bus_type variable to be a constant
structure as well, placing it into read-only memory which can not be
modified at runtime.
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Suggested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Ricardo B. Marliere <ricardo@marliere.net>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20240214-bus_cleanup-alsa-v1-1-8fedbb4afa94@marliere.net
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.
The patch 51d9760799 ("ALSA: hda/realtek:
Add quirks for ASUS Zenbook 2022 Models") modified the entry 1043:1e2e
from "ASUS UM3402" to "ASUS UM6702RA/RC" and added another entry for
"ASUS UM3402" with 104e:1ee2.
The first entry was correct, while the new one corresponds to model
"ASUS UM6702RA/RC"
Fix the model names for both devices.
Fixes: 51d9760799 ("ALSA: hda/realtek: Add quirks for ASUS Zenbook 2022 Models")
Signed-off-by: Jean-Loïc Charroud <lagiraudiere+linux@free.fr>
Link: https://lore.kernel.org/r/1656546983.650349575.1707867732866.JavaMail.zimbra@free.fr
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
clang-16 points out a control flow integrity (kcfi) issue when event
callbacks get converted to incompatible types:
sound/core/seq/seq_midi.c:135:30: error: cast from 'int (*)(struct snd_rawmidi_substream *, const char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
135 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/core/seq/seq_virmidi.c:83:31: error: cast from 'int (*)(struct snd_rawmidi_substream *, const unsigned char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
83 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
For addressing those errors, introduce wrapper functions that are used
for callbacks and bridge to the actual function call with pointer
cast.
The code was originally added with the initial ALSA merge in linux-2.5.4.
[ the patch description shamelessly copied from Arnd's original patch
-- tiwai ]
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240213101020.459183-1-arnd@kernel.org
Link: https://lore.kernel.org/r/20240213135343.16411-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
clang-16 started warning about incompatible function pointers here:
sound/pci/aw2/aw2-alsa.c:363:11: error: cast from 'void (*)(struct snd_pcm_substream *)' to 'snd_aw2_saa7146_it_cb' (aka 'void (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
363 | (snd_aw2_saa7146_it_cb)
| ^~~~~~~~~~~~~~~~~~~~~~~
364 | snd_pcm_period_elapsed,
| ~~~~~~~~~~~~~~~~~~~~~~
sound/pci/aw2/aw2-alsa.c:392:10: error: cast from 'void (*)(struct snd_pcm_substream *)' to 'snd_aw2_saa7146_it_cb' (aka 'void (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
392 | (snd_aw2_saa7146_it_cb)
| ^~~~~~~~~~~~~~~~~~~~~~~
393 | snd_pcm_period_elapsed,
| ~~~~~~~~~~~~~~~~~~~~~~
Add a forward declaration for struct snd_pcm_substrea to allow it to just
use the correct prototype.
Fixes: 98f2a97f20 ("[ALSA] Emagic Audiowerk 2 ALSA driver.")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240213101327.460191-1-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver creates an abstraction for different components by casting function
pointers to slightly incompatible types for each one to get the correct
argument even when the caller does not know those types. This is a
bit unreliable and not allowed in combination with control flow integrity
(KCFI):
sound/pci/ctxfi/ctatc.c:115:25: error: cast from 'int (*)(struct hw *, struct src_mgr **)' to 'create_t' (aka 'int (*)(struct hw *, void **)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
115 | [SRC] = { .create = (create_t)src_mgr_create,
| ^~~~~~~~~~~~~~~~~~~~~~~~
sound/pci/ctxfi/ctatc.c:116:20: error: cast from 'int (*)(struct src_mgr *)' to 'destroy_t' (aka 'int (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
116 | .destroy = (destroy_t)src_mgr_destroy },
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/pci/ctxfi/ctatc.c:117:27: error: cast from 'int (*)(struct hw *, struct srcimp_mgr **)' to 'create_t' (aka 'int (*)(struct hw *, void **)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
117 | [SRCIMP] = { .create = (create_t)srcimp_mgr_create,
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/pci/ctxfi/ctatc.c:118:20: error: cast from 'int (*)(struct srcimp_mgr *)' to 'destroy_t' (aka 'int (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
118 | .destroy = (destroy_t)srcimp_mgr_destroy },
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Change these to always pass void pointers and move the abstraction one level
down.
Fixes: 8cc7236148 ("ALSA: SB X-Fi driver merge")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240213101303.460008-1-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
There are two ways to opportunistically increment a device's runtime PM
usage count, calling either pm_runtime_get_if_active() or
pm_runtime_get_if_in_use(). The former has an argument to tell whether to
ignore the usage count or not, and the latter simply calls the former with
ign_usage_count set to false. The other users that want to ignore the
usage_count will have to explicitly set that argument to true which is a
bit cumbersome.
To make this function more practical to use, remove the ign_usage_count
argument from the function. The main implementation is in a static
function called pm_runtime_get_conditional() and implementations of
pm_runtime_get_if_active() and pm_runtime_get_if_in_use() are moved to
runtime.c.
Signed-off-by: Sakari Ailus <sakari.ailus@linux.intel.com>
Reviewed-by: Alex Elder <elder@linaro.org>
Reviewed-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Takashi Iwai <tiwai@suse.de> # sound/
Reviewed-by: Jacek Lawrynowicz <jacek.lawrynowicz@linux.intel.com> # drivers/accel/ivpu/
Acked-by: Rodrigo Vivi <rodrigo.vivi@intel.com> # drivers/gpu/drm/i915/
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Acked-by: Bjorn Helgaas <bhelgaas@google.com> # drivers/pci/
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
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.
This ends up with the allocation of a few additional bytes for the
state dumps even if it's not really used, but the code simplification
should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-29-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of all 4 pages no matter with
CONFIG_PM, but the code simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-28-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
The temporary buffers for PCM stream backups are conditionally
allocated since the sizes aren't too small.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-27-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with an additional allocation of a flag without
CONFIG_PM, but the code simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-26-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
The area for register dump is conditionally allocated instead of
ifdef now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-25-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-24-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-23-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-22-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of firmware caches if it's not really
used without CONFIG_PM, but the code simplification should justify the
cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-21-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-20-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-19-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
register dumps even if it's not really used, but the code
simplification should justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-18-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
This ends up with the allocation of a few additional bytes for the
image even if it's not really used, but the code-simplification should
justify the cost.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-17-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the memory with an additional devm_kmalloc(),
just put the image into the existing struct snd_ali. The allocation
size isn't too big, hence it works better with less allocation calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-16-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-15-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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/20240207155140.18238-13-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
For building properly, add the dummy functions for
snd_ak4531_suspend/resume() functions, too.
Just a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
For building properly, add the dummy functions for
snd_sbmixer_suspend/resume() functions, too.
Just a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240207155140.18238-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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/20240207155140.18238-2-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>
Implementation of support for audio controls in accordance with the
extension of the virtio sound device specification [1] planned for
virtio-v1.3-cs01.
The device can announce the VIRTIO_SND_F_CTLS feature. If the feature is
negotiated, then an additional field appears in the configuration space:
struct virtio_snd_config {
...
/* number of available control elements */
__le32 controls;
};
The driver can send the following requests to manage audio controls:
enum {
...
/* control element request types */
VIRTIO_SND_R_CTL_INFO = 0x0300,
VIRTIO_SND_R_CTL_ENUM_ITEMS,
VIRTIO_SND_R_CTL_READ,
VIRTIO_SND_R_CTL_WRITE,
VIRTIO_SND_R_CTL_TLV_READ,
VIRTIO_SND_R_CTL_TLV_WRITE,
VIRTIO_SND_R_CTL_TLV_COMMAND,
...
};
And the device can send the following audio control event notification:
enum {
...
/* control element event types */
VIRTIO_SND_EVT_CTL_NOTIFY = 0x1200,
...
};
See additional details in [1].
[1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html
Signed-off-by: Anton Yakovlev <anton.yakovlev@opensynergy.com>
Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac@opensynergy.com>
Link: https://lore.kernel.org/r/20240115133654.576068-2-aiswarya.cyriac@opensynergy.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic
CS35L56 amplifier") adds configs SND_HDA_SCODEC_CS35L56_{I2C,SPI},
which selects the non-existing config CS_DSP. Note the renaming in
commit d7cfdf17cb ("firmware: cs_dsp: Rename KConfig symbol CS_DSP ->
FW_CS_DSP"), though.
Select the intended config FW_CS_DSP.
This broken select command probably was not noticed as the configs also
select SND_HDA_CS_DSP_CONTROLS and this then selects FW_CS_DSP. So, the
select FW_CS_DSP could actually be dropped, but we will keep this
redundancy in place as the author originally also intended to have this
redundancy of selects in place.
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com>
Reviewed-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240209082044.3981-1-lukas.bulwahn@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Existing DSDT or SSDT platforms hard-code clock and frame shape
configurations. For validation, we'd like to use alternate
configurations. It's not always possible to generate new tables due to
missing symbols, and modifying existing objects usually leads to
AE_OBJECT_EXIST errors.
The mechanism suggested in this patch is to add a NEW ACPI controller
device with a different _ADR value. e.g.
Scope (_SB_.PC00.RP08.PXSX.HDAS) {
Device (SDWP)
{
Name (_ADR, 0x40000001) // _ADR: Address
The desired _ADR can be passed as a parameter with
options snd-intel-sdw-acpi sdw_ctrl_addr=0x40000001
This solution leads to minimal tables with just what the developers or
validation engineers need, and without overriding any of the existing
firmware definitions. It's consistent with the recommendation to
extend ACPI definitions and not redefine them with a risk of conflict.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20240208163750.92849-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The SWS JS201D need a different pinconfig from windows driver.
Add a quirk to use a specific pinconfig to SWS JS201D.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240205013802.51907-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SKUs, and associated SSIDs, are no longer going to include the
CS35L41. They may come back, but will need a different quirk.
Fixes: aa8e3ef4fe ("ALSA: hda/realtek: Add quirks for various HP ENVY models")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240202170842.321818-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This pull request adds Richard Fitzgerald's series with extensive fixes
for the CS35L56, he said:
These patches fix various things that were undocumented, unknown or
uncertain when the original driver code was written. And also a few
things that were just bugs.
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Merge tag 'asoc-fix-v6.8-rc2-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
This pull request adds Richard Fitzgerald's series with extensive fixes
for the CS35L56, he said:
These patches fix various things that were undocumented, unknown or
uncertain when the original driver code was written. And also a few
things that were just bugs.
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.
For devices with multiple clock sources connected to a selector, we need
to check what a clock selector control request has returned. This is
needed to ensure that a requested clock source is indeed selected and for
autoclock feature to work.
For devices with single clock source connected, if we get an error there
is nothing else we can do about it. We can't skip clock selector setup as
it is required by some devices. So lets just ignore error in this case.
This should fix various buggy Mackie devices:
[ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
There is also interesting info from the Windows documentation [1] (this
is probably why manufacturers dont't even test this feature):
"The USB Audio 2.0 driver doesn't support clock selection. The driver
uses the Clock Source Entity, which is selected by default and never
issues a Clock Selector Control SET CUR request."
Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-audio-drivers [1]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240201115308.17838-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vaio VJFE-ADL is equipped with ALC269VC, and it needs
ALC298_FIXUP_SPK_VOLUME quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240201122114.30080-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove an unused stub function that calls a non-existant function.
This function was accidentally added as part of commit
2144833e7b ("ALSA: hda: cirrus_scodec: Add KUnit test"). It was
a relic of an earlier version of the test that should have been
removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2144833e7b ("ALSA: hda: cirrus_scodec: Add KUnit test")
Link: https://msgid.link/r/20240129162737.497-19-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Check whether the firmware is already patched. If so, include the
firmware version in the firmware file name.
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 either:
- FIRMWARE_MISSING == 0, or
- it is a secured CS35L56 (which implies that is was already patched),
cs35l56_hw_init() will set preloaded_fw_ver to the (non-zero)
firmware version if either of these conditions is true.
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.
Backport note:
This won't apply to kernel versions older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://msgid.link/r/20240129162737.497-18-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the filename field layout to:
cs35l56-rev[-s]-dsp1-misc[-sub].[wmfw|bin]
This is to keep the same firmware file naming scheme as the
CS35L56 ASoC driver.
This is not a compatibility break because no firmware files have
been published.
The original field layout matched the ASoC driver, but the way the
ASoC driver used the wm_adsp driver config to form this filename
was bugged. Fixing the ASoC driver to use the correct wm_adsp config
strings means that the 's' flag (to indicate a secured part) has to
move to somewhere after the first '-'.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://msgid.link/r/20240129162737.497-17-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: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://msgid.link/r/20240129162737.497-16-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Add ASP1_FRAME_CONTROL1, ASP1_FRAME_CONTROL5 and the ASP1_TX?_INPUT
registers to the sequence used to initialize the ASP configuration.
Write this sequence to the cache and directly to the registers to
ensure that they match.
A system-specific firmware can patch these registers to values that are
not the silicon default, so that the CS35L56 boots already in the
configuration used by Windows or by "driverless" Windows setups such
as factory tuning.
These may not match how Linux is configuring the HDA codec. And anyway
on Linux the ALSA controls are used to configure routing options.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://msgid.link/r/20240129162737.497-10-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>
Quite a lot of fixes that came in since the merge window, a large
portion for for Qualcomm and ES8326.
The 8 DAI support for Qualcomm is just raising a constant to allow for
devies that otherwise only need DTs, and there's a few other device ID
updates for sunxi (Allwinner) and AMD platforms.
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Merge tag 'asoc-fix-v6.8-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
Quite a lot of fixes that came in since the merge window, a large
portion for for Qualcomm and ES8326.
The 8 DAI support for Qualcomm is just raising a constant to allow for
devies that otherwise only need DTs, and there's a few other device ID
updates for sunxi (Allwinner) and AMD platforms.
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
There currently exists two thinkpad headset jack fixups:
ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
ALC285_FIXUP_THINKPAD_HEADSET_JACK
The latter is applied to alc285 and alc287 thinkpads which contain
bass speakers.
However, the former was only being applied to alc285 thinkpads,
leaving non-bass alc287 thinkpads with no headset button controls.
This patch fixes that by adding ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
to the alc287 chains, allowing the detection of headset buttons.
Signed-off-by: José Relvas <josemonsantorelvas@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240131113407.34698-3-josemonsantorelvas@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
At the moment, we have a decent amount of integration tests (selftests)
covering different aspects of the sound subsystem. However, a lot of
of sound-related in-kernel functions remains uncovered. This patch
introduces the KUnit test for the core part of the sound subsystem.
It includes 10 test cases:
- Coverage of the format-related inline functions from 'pcm.h' header
file: snd_pcm_format_physical_width, snd_pcm_format_width,
snd_pcm_format_signed, test_format_endianness
- Coverage of the available bytes counting functions from 'pcm.h'
header: snd_pcm_capture_avail, snd_pcm_playback_avail
- Coverage of functions from pcm_misc: snd_pcm_format_set_silence,
snd_pcm_format_name
- Coverage of card-related functions from init.c: snd_card_set_id,
snd_component_add
This patch depends on the previous patches in this patch series as they
contain fix for the bug, which was found during the test development.
Without them, the test doesn't pass.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20240125223522.1122765-3-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix snd_pcm_format_name so it won't return NULL-pointer in case if it
can't find the format in the 'snd_pcm_format_names' list. Return
"Unknown" instead, as it is done if the number passed to the function
is larger than a list size.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20240125223522.1122765-2-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add 4 missing formats to 'snd_pcm_format_names' array in order to be
able to get their names with 'snd_pcm_format_name' function.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20240125223522.1122765-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Many devices with a single alternate setting do not have a Valid
Alternate Setting Control and validation performed by
validate_sample_rate_table_v2v3() doesn't work on them and is not
really needed. So check the presense of control before sending
altsetting validation requests.
MOTU Microbook IIc is suffering the most without this check. It
takes up to 40 seconds to bootup due to how slow it switches
sampling rates:
[ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60
[ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0
[ 2659.164829] usb 3-2: Product: MicroBook IIc
[ 2659.164830] usb 3-2: Manufacturer: MOTU
[ 2659.166204] usb 3-2: Found last interface = 3
[ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug
[ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3
[ 2679.322321] usb 3-2: Creating new data endpoint #3
[ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1
[ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110
[ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug
[ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84
[ 2694.444721] usb 3-2: Creating new data endpoint #84
[ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 67794f882a.
We need to explicitly set up the clock selector to workaround a problem
with the Behringer mixers. This was originally done in d2e8f64125
("ALSA: usb-audio: Explicitly set up the clock selector")
The problem with MOTU M Series mentioned in commit message was fixed in
a different way by checking control capabilities of clock selectors.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240128132338.819273-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This HP Laptop uses ALC236 codec with COEF 0x07 controlling the
mute LED. Enable existing quirk for this device.
Signed-off-by: Luka Guzenko <l.guzenko@web.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240128155704.2333812-1-l.guzenko@web.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If you connect an external headset/microphone to the 3.5mm jack on the
Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes
that and gives the user the ability to choose between internal and
headset mic.
Signed-off-by: David Senoner <seda18@rolmail.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark of the Unicorn released 896 mk3 FireWire in 2008 as part of the third
generation of its FireWire series. In 2011, 896 mk3 hybrid was released to
support USB protocol.
It supports sampling transfer frequency up to 192.0 kHz. The packet
format differs depending on both of current sampling transfer frequency
and the type of signal in optical interfaces. The model supports
transmission of PCM frames as well as MIDI messages.
The 896 mk3 FireWire consists of below ICs:
* Texas Instruments TSB41AB2
* Xilinx Spartan-3A FPGA, XC3S500E
* Texas Instruments TMS320C6722
* Microchip (Atmel) AT91SAM SAM7S256
It supports sampling transfer frequency up to 192.0 kHz. The packet
format differs depending on both of current sampling transfer frequency
and the type of signal in two pairs of optical interfaces. The model
supports transmission of PCM frames, while has no port for MIDi messages.
The model supports command mechanism to configure internal DSP. Hardware
meter information is available in the first 2 chunks of each data block
of tx packet.
This commit adds support for it. The 896 mk3 FireWire is just tested, but
the 896 mk3 Hybrid is not yet.
$ config-rom-pretty-printer < motu-896mk3fw.img
ROM header and bus information block
-----------------------------------------------------------------
1024 04100ce1 bus_info_length 4, crc_length 16, crc 3297
1028 31333934 bus_name "1394"
1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
1036 0001f200 company_id 0001f2 |
1040 00093add device_id 0000604893 | EUI-64 0547556791237341
root directory
-----------------------------------------------------------------
1044 0004ef04 directory_length 4, crc 61188
1048 030001f2 vendor
1052 0c0083c0 node capabilities: per IEEE 1394
1056 d1000002 --> unit directory at 1064
1060 8d000005 --> eui-64 leaf at 1080
unit directory at 1064
-----------------------------------------------------------------
1064 0003998d directory_length 3, crc 39309
1068 120001f2 specifier id
1072 13000017 version
1076 17101800 model
eui-64 leaf at 1080
-----------------------------------------------------------------
1080 0002cc82 leaf_length 2, crc 52354
1084 0001f200 company_id 0001f2 |
1088 00093add device_id 0000604893 | EUI-64 0547556791237341
$ config-rom-pretty-printer < motu-896mk3hybrid.img
ROM header and bus information block
-----------------------------------------------------------------
1024 04103cbe bus_info_length 4, crc_length 16, crc 15550
1028 31333934 bus_name "1394"
1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
1036 0001f200 company_id 0001f2 |
1040 000ae601 device_id 0000714241 | EUI-64 0547556791346689
root directory
-----------------------------------------------------------------
1044 0004ef04 directory_length 4, crc 61188
1048 030001f2 vendor
1052 0c0083c0 node capabilities: per IEEE 1394
1056 d1000002 --> unit directory at 1064
1060 8d000005 --> eui-64 leaf at 1080
unit directory at 1064
-----------------------------------------------------------------
1064 000394ac directory_length 3, crc 38060
1068 120001f2 specifier id
1072 13000037 version
1076 17102800 model
eui-64 leaf at 1080
-----------------------------------------------------------------
1080 0002cf69 leaf_length 2, crc 53097
1084 0001f200 company_id 0001f2 |
1088 000ae601 device_id 0000714241 | EUI-64 0547556791346689
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240129022711.254383-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Laptops 103C8C66, 103C8C67, 103C8C68, 103C8C6A use a dual speaker id
system where each speaker has its own speaker id.
The generic configuration table doesn't support this, so it needs its
own function.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240126164005.367021-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current method to set Channel Index when the system is missing _DSD
assumes that the channels alternate, which is not guaranteed.
Instead use the same methodology as the main driver does when _DSD
exists.
Fixes: 8c4c216db8 ("ALSA: hda: cs35l41: Add config table to support many laptops without _DSD")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240126164005.367021-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
receive_buf() is called from ttyport_receive_buf() that expects values
">= 0" from serdev_controller_receive_buf(), change its return type from
ssize_t to size_t.
The need for this clean-up was noticed while fixing a warning, see
commit 94d0539425 ("Bluetooth: btnxpuart: fix recv_buf() return value").
Changing the callback prototype to return an unsigned seems the best way
to document the API and ensure that is properly used.
GNSS drivers implementation of serdev receive_buf() callback return
directly the return value of gnss_insert_raw(). gnss_insert_raw()
returns a signed int, however this is not an issue since the value
returned is always positive, because of the kfifo_in() implementation.
gnss_insert_raw() could be changed to return also an unsigned, however
this is not implemented here as request by the GNSS maintainer Johan
Hovold.
Suggested-by: Jiri Slaby <jirislaby@kernel.org>
Link: https://lore.kernel.org/all/087be419-ec6b-47ad-851a-5e1e3ea5cfcc@kernel.org/
Signed-off-by: Francesco Dolcini <francesco.dolcini@toradex.com>
Acked-by: Jonathan Cameron <Jonathan.Cameron@huawei.com> #for-iio
Reviewed-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Rob Herring <robh@kernel.org>
Reviewed-by: Alex Elder <elder@linaro.org>
Acked-by: Maximilian Luz <luzmaximilian@gmail.com> # for platform/surface
Acked-by: Lee Jones <lee@kernel.org>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@linux.intel.com>
Link: https://lore.kernel.org/r/20240122180551.34429-1-francesco@dolcini.it
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Clock selector control might be read-only. Add corresponding checks
to prevent sending control requests that would fail.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240125205457.28258-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Use USEC_PER_MSEC rather than the hard coded value of 1000.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240125103117.2622095-5-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Whilst reading cirrus,buttons-ohms the error from
device_property_read_u32_array() is not checked, whilst there is a
preceding device_property_count_u32() which is checked the property
read can still fail. Add the missing check.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240125103117.2622095-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add some missing commas, refactor a couple small bits of code.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240125103117.2622095-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use more forward declarations, move header guards to cover other
includes, and rely less on including headers through other headers.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Link: https://msgid.link/r/20240125103117.2622095-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks for two HP G11 laptops that use a Realtek HDA codec combined
with four CS35L56 amplifiers using SPI.
The CS35L56 driver uses the component binding interface, so uses the same
setup code as the CS35L41 quirks.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240125123301.41692-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the generic parts of the hda_component implementation into a new
hda_component module. This will allow other HDA codecs to add support
for the component binding API without duplicating all the code.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Tested-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/20240124112607.77614-3-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Slightly re-work the code around cs35l41_generic_fixup() and the component
binding search so that it can be re-used for other amps that use the
component binding mechanism.
The match string is stored in struct scodec_dev_name instead of hardcoding
it in the match function.
The tas2781 does not use the amp index as part of the driver name match.
But its match format string does not include a field for the index, so
snprintf() would safely ignore the p->index argument. Because of this there
is no need for a special match function for this case, the CS35L41 code
can be re-used.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Tested-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/20240124112607.77614-2-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ES8326 features a headphone volume control register and four DAC
volume control registers.
We add new volume Kcontrols for these registers to enhance the
configurability of the volume settings, providing users with
greater flexibility.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240124064806.30511-2-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the wcd939x codec driver, there are two instances where semicolons
are used after closing braces of a switch-case statement. These
semicolons are not required and do not adhere to the coding style
guidelines.
This patch removes the unnecessary semicolons at the end of the
switch-case statements which cleans up the code and ensures consistency
with the rest of the kernel coding style.
Signed-off-by: Yang Li <yang.lee@linux.alibaba.com>
Link: https://msgid.link/r/20240124004425.54020-1-yang.lee@linux.alibaba.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The quirk table entries should be put in the USB ID order, but some
entries have been put in random places. Re-sort them.
Fixes: bf990c1023 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue")
Fixes: fd28941cff ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless")
Fixes: dfd5fe19db ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless")
Fixes: 4a63e68a29 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.")
Fixes: 7822baa844 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Fixes: 4fb7c24f69 ("ALSA: usb-audio: Add quirk for Fiero SC-01")
Fixes: 2307a0e1ca ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)")
Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RODE NT-USB+ is marketed as a professional usb microphone, however the
usb audio interface is a mess:
[ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd
[ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4
[ 1.503912] usb 1-5: config 1 has no interface number 4
[ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09
[ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 1.519697] usb 1-5: Product: RØDE NT-USB+
[ 1.519699] usb 1-5: Manufacturer: RØDE
[ 1.519700] usb 1-5: SerialNumber: 1D773A1A
[ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82
[ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82
[ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2
Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get.
I have asked Rode support to fix it, but they show no interest.
Signed-off-by: Sean Young <sean@mess.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
virtqueue_enable_cb() will call virtqueue_poll() which will check if
queue is broken at beginning, so remove the virtqueue_is_broken() call
Signed-off-by: Li RongQing <lirongqing@baidu.com>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Link: https://lore.kernel.org/r/20240124120834.49410-1-lirongqing@baidu.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSID 0x0c0d platform. It can't mute speaker when HP plugged.
This patch add quirk to fill speaker pin verbtable.
And disable speaker passthrough.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/38b82976a875451d833d514cee34ff6a@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The framer codec interacts with a framer.
It allows to use some of the framer timeslots as audio channels to
transport audio data over the framer E1/T1/J1 lines.
It also reports line carrier detection events through the ALSA jack
detection feature.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Reviewed-by: Christophe Leroy <christophe.leroy@csgroup.eu>
Link: https://msgid.link/r/20240123165615.250303-2-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 086b957cc1 ("ALSA: usb-audio: Skip the clock selector
inquiry for single connections") we are already skipping clock selector
inquiry if only one clock source is connected, but we are still sending
a set request. Lets skip that too.
This should fix errors when setting a sample rate on devices that don't
have any controls present within the clock selector. An example of such
device is the new revision of MOTU M Series (07fd:000b):
AudioControl Interface Descriptor:
bLength 8
bDescriptorType 36
bDescriptorSubtype 11 (CLOCK_SELECTOR)
bClockID 1
bNrInPins 1
baCSourceID(0) 2
bmControls 0x00
iClockSelector 0
Perhaps we also should check if clock selectors are readable and writeable
like we already do for clock sources, but this is out of scope of this
patch.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240123134635.54026-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device fails to initialize otherwise, giving the following error:
[ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1
Signed-off-by: Julian Sikorski <belegdol+github@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Customer has reported an issue with specific desktop platform
where two CS42L42 codecs are connected to CS8409 HDA bridge.
If "Master Volume Control" is created then on Ubuntu OS UCM
left/right balance slider in UI audio settings has no effect.
This patch will fix this issue for a target paltform.
Fixes: 20e5077241 ("ALSA: hda/cs8409: Add support for dolphin")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apollo Lake seems to also suffer from IRQ timing issues. After being up for ~4
minutes, a Pentium N4200 system ends up falling back to workqueue-based IRQ
handling:
[ 208.019906] snd_hda_intel 0000:00:0e.0: IRQ timing workaround is activated
for card #0. Suggest a bigger bdl_pos_adj.
Unfortunately, the Baytrail and Braswell workaround value of 32 samples isn't
enough to fix the issue here. Default to 64 samples.
Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240122114512.55808-3-rsalvaterra@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have self-explanatory constants for Intel HDA devices, let's use them instead
of magic numbers and code comments.
Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240122114512.55808-2-rsalvaterra@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Johan Hovold <johan+linaro@kernel.org>:
To reduce the risk of speaker damage the PA gain needs to be limited on
machines like the Lenovo Thinkpad X13s until we have active speaker
protection in place.
Limit the gain to the current default setting provided by the UCM
configuration which most user have so far been using (due to a bug in
the configuration files which prevented hardware volume control [1]).
The wsa883x PA volume control also turned out to be broken, which meant
that the default setting used by UCM configuration is actually the
lowest level (-3 dB). With the codec driver fixed, hardware volume
control also works as expected.
Note that the new wsa884x driver most likely suffers from a similar bug,
I'll send a fix for that once I've got that confirmed.
Included is also a related fix for the LPASS WSA macro driver, which
was changing the digital gain setting behind the back of user space and
which can result in excessive (or too low) digital gain.
There are further Qualcomm codec drivers that similarly appear to
manipulate various gain settings, but on closer inspection it turns out
that they only write back the current settings. Tests reveal that these
writes are indeed needed for any prior updates to take effect (at least
for the WSA and RX macros).
[1] https://github.com/alsa-project/alsa-ucm-conf/pull/382
The UCM configuration for the Lenovo ThinkPad X13s has up until now
been setting the speaker PA volume to the minimum -3 dB when enabling
the speakers, but this does not prevent the user from increasing the
volume further.
Limit the digital gain and PA volumes to a combined -3 dB in the machine
driver to reduce the risk of speaker damage until we have active speaker
protection in place (or higher safe levels have been established).
Note that the PA volume limit cannot be set lower than 0 dB or
PulseAudio gets confused when the first 16 levels all map to -3 dB.
Also note that this will probably need to be generalised using
machine-specific limits, but a common limit should do for now.
Cc: <stable@vger.kernel.org> # 6.5
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122181819.4038-3-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Neil Armstrong <neil.armstrong@linaro.org>:
Add the main WCD9390/WCD9395 Audio Codec driver to support:
- 4 ADC inputs for up to 5 Analog Microphones
- 4 DMIC inputs for up to 8 Digital Microphones
- 4 Microphone BIAS
- Stereo Headphone output
- Mono EAR output
- MBHC engine for Headset Detection
This adds:
- bindings
- MBHC changes to support Type-C muc
- Soundwire Slave driver
- Code driver
The USB Mux subsystem support will be submitted separately
since it's a functionally separate subsystem connected over
I2C with it's own reset signal.
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series is a continuation of [1] to provide several fixes and
improvements to the SOF drivers targeting the Vangogh platform, as found on
Valve's Steam Deck OLED. The previous series only handled the legacy ACP
drivers.
Also note that, since v2, this patch series requires an updated SOF
topology to provide a correct DAI link ID for the BT codec. The binary
file should be located under:
/usr/lib/firmware/amd/sof-tplg/sof-vangogh-nau8821-max.tplg
Alternatively, as a temporary workaround, patch [2] can be used to adapt
the driver to the broken topology.
Another requirement to get this functional is patch [3] from v1, which has
been dropped from the series to be upstreamed via the SOF development
workflow.
[1]: https://lore.kernel.org/all/20231209203229.878730-1-cristian.ciocaltea@collabora.com/
[2]: https://lore.kernel.org/all/20231209205351.880797-11-cristian.ciocaltea@collabora.com/
[3]: https://lore.kernel.org/all/20231209205351.880797-12-cristian.ciocaltea@collabora.com/
Probe calls wcd938x_populate_dt_data() which already prints all the
error cases with dev_err_probe(), so skip the additional dev_err().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
WCD938x sound codec driver ignores return status of getting regulators
and returns EINVAL instead of EPROBE_DEFER. If regulator provider
probes after the codec, system is left without probed audio:
wcd938x_codec audio-codec: wcd938x_probe: Fail to obtain platform data
wcd938x_codec: probe of audio-codec failed with error -22
Fixes: 16572522ae ("ASoC: codecs: wcd938x-sdw: add SoundWire driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add condition check for cpu dai link initialization for amplifier
codec path, as same pcm id uses for both headset and speaker path
for RENOIR platforms.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://msgid.link/r/20240118143023.1903984-3-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The lowest headphones volume setting does not mute so the leave the TLV
mute flag unset.
This is specifically needed to let the sound server use the lowest gain
setting.
Fixes: c03226ba15 ("ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR")
Cc: <stable@vger.kernel.org> # 6.5
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122091130.27463-1-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The vendor driver appears to be modifying the gain settings behind the
back of user space but these hacks never made it upstream except for
some essentially dead code that adds a constant zero to the current gain
setting on DAPM events.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-5-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The LPASS WSA macro codec driver is updating the digital gain settings
behind the back of user space on DAPM events if companding has been
enabled.
As compander control is exported to user space, this can result in the
digital gain setting being incremented (or decremented) every time the
sound server is started and the codec suspended depending on what the
UCM configuration looks like.
Soon enough playback will become distorted (or too quiet).
This is specifically a problem on the Lenovo ThinkPad X13s as this
bypasses the limit for the digital gain setting that has been set by the
machine driver.
Fix this by simply dropping the compander gain offset hack. If someone
cares about modelling the impact of the compander setting this can
possibly be done by exporting it as a volume control later.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Cc: stable@vger.kernel.org # 5.11
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-4-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The PA gain can be set in steps of 1.5 dB from -3 dB to 18 dB, that is,
in 15 levels.
Fix the dB values for the PA volume control as experiments using wsa8835
show that the first 16 levels all map to the same lowest gain while the
last three map to the highest gain.
These values specifically need to be correct for the sound server to
provide proper volume control.
Note that level 0 (-3 dB) does not mute the PA so the mute flag should
also not be set.
Fixes: cdb09e6231 ("ASoC: codecs: wsa883x: add control, dapm widgets and map")
Cc: stable@vger.kernel.org # 6.0
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-2-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We get a noise issue during the startup of recording. We update the
register setting and dapm widgets to fix this issue.
we change callback type of es8326_mute function to mute_stream.
ES8326_ADC_MUTE is moved to es8326_mute function so it can
be turned on at last and turned off at first.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-6-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We modify the register settings to minimize headphone pop noise
during ES8326 power-up and music start/stop.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-5-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
