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ASoC: Updates for v6.7
More updates for v6,7 following the early merge request:
- Fixes for handling of component name prefixing when name prefixes
are used by the machine driver.
- Fixes for noise when stopping some Sounwire CODECs.
- Support for AMD ACP 6.3 and 7.0, Awinc AW88399, more Intel
platforms and more Qualcomm SC7180 platforms.
The existing code checks for the correct state transition after sending
a command. However, it is possible for the message box to return -1,
which indicates an error, if an error has occurred in the firmware.
We can detect if the error has occurred, and return a different error.
In addition, there is no recovering from a CSPL error, so the retry
mechanism is not needed in this case, and we can return immediately.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20231026150558.2105827-9-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To ensure the chip has correctly reset during probe and system suspend,
we need to force a software reset, in case of systems where the
hardware reset is not available.
The software reset register was labelled as volatile but not readable,
however, it is readable, (just returns 0x0). Adding it to readable
registers means it will be correctly treated as volatile, and thus
will not be cached.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231026150558.2105827-6-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The .remove() callback for a platform driver returns an int which makes
many driver authors wrongly assume it's possible to do error handling by
returning an error code. However the value returned is (mostly) ignored
and this typically results in resource leaks. To improve here there is a
quest to make the remove callback return void. In the first step of this
quest all drivers are converted to .remove_new() which already returns
void.
simple_util_remove() returned zero unconditionally. Make it return void
instead and convert all users to struct platform_device::remove_new().
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Herve Codina <herve.codina@bootlin.com>
Link: https://lore.kernel.org/r/20231013221945.1489203-13-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Updates for v6.7
This is quite a large set of changes but mostly due to API cleanups and
in driver specific ways rather than due to anything subsystem wide.
Highlights include:
- Standardisation of API prefixes on snd_soc_, removing asoc_.
- GPIO API usage improvements.
- Support for HDA patches.
- Lots of work on SOF, including crash dump support.
- Support for AMD platforms with es83xx, Awinc AT87390, many Intel
platforms, many Mediatek platforms, Qualcomm SM6115, Richtek RTQ9128
and Texas Instruments TAS575x.
[ the merge conflicts around SOF Intel HD-audio and CS35L41 subcodec
drivers are resolved here -- tiwai ]
AudioDSP firmware is the one who kicks SDxFIFOS calculation when a
stream is decoupled mode. During firmware bring up procedure, there is
no firmware running and the code-loading stream is always a decoupled
one. So, there is none to trigger the calculation and we end up with
false-positive timeout (-110) messages.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20231006102857.749143-4-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDAudio stream setup procedure differs between revisions of the
controller device. Currently the differences are handled directly within
AudioDSP platform drivers with if-statements. Implement a more generic
approach and expose a function that a platform driver may use to ensure
the correct procedure is followed each time.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20230926080623.43927-3-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On init we have sequence:
for_each_card_prelinks(card, i, dai_link) {
ret = snd_soc_add_pcm_runtime(card, dai_link);
ret = init_some_other_things(...);
if (ret)
goto probe_end:
for_each_card_rtds(card, rtd) {
ret = soc_init_pcm_runtime(card, rtd);
probe_end:
while on exit:
for_each_card_rtds(card, rtd)
snd_soc_link_exit(rtd);
If init_some_other_things() step fails due to error we end up with
not fully setup rtds and try to call snd_soc_link_exit on them, which
depending on contents on .link_exit handler, can end up dereferencing
NULL pointer.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230929103243.705433-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC is using 2 type of prefix (asoc_xxx() vs snd_soc_xxx()),
but these are unified into snd_soc_xxx().
simple_card / audio_graph drivers are historically using
asoc_xxx() prefix too. simple_card / audio_graph are not
ASoC framework, so let's use simple_card_xxx_() / audio_graph_xxx()
for global function prefix.
This patch has asoc_xxx() as define to keep compatible.
It will be removed if all drivers were switched to new style.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87edj4s26a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
When a firmware crashes it creats a panic information into a telemetry
slot. The panic format is defined by Zephyr, includes stack and
additional information to help to identify the reason for the crash.
Part of the firmware exception handling the firmware also sends an
EXCEPTION_CAUGHT notification.
This series implements the kernel side handling of the exception: print
information into the kernel log export the whole telemetry slot to user
space for tools extract additional information from the panic dump.
On Intel platforms there is a strict order requirement for the DMA
programming:
DSP side configures the buffer and sets the GEN bit
Host side sets the RUN bit.
In order to follow this flow, a new global message type has been added to
prepare the DSP side of the DMA:
host sends LOAD_LIBRARY_PREPARE with the dma_id
DSP side sets its buffer and sets the GEN bit
Host sets the RUN bit
Host sends LOAD_LIBRARY with dma_id and lib_id
DSP receives the library data.
It is up to the platform code to use the new prepare stage message and how
to handle the reply to it from the firmware, which can indicate that the
message type is not supported/handled.
In this case the kernel should proceed to the LOAD_LIBRARY stage assuming
a single stage library loading:
host sends LOAD_LIBRARY_PREPARE with the dma_id
DSP replies that the message type is not supported/handled
Host acknowledges the return code and sets the RUN bit
Host sends LOAD_LIBRARY with dma_id and lib_id
DSP receives the library data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://lore.kernel.org/r/20230915114018.1701-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pass the PCI SSID of the audio interface through to the machine driver.
This allows the machine driver to use the SSID to uniquely identify the
specific hardware configuration and apply any platform-specific
configuration.
struct snd_sof_pdata is passed around inside the SOF code, but it then
passes configuration information to the machine driver through
struct snd_soc_acpi_mach and struct snd_soc_acpi_mach_params. So SSID
information has been added to both snd_sof_pdata and
snd_soc_acpi_mach_params.
PCI does not define 0x0000 as an invalid value so we can't use zero to
indicate that the struct member was not written. Instead a flag is
included to indicate that a value has been written to the
subsystem_vendor and subsystem_device members.
sof_pci_probe() creates the struct snd_sof_pdata. It is passed a struct
pci_dev so it can fill in the SSID value.
sof_machine_check() finds the appropriate struct snd_soc_acpi_mach. It
copies the SSID information across to the struct snd_soc_acpi_mach_params.
This done before calling any custom set_mach_params() so that it could be
used by the set_mach_params() callback to apply variant params.
The machine driver receives the struct snd_soc_acpi_mach as its
platform_data.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add members to struct snd_soc_card to store the PCI subsystem ID (SSID)
of the soundcard.
The PCI specification provides two registers to store a vendor-specific
SSID that can be read by drivers to uniquely identify a particular
"soundcard". This is defined in the PCI specification to distinguish
products that use the same silicon (and therefore have the same silicon
ID) so that product-specific differences can be applied.
PCI only defines 0xFFFF as an invalid value. 0x0000 is not defined as
invalid. So the usual pattern of zero-filling the struct and then
assuming a zero value unset will not work. A flag is included to
indicate when the SSID information has been filled in.
Unlike DMI information, which has a free-format entirely up to the vendor,
the PCI SSID has a strictly defined format and a registry of vendor IDs.
It is usual in Windows drivers that the SSID is used as the sole identifier
of the specific end-product and the Windows driver contains tables mapping
that to information about the hardware setup, rather than using ACPI
properties.
This SSID is important information for ASoC components that need to apply
hardware-specific configuration on PCI-based systems.
As the SSID is a generic part of the PCI specification and is treated as
identifying the "soundcard", it is reasonable to include this information
in struct snd_soc_card, instead of components inventing their own custom
ways to pass this information around.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Linus Walleij <linus.walleij@linaro.org>:
The Maxim devices are pretty straight-forward to convert
over to use GPIO descriptors, so let's do it.
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series contains several fixes and improvements to drivers
based on the CS35l41 audio codec.
It has been verified on Valve's Steam Deck, except the HDA related patches.
Enabling the active/passive shared boosts requires setting SYNC_EN, but
*not* before receiving the PLL Lock signal.
Due to improper error handling, it was not obvious that waiting for the
completion operation times out and, consequently, the shared boost is
never activated.
Further investigations revealed the signal is triggered while
snd_pcm_start() is executed, right after receiving the
SNDRV_PCM_TRIGGER_START command, which happens long after the
SND_SOC_DAPM_PRE_PMU event handler is invoked as part of
snd_pcm_prepare(). That is where cs35l41_global_enable() is called
from.
Increasing the wait duration doesn't help, as it only causes an
unnecessary delay in the invocation of snd_pcm_start(). Moving the wait
and the subsequent regmap operations to the SNDRV_PCM_TRIGGER_START
callback is not a solution either, since they would be executed in an
IRQ-off atomic context.
Solve the issue by setting the SYNC_EN bit in PWR_CTRL3 register right
after receiving the PLL Lock interrupt.
Additionally, drop the unnecessary writes to PWR_CTRL1 register, part of
the original mdsync_up_seq, which would have toggled GLOBAL_EN with
unwanted consequences on PLL locking behavior.
Fixes: f503056493 ("ALSA: cs35l41: Add shared boost feature")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: David Rhodes <david.rhodes@cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-5-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MAX9768 is pretty straight forward to convert to GPIO
descriptors.
To name the GPIO properties, I looke at the bindings in
maxim,max9759.yaml which names these GPIO "mute" and
"shutdown" respectively.
No board files using platform data exist in the kernel, new
users can use GPIO descriptor tables if desired.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230911-descriptors-asoc-max-v2-1-b9d793fb768e@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Updates for v6.6
The rest of the updates for v6.6, some of the highlights include:
- A big API cleanup from Morimoto-san, rationalising the places we put
functions.
- Lots of work on the SOF framework, AMD and Intel drivers, including a
lot of cleanup and new device support.
- Standardisation of the presentation of jacks from drivers.
- Provision of some generic sound card DT properties.
- Conversion oof more drivers to the maple tree register cache.
- New drivers for AMD Van Gogh, AWInic AW88261, Cirrus Logic cs42l43,
various Intel platforms, Mediatek MT7986, RealTek RT1017 and StarFive
JH7110.
When the UMP Endpoint is declared as "static", that is, no dynamic
reassignment of UMP Groups, it makes little sense to expose always all
16 groups with 16 substreams. Many of those substreams are disabled
groups, hence they are useless, but applications don't know it and try
to open / access all those substreams unnecessarily.
This patch limits the number of UMP legacy rawmidi substreams only to
the active groups. The behavior is changed only for the static
endpoint (i.e. devices without UMP v1.1 feature implemented or with
the static block flag is set).
Fixes: 0b5288f5fe ("ALSA: ump: Add legacy raw MIDI support")
Link: https://lore.kernel.org/r/20230824075108.29958-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Xingyu Wu <xingyu.wu@starfivetech.com>:
This patch series adds I2S support for the StarFive JH7110 RISC-V
SoC based on Designware I2S controller. There has three I2S channels
(RX/TX0/TX1) on the JH7110 SoC, one of which is for record(RX) and
two for playback(TX).
The first patch adds support for the StarFive JH7110 SoC in the
Designware I2S bindings.
The second patch adds the ops to get data from platform bus in the
I2S driver.
The third patch adds support for the StarFive JH7110 SoC in
the Designware I2S driver.
The fourth patch fixes the name of I2STX1 pinmux.
The last patch adds device node of I2S RX/TX0/TX1 in JH7110 dts.
This patch series is based on Linux-next(20230818) which is merge
clock, syscon and dma nodes for the StarFive JH7110 SoC.
The series has been tested and works normally on the VisionFive 2
board by plugging an audio expansion board.
Add StarFive JH7110(TX0/TX1/RX channels) SoC support in the
designware I2S driver and a flag to check if it is on the JH7110 SoC.
These channels need to enable clocks, resets and syscon register on the
JH7110 SoC. So add init ops in platform data for the JH7110 SoC to do this.
Their resets should be deassert before changing the parent of clocks so
these are done in the init ops of platform data.
The I2S controllers use DMA controller by platform data on the JH7110
and their settings about snd_dmaengine_dai_dma_data() should be added
in the dw_configure_dai_by_pd(). And use dmaengine PCM registration if
these do not have IRQ on the JH7110 SoC.
Signed-off-by: Xingyu Wu <xingyu.wu@starfivetech.com>
Link: https://lore.kernel.org/r/20230821144151.207339-4-xingyu.wu@starfivetech.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS42L43 is an audio CODEC with integrated MIPI SoundWire interface
(Version 1.2.1 compliant), I2C, SPI, and I2S/TDM interfaces designed
for portable applications. It provides a high dynamic range, stereo
DAC for headphone output, two integrated Class D amplifiers for
loudspeakers, and two ADCs for wired headset microphone input or
stereo line input. PDM inputs are provided for digital microphones.
The ASoC component provides the majority of the functionality of the
device, all the audio functions.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230804104602.395892-7-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
