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[ Upstream commit c2c0b67dca3cb3b3cea0dd60075a1c5ba77e2fcd ]
The tas2781-i2c driver gets an IRQ from either ACPI or device tree,
then proceeds to check if the IRQ has a corresponding GPIO and in
case it does enforce the GPIO as input and set a label on it.
This is abuse of the API:
- First we cannot guarantee that the numberspaces of the GPIOs and
the IRQs are the same, i.e that an IRQ number corresponds to
a GPIO number like that.
- Second, GPIO chips and IRQ chips should be treated as orthogonal
APIs, the irqchip needs to ascertain that the backing GPIO line
is set to input etc just using the irqchip.
- Third it is using the legacy <linux/gpio.h> API which should not
be used in new code yet this was added just a year ago.
Delete the offending code.
If this creates problems the GPIO and irqchip maintainers can help
to fix the issues.
It *should* not create any problems, because the irq isn't
used anywhere in the driver, it's just obtained and then
left unused.
Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver")
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://patch.msgid.link/20240807-asoc-tas-gpios-v2-1-bd0f2705d58b@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit e6ce8a28c768dbbad3f818db286cd0f4c7a921a8 ]
The UMP 1.1 spec says that an RPN/NRPN should be sent when one of the
following occurs:
* a CC 38 is received
* a subsequent CC 6 is received
* a CC 98, 99, 100, and 101 is received, indicating the last RPN/NRPN
message has ended and a new one has started
That said, we should send a partial data even if it's not fully
filled. Let's change the UMP conversion helper code to follow that
rule.
Link: https://patch.msgid.link/20240731130528.12600-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit e42066df07c0fcedebb32ed56f8bc39b4bf86337 ]
Use the late-read buffer in the CS35L56 SoundWire interface to
read OTP memory.
The OTP memory has a longer access latency than chip registers
and cannot guarantee to return the data value in the SoundWire
control response if the bus clock is >4.8 MHz. The Cirrus
SoundWire peripheral IP exposes the bridge-to-bus read buffer
and status bits. For a read from OTP the bridge status bits are
polled to wait for the OTP data to be loaded into the read buffer
and the data is then read from there.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration")
Link: https://patch.msgid.link/20240805140839.26042-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 5d7e328e20b3d2bd3e1e8bea7a868ab8892aeed1 ]
This patch reverts a series of commits that allowed for the ASP
registers to be owned by either the driver or the firmware. Nothing
currently depends on the functionality that is being reverted, so
it is safe to remove.
The commits being reverted are (last 3 are bugfixes to the first 2):
commit 72a77d7631c6
("ASoC: cs35l56: Fix to ensure ASP1 registers match cache")
commit 07f7d6e7a124
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
commit 4703b014f28b
("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()")
commit c14f09f010cc
("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization")
commit dfd2ffb37399
("ASoC: cs35l56: Prevent overwriting firmware ASP config")
These reverts have been squashed into a single commit because there
would be no reason to revert only some of them (which would just
reintroduce bugs).
The changes introduced by the commits were well-intentioned but
somewhat misguided. ACPI does not provide any information about how
audio hardware is linked together, so that information has to be
hardcoded into drivers. On Windows the firmware is customized to
statically setup appropriate configuration of the audio links,
and the intent of the commits was to re-use this information if the
Linux host drivers aren't taking control of the ASP. This would
avoid having to hardcode the ASP config into the machine driver on
some systems.
However, this added complexity and race conditions into the driver.
It also complicates implementation of new code.
The only case where the ASP is used but the host is not taking
ownership is when CS35L56 is used in SoundWire mode with the ASP
as a reference audio interconnect. But even in that case it's not
necessarily required even if the firmware initialized it. Typically
it is used to avoid the host SDCA drivers having to be capable of
aggregating capture paths from multiple SoundWire peripherals. But
the SOF SoundWire support is capable of doing that aggregation.
Reverting all these commits significantly simplifies the driver.
Let's just use the normal Linux mechanisms of the machine driver and
ALSA controls to set things up instead of trying to use the firmware
to do use-case setup.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: e42066df07c0 ("ASoC: cs35l56: Handle OTP read latency over SoundWire")
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 9f774c757e3fb2ac32dc4377e8f21f3364a8df81 ]
In only loading RCA (Reconfigurable Architecture) binary case, no DSP
program will be working inside tas2563/tas2781, that is dsp-bypass mode,
do not support speaker protection, or audio acoustic algorithms in this
mode.
Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240614133646.910-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 244389bd42870640c4b5ef672a360da329b579ed upstream.
Change CS35L56_MAIN_RENDER_USER_VOLUME_MAX to 48, to limit the maximum
value of the Speaker Volume control to +12dB. The minimum value is
unchanged so that the default 0dB has the same integer control value.
The original maximum of 400 (+100dB) was the largest value that can be
mathematically handled by the DSP. The actual maximum amplification is
+12dB.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703095517.208077-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fix the typo in the comment for SNDRV_PCM_RATE_KNOT
Signed-off-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20240528191850.63314-1-alexandre.belloni@bootlin.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A bunch of fixes that came in during the merge window, all driver
specific and none of them especially remarkable.
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Calibrated data will be set to default after loading DSP config params,
which will cause speaker protection work abnormally. Reload calibrated
data after loading DSP config params. Remove declaration of unused API
which load calibrated data in wrong sequence, changed the copyright year
and correct file name in license
header.
Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://msgid.link/r/20240518141546.1742-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.10
This is a very big update, in large part due to extensive work the Intel
people have been doing in their drivers though it's also been busy
elsewhere. There's also a big overhaul of the DAPM documentation from
Luca Ceresoli arising from the work he did putting together his recent
ELC talk, and he also contributed a new tool for visualising the DAPM
state.
- A new tool dapm-graph for visualising the DAPM state.
- Substantial fixes and clarifications for the DAPM documentation.
- Very large updates throughout the Intel audio drivers.
- Cleanups of accessors for driver data, module labelling, and for
constification.
- Modernsation and cleanup work in the Mediatek drivers.
- Several fixes and features for the DaVinci I2S driver.
- New drivers for several AMD and Intel platforms, Nuvoton NAU8325,
Rockchip RK3308 and Texas Instruments PCM6240.
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For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull HD-audio CONFIG_PM cleanup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add HD Audio PCI ID and HDMI codec vendor ID for Intel Battlemage.
Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah@intel.com>
Link: https://lore.kernel.org/r/20240506052531.1150062-1-chaitanya.kumar.borah@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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CONFIG_PM is almost mandatory nowadays for real systems, but we have
lots of CONFIG_PM dependent code in snd-hda-codec helper code.
Let's reduce the dependencies of CONFIG_PM now. The only visible
drawback would be a couple of superfluous trace entries for runtime
PM, but we can live with that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-3-tiwai@suse.de
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A common pattern in sound drivers is getting 'struct snd_soc_pcm_runtime'
from 'struct snd_pcm_substream' opaque pointer private_data field with
snd_soc_substream_to_rtd(). However 'private_data' appears in several
other structures as well, including 'struct snd_compr_stream'. The
field might not hold the same type for every structure, although seems
the case at least for 'struct snd_compr_stream', so code can easily make
a mistake by using macro for wrong structure passed as argument.
Switch from macro to inline function, so such mistake will be build-time
detectable.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240501175127.34301-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
This is much larger than is ideal, partly due to your holiday but also
due to several vendors having come in with relatively large fixes at
similar times. It's all driver specific stuff.
The meson fixes from Jerome fix some rare timing issues with blocking
operations happening in triggers, plus the continuous clock support
which fixes clocking for some platforms. The SOF series from Peter
builds to the fix to avoid spurious resets of ChainDMA which triggered
errors in cleanup paths with both PulseAudio and PipeWire, and there's
also some simple new debugfs files from Pierre which make support a lot
eaiser.
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Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
A set of changes that aims to improve readability of cohesiveness of the
pcm code for the avs-driver.
Start off with a change that synchronizes DAI open/close - DAIs are
started up in ascending order yet their shutdown does not follow the
scheme - it is done in the ascending order too, rather than desceding
one. This patch is a dependency for the next one in line.
To align the HDAudio DAI startup/shutdown with the non-HDAudio
equivalents, relocate the code from component to DAI. The reason above
is a dependency stems from codec driver requirements - HDAudio code
found in sound/pci/hda/ expects substream->runtime->private_data to
point to a valid stream (HOST) pointer.
With the hard part done, the follow up changes update the existing code
to reduce it is complexity - removal of duplicates, renaming of
ambiguous functions and adding new fields to DAI-data object so that the
number of local variables and casts is reduced.
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The core code does not modify the 'struct snd_sof_dsp_ops', stored in
'struct sof_dev_desc'. Make the pointer to 'struct snd_sof_dsp_ops'
const to annotate this, make code a bit safer and allow individual
drivers to also define this structure as const.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-4-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some new SOF changes depend on the fixes there.
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During startup snd_soc_dai_startup() is launched in ascending order and
the exact same thing is done during shutdown procedure. Reverse the
order in the latter so that it is symmetric to the former.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Pull emu10k1 fixes from Oswald Buddenhagen
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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... into snd_emu1010_load_firmware_entry(). This makes it clearer that
these steps belong together tightly, as implied by prior commits.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-5-oswald.buddenhagen@gmx.de>
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It is a low-level I/O access function, so io.c is the natural place for
it.
While we're moving the code, reduce the scope of some variables, use
compound assignment operators, and add/adjust some comments.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-4-oswald.buddenhagen@gmx.de>
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The FPGA access through the GPIO port does not interfere with other
sound processor register access, so there is no need to subject it to
emu_lock. And after moving all FPGA access out of the interrupt handler,
it does not need to be IRQ-safe, either.
What's more, attaching the dock causes a firmware upload, which takes
several seconds. We really don't want to disable IRQs for this long, and
even less also have someone else spin with IRQs disabled waiting for us.
Therefore, use a mutex for FPGA access locking.
This makes the code somewhat more noisy, as we need to wrap bigger
sections into the mutex, as it needs to enclose the spinlocks.
The latter has the "side effect" of fixing dock FPGA programming in a
corner case: a really badly timed mixer access right between entering
FPGA programming mode and uploading the netlist would mess up the
protocol.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-5-oswald.buddenhagen@gmx.de>
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The actual event processing was already done by workqueue items. We can
move the event dispatching there as well, rather than doing it already
in the interrupt handler callback.
This change has a rather profound "side effect" on the reliability of
the FPGA programming: once we enter programming mode, we must not issue
any snd_emu1010_fpga_{read,write}() calls until we're done, as these
would badly mess up the programming protocol. But exactly that would
happen when trying to program the dock, as that triggers GPIO interrupts
as a side effect. This is mitigated by deferring the actual interrupt
handling, as workqueue items are not re-entrant.
To avoid scheduling the dispatcher on non-events, we now explicitly
ignore GPIO IRQs triggered by "uninteresting" pins, which happens a lot
as a side effect of calling snd_emu1010_fpga_{read,write}().
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-4-oswald.buddenhagen@gmx.de>
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Update header inclusions to follow IWYU (Include What You Use)
principle.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20240422151513.2052167-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Many modern codecs support 705.6kHz and 768kHz sample rates. Current HW
params fail to set 705.6kHz and 768kHz sample rates as these are not in the
known-rates list.
Add these new rates to the known-rates list to allow them.
Also add defines in pcm.h so that drivers can use it.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-3-pavel.hofman@ivitera.com>
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Some systems requires different max PCM Gains settings than the default.
The current default value, when running firmware is 17.5 dB, which is
used for all systems. Some systems require lower values.
Value when running without firmware is 4.5 dB and remains unchanged.
Since the gain value is dependent on Tuning and Firmware, it can
change, so it cannot be saved in _DSD. Instead we can store it inside
a configuration binary file alongside the Firmware and Tuning files.
The gain value increments in steps of 1 dB, with value 0 representing
0.5 dB. The max value is 20, which corresponds to 20.5 dB.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-2-sbinding@opensource.cirrus.com>
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Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to
be able to select PIO mode as alternative for CORB based command sending
while retaining the RIRB functionality to receive unsolicited responses.
This mode differs from the azx single_cmd mode when RIRB is disabled.
The mixed mode is needed on Lunar Lake family because it is recommended to
use Immediate Command Response (PIO mode) instead of CORB for HDA commands.
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>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-2-peter.ujfalusi@linux.intel.com>
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Only populate the ASP1 config registers in the regmap cache if the
ASP DAI is used. This prevents regcache_sync() from overwriting
these registers with their defaults when the firmware owns
control of these registers.
On a SoundWire system the ASP could be owned by the firmware to
share reference audio with the firmware on other cs35l56. Or it
can be used as a normal codec-codec interface owned by the driver.
The driver must not overwrite the registers if the firmware has
control of them.
The original implementation for this in commit 07f7d6e7a124
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") was
to still provide defaults for these registers, assuming that if
they were never reconfigured from defaults then regcache_sync()
would not write them out because they are not dirty. Unfortunately
regcache_sync() is not that smart. If the chip has not reset (so
the driver has not called regcache_mark_dirty()) a regcache_sync()
could write out registers that are not dirty.
To avoid accidental overwriting of the ASP registers, they are
removed from the table of defaults and instead are populated with
defaults only if one of the ASP DAI configuration functions is
called. So if the DAI has never been configured, the firmware is
assumed to have ownership of these registers, and the regmap cache
will not contain any entries for them.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://msgid.link/r/20240408101803.43183-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Resulting from more reverse engineering in the course of debugging.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-15-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The hardware supports S16LE and U8 samples, while U16LE and S8 (which
the driver implicitly claims to support) require sign flipping.
Note that this matters only for the GUS patch loader, as the implemented
SoundFont v2.01 spec is limited to S16LE.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The `client` parameter was not used, so eliminate it from the call
chain.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-3-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Set of changes targeting the avs-driver only. No new features, patchset
either fixes or fortifies existing code.
Patchset starts off with a fix for debugbility on ICL+ platforms which I
have forgotten to fixup when providing support for these initially.
The next two address copier module initialization, most importantly,
silence the gcc 'field-spanning write' false-positive.
The following four:
6/13 ASoC: Intel: avs: Replace risky functions with safer variants
7/13 ASoC: Intel: avs: Fix potential integer overflow
8/13 ASoC: Intel: avs: Test result of avs_get_module_entry()
9/13 ASoC: Intel: avs: Remove dead code
address problems found out by Coverity static analysis tool.
The last two worth mentioning are: recommendation from the firmware team
to wake subsystem from D0ix when starting any pipeline -and- shielding
against invalid period/buffer sizes. Audio format shall be taken into
consideration when calculating either of these.
Amadeusz Sławiński (2):
ASoC: Intel: avs: Restore stream decoupling on prepare
ASoC: Intel: avs: Add assert_static to guarantee ABI sizes
Cezary Rojewski (11):
ASoC: Intel: avs: Fix debug-slot offset calculation
ASoC: Intel: avs: Silence false-positive memcpy() warnings
ASoC: Intel: avs: Fix config_length for config-less copiers
ASoC: Intel: avs: Fix ASRC module initialization
ASoC: Intel: avs: Replace risky functions with safer variants
ASoC: Intel: avs: Fix potential integer overflow
ASoC: Intel: avs: Test result of avs_get_module_entry()
ASoC: Intel: avs: Remove dead code
ASoC: Intel: avs: Wake from D0ix when starting streaming
ASoC: Intel: avs: Init debugfs before booting firmware
ASoC: Intel: avs: Rule invalid buffer and period sizes out
sound/soc/intel/avs/avs.h | 1 +
sound/soc/intel/avs/cldma.c | 2 +-
sound/soc/intel/avs/core.c | 4 +--
sound/soc/intel/avs/icl.c | 12 ++++++---
sound/soc/intel/avs/loader.c | 6 +++--
sound/soc/intel/avs/messages.h | 47 ++++++++++++++++++++++++++++++++--
sound/soc/intel/avs/path.c | 13 ++++------
sound/soc/intel/avs/pcm.c | 34 +++++++++++++++++++++++-
sound/soc/intel/avs/probes.c | 14 ++++++----
9 files changed, 109 insertions(+), 24 deletions(-)
--
2.25.1
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
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Some of these, particularly the wm_adsp one in the immediate case, are
needed as a basis for new work.
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Adds wrapper function snd_ctl_find_id_mixer_locked(). This is
identical to snd_ctl_find_id_mixer() except that it can be called
from code that is already holding controls_rwsem.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240401100210.61277-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since we put dvc_tlv static variable to a header file it's copied to
each module that includes the header. But not all of them are actually
used it.
Fix this W=1 build warning:
include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not
used [-Wunused-const-variable=]
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/
Fixes: ae065d0ce9e3 ("ALSA: hda/tas2781: remove digital gain kcontrol")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <0e461545a2a6e9b6152985143e50526322e5f76b.1711665731.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of device-specific small fixes: a series of fixes for
TAS2781 HD-audio codec, ASoC SOF, Cirrus CS35L56 and a couple of
legacy drivers"
* tag 'sound-6.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/tas2781: remove useless dev_dbg from playback_hook
ALSA: hda/tas2781: add debug statements to kcontrols
ALSA: hda/tas2781: add locks to kcontrols
ALSA: hda/tas2781: remove digital gain kcontrol
ALSA: aoa: avoid false-positive format truncation warning
ALSA: sh: aica: reorder cleanup operations to avoid UAF bugs
ALSA: hda: cs35l56: Set the init_done flag before component_add()
ALSA: hda: cs35l56: Raise device name message log level
ASoC: SOF: ipc4-topology: support NHLT device type
ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function
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Add new flags to tplg_quirk_mask to detect and append codec/amplifier
tplg suffix to topology file name at runtime. With this feature we
could implement an enumeration entry for all boards which implement
same headphone codec regardless the speaker amplifier type.
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://msgid.link/r/20240327162408.63953-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add helper functions to get tplg suffix string for specific headphone
codec or speaker amplifier. The string could be used to compose the
default topology file name for specific headphone codec and speaker
amplifier combination.
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://msgid.link/r/20240327162408.63953-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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As we moved ssp-common files to new locations with new names, changing
the naming convention from sof_ssp_ to snd_soc_acpi_intel_.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@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://msgid.link/r/20240327162408.63953-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Moving ssp-common header file from sound/soc/intel/common directory to
include/sound directory and rename the file. Keep file content
unchanged for tracking purpose.
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://msgid.link/r/20240327162408.63953-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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No more users.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240326171134.1414462-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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No macro currently allows handling a stereo control that has left and right
in the same register and whose minimum register value is not zero. Add one
that does that.
Note that even though the snd_soc_*_volsw_range() look more appropriate
given the _range suffix, they are not suitable because they don't honor the
two shift values. The snd_soc_*_volsw() look more generic and are suitable
for the task.
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-3-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The pplcllpl/u can be used to save the Link Connection Linear Link
Position register value to be used for compensation of the LLP register
value in case the counter is not reset (after pause/resume or
stop/start without closing the stream).
The LLP can be used along with PPHCLDP to calculate delay caused by the DSP
processing for HDA links.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@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/20240321130814.4412-17-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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At the moment we cannot instantiate two dmaengine_pcms with the same
parent device, as the components will be named the same, leading to
conflicts.
Add 'name' field to the snd_dmaengine_pcm_config, and use that (if
defined) as the component name instead of deriving the component name
from the device.
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ideasonboard.com>
Link: https://msgid.link/r/20240319-xilinx-dp-audio-v2-1-92d6d3a7ca7e@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a helper function intel_nhlt_ssp_device_type() to detect the type
of specific SSP port. The result is nhlt_device_type enum type which
could be NHLT_DEVICE_BT or NHLT_DEVICE_I2S.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Péter 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>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20231127120657.19764-2-peter.ujfalusi@linux.intel.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This was a relatively calm development cycle. Most of changes are
rather small device-specific fixes and enhancements. The only
significant changes in ALSA core are code refactoring with the recent
cleanup infrastructure, which should bring no functionality changes.
Some highlights below:
Core:
- Lots of cleanups in ALSA core code with automatic kfree cleanup and
locking guard macros
- New ALSA core kunit test
ASoC:
- SoundWire support for AMD ACP 6.3 systems
- Support for reporting version information for AVS firmware
- Support DSPless mode for Intel Soundwire systems
- Support for configuring CS35L56 amplifiers using EFI calibration
data
- Log which component is being operated on as part of power
management trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
HD- and USB-audio:
- More Cirrus HD-audio codec support
- TAS2781 HD-audio codec fixes
- Scarlett2 mixer fixes
Others:
- Enhancement of virtio driver for audio control supports
- Cleanups of legacy PM code with new macros
- Firewire sound updates"
* tag 'sound-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (307 commits)
ALSA: usb-audio: Stop parsing channels bits when all channels are found.
ALSA: hda/tas2781: remove unnecessary runtime_pm calls
ALSA: hda/realtek - ALC236 fix volume mute & mic mute LED on some HP models
ALSA: aaci: Delete unused variable in aaci_do_suspend
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range
ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values
ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection
ALSA: hda/tas2781: restore power state after system_resume
ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend
ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend
ALSA: hda/tas2781: add lock to system_suspend
ALSA: hda/tas2781: use dev_dbg in system_resume
ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops
platform/x86: serial-multi-instantiate: Add support for CS35L54 and CS35L57
ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57
ASoC: cs35l56: Add support for CS35L54 and CS35L57
ASoC: Intel: catpt: Carefully use PCI bitwise constants
ALSA: hda: hda_component: Include sound/hda_codec.h
ALSA: hda: hda_component: Add missing #include guards
...
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