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ASoC core code does not modify contents of 'of_phandle_args' in 'struct
snd_soc_dai_link_component', so the pointer can be made as a pointer to
const. This makes code safer, serves as clear annotation of core's
intentions and allows putting pointed structures in rodata (if ever
applicable).
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-2-8004f346ee38@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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ASoC core code does not modify contents of 'of_phandle_args' in 'struct
snd_soc_dai_driver', so the pointer can be made as a pointer to const.
This makes code safer, serves as clear annotation of core's intentions
and allows putting pointed structures in rodata (if ever applicable).
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-1-8004f346ee38@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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The core ASoC code does not modify contents of the
'auto_selectable_formats' array passed in 'struct snd_soc_dai_ops', so
make it const for code safety.
Reviewed-by: Herve Codina <[email protected]>
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Jai Luthra <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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soc-pcm.c :: soc_get_playback_capture() will indicate error
if both playback_only / capture_only were true.
Thus, graph_util_parse_link_direction() which setup playback_only /
capture_only don't need to check it.
And, its return value is not used on existing driver. Let's remove it.
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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We need this to get the i.MX platforms working in CI again.
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Clean up the top level include/drm directory by grouping all the Intel
specific files under a common subdirectory.
v2: Also change Documentation/gpu/i915.rst (Andi)
Cc: Daniel Vetter <[email protected]>
Cc: Dave Airlie <[email protected]>
Cc: Lucas De Marchi <[email protected]>
Cc: Tomas Winkler <[email protected]>
Cc: Jaroslav Kysela <[email protected]>
Cc: Takashi Iwai <[email protected]>
Acked-by: Lucas De Marchi <[email protected]>
Acked-by: Rodrigo Vivi <[email protected]>
Signed-off-by: Jani Nikula <[email protected]>
Link: https://patchwork.freedesktop.org/patch/msgid/a8c07233a8234858eb6711140482ef8db4c91cf4.1717075103.git.jani.nikula@intel.com
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Merge series from Krzysztof Kozlowski <[email protected]>:
X1E80100 CRD is the first board, which comes with four speakers, so we
still keep fixing and adding missing pieces.
The board has speaker arranged as left front+back and then right
front+back. Using default channel mapping causes front right speaker to
play left back stream.
Adjust the channel maps for frontend DAIs to fix stereo and four-channel
playback.
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Merge series from Pierre-Louis Bossart <[email protected]>:
More simplifications from Brent Lu for Chromebooks, Bard Liao for
SoundWire, and support for new Dell SKUs w/ Cirrus Logic codecs added
by Charles Keepax.
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Merge series from Sameer Pujar <[email protected]>:
The AHUB HW modules are interconnected with CIF which are capable of
supporting Channel and Sample bit format conversion. Due to this, the
I2S Client can have different Channel and Sample bit from the hw_params()
and this config is passed from CIF port of I2S DT node which can help to
perform this conversion.
- First change to split simple_fixup_sample_fmt to support returning
sample format value
- Second patch to support Tegra I2S client channel and sample format
programming based on CIF port from DT node.
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An new interface, sof_dai_get_tdm_slots(), is added for machine driver
to get tdm slot number from topology. The dai_get_param() callback
needs to support new parameter type SOF_DAI_PARAM_INTEL_SSP_TDM_SLOTS
by returning the tdm slot number of specific SSP port.
Reviewed-by: Bard Liao <[email protected]>
Signed-off-by: Brent Lu <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Fix the typo in the comment for SNDRV_PCM_RATE_KNOT
Signed-off-by: Alexandre Belloni <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Split the simple_fixup_sample_fmt() into two functions by adding
one more function named simple_util_get_sample_fmt() to return
the sample format value.
This is useful for drivers that wish to simply get the sample format
without setting the mask.
Signed-off-by: Mohan Kumar <[email protected]>
Signed-off-by: Sameer Pujar <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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snd_soc_tplg_component_load() does not modify its "*ops" argument. It
only read some values and stores it in "soc_tplg.ops".
This argument and the ops field in "struct soc_tplg" can be made const.
Signed-off-by: Christophe JAILLET <[email protected]>
Reviewed-by: Amadeusz Sławiński <[email protected]>
Link: https://msgid.link/r/f2f983e791d7f941a95556bb147f426a345d84d4.1715526069.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <[email protected]>
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There is no need for implementations of DAI set_channel_map() to modify
contents of passed arrays with actual channel mapping. Additionally,
the caller keeps full ownership of the array.
Constify these pointer arguments so the code will be safer and easier to
read (documenting the caller's ownership).
Acked-by: Charles Keepax <[email protected]>
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://msgid.link/r/20240520-asoc-x1e80100-4-channel-mapping-v4-1-f657159b4aad@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Reviewed-by: Bard Liao <[email protected]>
Reviewed-by: Péter Ujfalusi <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Pull HD-audio CONFIG_PM cleanup.
Signed-off-by: Takashi Iwai <[email protected]>
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Add HD Audio PCI ID and HDMI codec vendor ID for Intel Battlemage.
Signed-off-by: Chaitanya Kumar Borah <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
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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 <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>:
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 <[email protected]>
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Pull emu10k1 fixes from Oswald Buddenhagen
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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Update header inclusions to follow IWYU (Include What You Use)
principle.
Signed-off-by: Andy Shevchenko <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Reviewed-by: Pierre-Louis Bossart <[email protected]>
Reviewed-by: Bard Liao <[email protected]>
Reviewed-by: Liam Girdwood <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Message-ID: <[email protected]>
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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 <[email protected]>
Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Resulting from more reverse engineering in the course of debugging.
Signed-off-by: Oswald Buddenhagen <[email protected]>
Message-ID: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Message-ID: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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The `client` parameter was not used, so eliminate it from the call
chain.
Signed-off-by: Oswald Buddenhagen <[email protected]>
Message-ID: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Merge series from Cezary Rojewski <[email protected]>:
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 <[email protected]>
Reviewed-by: Takashi Iwai <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Closes: https://lore.kernel.org/oe-kbuild-all/[email protected]/
Fixes: ae065d0ce9e3 ("ALSA: hda/tas2781: remove digital gain kcontrol")
Signed-off-by: Gergo Koteles <[email protected]>
Message-ID: <0e461545a2a6e9b6152985143e50526322e5f76b.1711665731.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Brent Lu <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Brent Lu <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Brent Lu <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Link: https://msgid.link/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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