| Age | Commit message (Collapse) | Author |
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snd_soc_unregister_component_by_driver()
We have below 2 functions, but these are very similar
(A) snd_soc_unregister_component_by_driver()
(B) snd_soc_unregister_component()
(A) void snd_soc_unregister_component_by_driver(...)
{
...
(a) mutex_lock(&client_mutex);
^ (X) component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
| if (!component) ^^^^^^^^^^^^^^^^^^^^^^
| goto out;
(b)
| snd_soc_del_component_unlocked(component);
v
out:
(c) mutex_unlock(&client_mutex);
}
(B) void snd_soc_unregister_component_by_driver(...)
{
(a) mutex_lock(&client_mutex);
^ while (1) {
| (X) struct snd_soc_component *component = snd_soc_lookup_component_nolocked(dev, NULL);
| ^^^^
(b) if (!component)
| break;
|
| snd_soc_del_component_unlocked(component);
v }
(c) mutex_unlock(&client_mutex);
}
Both are calling lock (a), find component and remove it (b), and
unlock (c). The big diff is whether use driver name for lookup() or
not (X).
Merge these into snd_soc_unregister_component_by_driver() (B), and
snd_soc_unregister_component_by_driver() (A) can be macro.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87h61qy2vn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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s/devince/device/
It's used only internally, so no any behavior changes.
Fixes: 37e0e14128e0 ("ALSA: ump: Support UMP Endpoint and Function Block parsing")
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://patch.msgid.link/20250511141147.10246-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is no snd_wavefront_xxx() implementation, and no one is using it.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87msbmpqls.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is no snd_gf1_lfo_xxx() implementation, and no one is using it.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87o6w2pqm8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We don't need these definitions. Remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87ldr6pqlh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is no snd_soc_disconnect_sync() implementation, and no one is
using it. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87jz6qpql7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_gus_use_dec(), snd_gus_use_inc() and snd_gf1_print_voice_registers()
last uses were removed in 2007 by
commit e5723b41abe5 ("[ALSA] Remove sequencer instrument layer")
Remove them.
While there, remove big #if 0 blocks next to the code being deleted.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Link: https://patch.msgid.link/20250508000225.195766-1-linux@treblig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_hdac_stream_get_spbmaxfifo() was originally added in 2015
in commit ee8bc4df1b5a ("ALSA: hdac: Add support to enable SPIB for hdac
ext stream")
when it was originally called snd_hdac_ext_stream_set_spbmaxfifo,
it was renamed snd_hdac_ext_stream_get_spbmaxfifo shortly after
and was finally renamed to snd_hdac_stream_get_spbmaxfifo in 2022.
But it was never used.
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Link: https://patch.msgid.link/20250505011037.340592-1-linux@treblig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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"Acoustic Tuning" debugfs node is a bridge to the acoustic tuning tool
which can tune the chips' acoustic effect.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20250507094616.210-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
CS35L63 is a Mono Class-D PC Smart Amplifier, with Speaker Protection
and Audio Enhancement Algorithms.
CS35L63 uses a similar control interface to CS35L56 so support for
it can be added into the CS35L56 driver.
CS35L63 only has SoundWire and I2C control interfaces.
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Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
SOF will load the function topologies by default. However, user may want
to use the monolithic topology. Add a flag amd a module parameter to
allow user specify the topology or not using function topologies.
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Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>:
Fix a small bug that can cause the sof_sdw machine driver to fail probe
after the first time it has probed. Also do some minor tidy up on the
handling of the platform_component of the dai links.
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Merge series from "Peng Fan (OSS)" <peng.fan@oss.nxp.com>:
This patchset is separate from [1], and not merging changes in one
patch. So separate changes into three patches for each chip.
- sort headers
- Drop legacy platform support
- Convert to GPIO descriptors
of_gpio.h is deprecated, update the driver to use GPIO descriptors.
- Use devm_gpiod_get_optional to get GPIO descriptor with default
polarity GPIOD_OUT_LOW, set consumer name.
- Use gpiod_set_value_cansleep to configure output value.
I not have platforms to test, just do the patches with my best efforts,
and make build pass.
[1] https://lore.kernel.org/all/20250408-asoc-gpio-v1-0-c0db9d3fd6e9@nxp.com/
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There is no in-tree user of "include/sound/cs42l52.h", so move
'struct cs42l52_platform_data ' to cs42l52.c and remove the header file.
And platform data is mostly for legacy platforms that create devices
non using device tree. So drop cs42l52.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-8-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is no in-tree user of "include/sound/cs42l56.h", so move
'struct cs42l73_platform_data ' to cs42l73.c and remove the header file.
And platform data is mostly for legacy platforms that create devices
non using device tree. So drop cs42l73.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-5-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is no in-tree user of "include/sound/cs42l56.h", so move
'struct cs42l56_platform_data' to cs42l56.c and remove the header file.
And platform data is mostly for platforms that create
devices non using device tree. CS42L56 is a discontinued product,
there is less possibility that new users will use legacy method
to create devices. So drop cs42l56.h to prepare using GPIOD API.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250506-csl42x-v3-2-e9496db544c4@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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SOF driver will load required function topologies dynamically. However,
we prefer using the monolithic topology. Add a flag to allow user not
using the function topologies.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://patch.msgid.link/20250506113311.45487-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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On CS35L63 the DIE_STS registers are populated by the Firmware from
OTP, so the driver can read these registers directly, rather than
obtaining them from OTP.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250407151842.143393-6-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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CS35L63 uses a similar control interface to CS35L56 so support for
it can be added into the CS35L56 driver.
New regmap configs have been added to support CS35L63.
CS35L63 only has SoundWire and I2C control interfaces.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250407151842.143393-5-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Registers to set Mute, Volume and Posture are inside firmware,
which means they should be added to the list of registers set inside
firmware, in case they vary across Device or Revision.
These three registers are also used for controls, so additional
handling is required to be able to obtain and set the register inside
ALSA controls.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250407151842.143393-4-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Firmware based registers may be different addresses across different
device ids and revision ids. Create a structure to store and access
these addresses.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250407151842.143393-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Philipp Stanner <phasta@kernel.org>:
A year ago we spent quite some work trying to get PCI into better shape.
Some pci_ functions can be sometimes managed with devres, which is
obviously bad. We want to provide an obvious API, where pci_ functions
are never, and pcim_ functions are always managed.
Thus, everyone enabling his device with pcim_enable_device() must be
ported to pcim_ functions. Porting all users will later enable us to
significantly simplify parts of the PCI subsystem. See here [1] for
details.
This patch series does that for sound.
Feel free to squash the commits as you see fit.
P.
[1] https://elixir.bootlin.com/linux/v6.14-rc4/source/drivers/pci/devres.c#L18
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There is no point in passing num_platforms into
asoc_sdw_init_simple_dai_link(). Firstly, as a single pointer for the
component name is passed in only a single string can be passed and
secondly if it is a complex DAI with multiple platforms it would make
more sense to use asoc_sdw_init_dai_link().
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250505141409.2614010-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_jack_set_parent() was added as part of 2008's
commit e76d8ceaaff9 ("ALSA: Add jack reporting API")
but hasn't been used.
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250502235219.1000429-6-linux@treblig.org
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snd_device_get_state() last use was removed in 2022 by
commit 7e1afce5866e ("ALSA: usb-audio: Inform the delayed registration more
properly")
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250502235219.1000429-5-linux@treblig.org
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snd_dmaengine_pcm_open_request_chan() last use was removed in 2022's
commit b401d1fd8053 ("ASoC: pxa: remove unused board support")
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250502235219.1000429-3-linux@treblig.org
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The last use of snd_pcm_rate_range_to_bits() was removed in 2016 by
commit b6b6e4d670c9 ("ASoC: topology: Fix setting of stream rates, rate_min
and rate_max")
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250502235219.1000429-2-linux@treblig.org
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Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
We are using dummy component/dlc, but didn't have its check funciton.
This patch adds it.
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We have snd_soc_xxx_is_dummy() functions, but not for dlc.
Let's add it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87cydc8vup.wl-kuninori.morimoto.gx@renesas.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: Fixes for v6.15
A moderately large batch of fixes for v6.15, many driver specific
including cleanups for the enabling of the Cirrus KUnit tests and a fix
for a nasty crash on resume on AMD systems. We also have one core fix,
for an ordering issue between DAPM and DPCM which could leave things
incorrectly unpowered.
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The volume control for cs35l56 speakers has a maximum gain of +12 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the case of systems which use the soc_sdw_* driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 400, which is 0 dB maximum volume inside this driver.
The volume limit should be applied to both soundwire and soundwire
bridge configurations.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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SND_SOC_TAS2781_FMWLIB
Most codes in tas2781_spi_fwlib.c are same as tas2781-fmwlib.c, mainly for
firmware parsing, only differece is the register reading, bit update and
book switching in i2c and spi. The main purpose of this patch is for code
cleaup and arrange the shared part for i2c and spi.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250429111055.567-1-shenghao-ding@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The conversion function from MIDI 1.0 to UMP packet contains an
internal buffer to keep the incoming MIDI bytes, and its size is 4, as
it was supposed to be the max size for a MIDI1 UMP packet data.
However, the implementation overlooked that SysEx is handled in a
different format, and it can be up to 6 bytes, as found in
do_convert_to_ump(). It leads eventually to a buffer overflow, and
may corrupt the memory when a longer SysEx message is received.
The fix is simply to extend the buffer size to 6 to fit with the SysEx
UMP message.
Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support")
Reported-by: Argusee <vr@darknavy.com>
Link: https://patch.msgid.link/20250429124845.25128-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is no in-tree user to create the device using platform data
'struct tpa6130a2_platform_data', so drop the dead code.
Signed-off-by: Peng Fan <peng.fan@nxp.com>
Link: https://patch.msgid.link/20250414-asoc-tpa6130a2-v1-2-5f4052e656a0@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
formance low-power audio DSP with analog and
PDM digital inputs and support for low-power always-on voice-trigger
functionality.
This series adds the devicetree bindings and the ASoC codec driver.
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Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
A codec endpoint may not be used. We could check the present SDCA
functions to know if the endpoint is used or not. Skip the endpoint
which is not used. And load the topology dynamically for each endpoint.
With this feature, we don't need to use the quirk to determine the
existence of the optional codec DAIs.
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Add a codec driver for the Cirrus Logic CS48L32 audio DSP.
The CS48L32 is a low-power audio DSP with microphone inputs for
"Always on Voice" (i.e. voice trigger) and voice command processing.
It has a programmable Halo Core DSP and a variety of power-efficient
fixed-function audio processors, with configurable digital mixing
and routing.
There are two I2S/TDM audio serial ports.
Four analogue inputs are available through IN1. These feed into a
2-channel ADC through an analogue mux. There is an ALSA control for
each IN1 ADC channel to select which analogue input to use.
A dedicated digital mic (DMIC) PDM input is available on IN2.
Two PDM outputs can feed DMIC inputs on another codec or a host DMIC/PDM
input.
An on-board regulator provides a power supply or bias voltage to
attached microphones. Three switchable MICBIAS outputs are fed from this
allowing only the microphone in use to be powered-up. There are DAPM
widgets for these outputs: MICBIAS1A, MICBIAS1B and MICBIAS1C. The machine
driver must create a DAPM route from the required MICBIAS1x widget to the
INn input widgets to make the MICBIAS switch on when the audio input is
powered-up. For example if the microphone feeding CS48L32 pin IN1LN_1 is
powered from MICBIAS1A, the machine driver must create the path:
(sink) IN1LN_1 <----- (source) MICBIAS1A
Co-developed-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Co-developed-by: Qi Zhou <qi.zhou@cirrus.com>
Signed-off-by: Qi Zhou <qi.zhou@cirrus.com>
Co-developed-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250415115016.505777-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add the __counted_by() compiler attribute to the flexible array member
'data' to improve access bounds-checking via CONFIG_UBSAN_BOUNDS and
CONFIG_FORTIFY_SOURCE.
No functional changes intended.
Signed-off-by: Thorsten Blum <thorsten.blum@linux.dev>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250415090354.92211-2-thorsten.blum@linux.dev
Signed-off-by: Mark Brown <broonie@kernel.org>
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We always use a single topology that contains all PCM devices belonging
to a machine configuration.
However, with SDCA, we want to be able to load function topologies based
on the supported device functions. This change is in preparation for
loading those function topologies.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250414063239.85200-4-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Create a USB BE component that will register a new USB port to the ASoC USB
framework. This will handle determination on if the requested audio
profile is supported by the USB device currently selected.
Check for if the PCM format is supported during the hw_params callback. If
the profile is not supported then the userspace ALSA entity will receive an
error, and can take further action.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-25-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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USB SND needs to know how the USB offload path is being routed. This would
allow for applications to open the corresponding sound card and pcm device
when it wants to take the audio offload path. This callback should return
the mapped indexes based on the USB SND device information.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-18-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Expose API for creation of a jack control for notifying of available
devices that are plugged in/discovered, and that support offloading. This
allows for control names to be standardized across implementations of USB
audio offloading.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-17-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Introduce a helper to check if a particular PCM format is supported by the
USB audio device connected. If the USB audio device does not have an
audio profile which can support the requested format, then notify the USB
backend.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-16-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Some platforms may have support for offloading USB audio devices to a
dedicated audio DSP. Introduce a set of APIs that allow for management of
USB sound card and PCM devices enumerated by the USB SND class driver.
This allows for the ASoC components to be aware of what USB devices are
available for offloading.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-15-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Add an USB jack type, in order to support notifying of a valid USB audio
device. Since USB audio devices can have a slew of different
configurations that reach beyond the basic headset and headphone use cases,
classify these devices differently.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-8-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
The patchset is fairly straightforward - add support for Automotive
platforms based on new DSP architecture, Frisco Lake (FCL), a
PantherLake (PTL)-based platform is an example of. The cAVS
architecture which all Intel AudioDSP followed for years ends with
RaptorLake familty. Like all the major updates, this one received new
name too - Audio Context Engine (ACE).
While the range of improvements and changes on the firmware/hardware
side is large, software survives this evolution without need of any
major refactoring. Additional hardware changes brought with LunarLake
(LNL, ACE 2.0) call for update in PCM-area. The GPDMAs previously
utilized for non-HDAudio transfer types are no longer there, everything
is running through HDAudio LINK on the Back-End side now.
In terms of code, the mtl.c file, provided with patch 05 'ASoC: Intel:
avs: PTL-based platforms support' hosts largest number of new handlers -
new IRQ and INT control and DSP-cores management. Combined with lnl.c
and ptl.c which layer the architecture changes done over ACE
generations, provide support for PTL-based platforms e.g.: FCL.
The inheritance in summary:
mtl.c <- lnl.c <- ptl.c
The functional update to HDAudio library is there to help avs-driver
read certain capabilities directly from the hardware. Once the pointer
to LINK is obtained, there is no need to call AudioDSP firmware to get
the caps.
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Starting with LunarLake (LNL) and onward, some hardware capabilities are
visible to the sound driver directly. At the same time, these may no
longer be visible to the AudioDSP firmware. Update resource allocation
function to rely on the registers when possible.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Link: https://patch.msgid.link/20250407112352.3720779-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Starting with LNL platform, Intel HDAudio Links carry IDs specifying
non-HDAudio transfer type they help facilitate e.g.: 0xC0 for I2S as
defined by AZX_REG_ML_LEPTR_ID_INTEL_SSP.
The mechanism accounts for LEPTR register as it is Reserved if
LCAP.ALT for given Link equals 0.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Link: https://patch.msgid.link/20250407112352.3720779-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since commit e894efef9ac7 ("ASoC: core: add support to card rebind")
there is a support for card rebind. The support is only partial though.
Let's consider the following scenarios both of which aim to enumerate a
sound card:
1)
snd_soc_add_component(comp1);
(...)
snd_soc_register_card(card1);
2)
snd_soc_register_card(card1);
(...)
snd_soc_add_component(comp1);
For the sake of simplicity, let comp1 be the last dependency needed for
the card1 to enumerate.
Case 1) will end up succeeding whereas 2) is a certain fail -
snd_soc_bind_card() does not honor unbind_card_list so even a non-fatal
return code of EPROBE_DEFER will cause the card to collapse. Given the
typical usecase of platform_device serving as a card->dev and its
probe() ending with:
int carddev_probe(struct platform_device *pdev)
{
(...)
return devm_snd_soc_register_card(dev, card);
}
failure to register card triggers device_unbind_cleanup() -
really_probe() in dd.c.
To allow for card registration to be deferred while being friendly
towards existing users of devm_snd_soc_register_card(), add new
card->devres_dev field, and devm_xxx() variants for card registration:
devm_snd_soc_register_deferrable_card() (external)
devm_snd_soc_bind_card() (internal)
In essence, if requested, devm_snd_soc_bind_card() replaces
snd_soc_bind_card(). The rebind procedure takes care of destroying
old devres before attempting the new bind. This makes sure nothing is
left hanging if binding fails and card becomes unbound but is still
registered to the ASoC framework.
To allow snd_soc_bind_card() to be reused by the deferrable friends,
move 'client_mutex' locking to the function's callers and select between
devm_xxx and non-devm_xxx variants of snd_soc_bind_card() based on
card->devres_dev.
On top of the feature, the refactoring brings two benefits:
a) single lock/unlock of 'client_mutex' in snd_soc_add_component()
instead of ambiguous unlock and immediate lock in
snd_soc_try_rebind_card()
b) all unbind_card_list manipulations done under 'client_mutex'
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250404101622.3673850-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire
Pull soundwire updates from Vinod Koul:
- Support for SoundWire Bulk Register Access (BRA) protocol in core
along with Intel driver support and ASoC bits required
- AMD driver updates and support for ACP 7.0 and 7.1 platforms
* tag 'soundwire-6.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire: (28 commits)
soundwire: take in count the bandwidth of a prepared stream
ASoC: rt711-sdca: add DP0 support
soundwire: debugfs: add interface for BPT/BRA transfers
ASoC: SOF: Intel: hda-sdw-bpt: add CHAIN_DMA support
soundwire: intel_ace2x: add BPT send_async/wait callbacks
soundwire: intel: add BPT context definition
ASoC: SOF: Intel: hda-sdw-bpt: add helpers for SoundWire BPT DMA
soundwire: intel_auxdevice: add indirection for BPT send_async/wait
soundwire: cadence: add BTP/BRA helpers to format data
soundwire: bus: add bpt_stream pointer
soundwire: bus: add send_async/wait APIs for BPT protocol
soundwire: stream: reuse existing code for BPT stream
soundwire: stream: special-case the bus compute_params() routine
soundwire: stream: extend sdw_alloc_stream() to take 'type' parameter
soundwire: extend sdw_stream_type to BPT
soundwire: cadence: add BTP support for DP0
Documentation: driver: add SoundWire BRA description
soundwire: amd: change the log level for command response log
soundwire: slave: fix an OF node reference leak in soundwire slave device
soundwire: Use str_enable_disable-like helpers
...
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