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struct gpio_chip now has callbacks for setting line values that return
an integer, allowing to indicate failures. Convert the driver to using
them.
Signed-off-by: Bartosz Golaszewski <bartosz.golaszewski@linaro.org>
Link: https://patch.msgid.link/20250408-gpiochip-set-rv-sound-v1-3-dd54b6ca1ef9@linaro.org
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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struct gpio_chip now has callbacks for setting line values that return
an integer, allowing to indicate failures. Convert the driver to using
them.
Signed-off-by: Bartosz Golaszewski <bartosz.golaszewski@linaro.org>
Link: https://patch.msgid.link/20250408-gpiochip-set-rv-sound-v1-2-dd54b6ca1ef9@linaro.org
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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struct gpio_chip now has callbacks for setting line values that return
an integer, allowing to indicate failures. Convert the driver to using
them.
Signed-off-by: Bartosz Golaszewski <bartosz.golaszewski@linaro.org>
Link: https://patch.msgid.link/20250408-gpiochip-set-rv-sound-v1-1-dd54b6ca1ef9@linaro.org
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add an of_match_table with 'realtek,alc203' to provide
DT support for the AC97 generic codec driver.
Signed-off-by: Keguang Zhang <keguang.zhang@gmail.com>
Link: https://patch.msgid.link/20250409-loongson1-ac97-v2-4-65d5db96a046@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add AC97 driver for Loongson-1 SoCs.
Signed-off-by: Keguang Zhang <keguang.zhang@gmail.com>
Link: https://patch.msgid.link/20250409-loongson1-ac97-v2-3-65d5db96a046@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Issue:
When multiple audio streams share a common BE DAI, the BE DAI
widget can be powered up before its hardware parameters are configured.
This incorrect sequence leads to intermittent pcm_write errors.
For example, the below Tegra use-case throws an error:
aplay(2 streams) -> AMX(mux) -> ADX(demux) -> arecord(2 streams),
here, 'AMX TX' and 'ADX RX' are common BE DAIs.
For above usecase when failure happens below sequence is observed:
aplay(1) FE open()
- BE DAI callbacks added to the list
- BE DAI state = SND_SOC_DPCM_STATE_OPEN
aplay(2) FE open()
- BE DAI callbacks are not added to the list as the state is
already SND_SOC_DPCM_STATE_OPEN during aplay(1) FE open().
aplay(2) FE hw_params()
- BE DAI hw_params() callback ignored
aplay(2) FE prepare()
- Widget is powered ON without BE DAI hw_params() call
aplay(1) FE hw_params()
- BE DAI hw_params() is now called
Fix:
Add BE DAIs in the list if its state is either SND_SOC_DPCM_STATE_OPEN
or SND_SOC_DPCM_STATE_HW_PARAMS as well.
It ensures the widget is powered ON after BE DAI hw_params() callback.
Fixes: 0c25db3f7621 ("ASoC: soc-pcm: Don't reconnect an already active BE")
Signed-off-by: Sheetal <sheetal@nvidia.com>
Link: https://patch.msgid.link/20250404105953.2784819-1-sheetal@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Incorrect types are used as sizeof() arguments in devm_kcalloc().
It should be sizeof(dai_link_data) for link_data instead of
sizeof(snd_soc_dai_link).
This is found by our static analysis tool.
Signed-off-by: Chenyuan Yang <chenyuan0y@gmail.com>
Link: https://patch.msgid.link/20250406210854.149316-1-chenyuan0y@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Depend on SND_SOC_CS_AMP_LIB instead of selecting it.
KUNIT_ALL_TESTS should only build tests for components that are
already being built, it should not cause other stuff to be added
to the build.
Fixes: 177862317a98 ("ASoC: cs-amp-lib: Add KUnit test for calibration helpers")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250411123608.1676462-3-rf@opensource.cirrus.com
Reviewed-by: David Gow <davidgow@google.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 set of small fixes, quirks and device ID additions that came in since
-rc1, none of them super stand out. There's also a change to Srini's
email address in MAINTAINERS.
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The test-component driver implements various stub callbacks. One
of the ones it doesn't implement is set_tdm_slot. This has no
functional impact on whether ASoC core believes test-component to
do TDM or not, it just means that any TDM configuration can't
readily be dumped for debugging purposes like it can with the other
callbacks.
Add a stub implementation to allow for this. The output uses dev_info
rather than dev_dbg, to be in line with the set_fmt stub implementation
above.
Signed-off-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20250410-test-component-tdm-slot-v1-1-9c3a7162fa7a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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If the vendor USB offload class driver is not ready/initialized before USB
SND discovers attached devices, utilize snd_usb_rediscover_devices() to
find all currently attached devices, so that the ASoC entities are notified
on available USB audio devices.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-32-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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In order to allow userspace/applications know about USB offloading status,
expose a sound kcontrol that fetches information about which sound card
and PCM index the USB device is mapped to for supporting offloading. In
the USB audio offloading framework, the ASoC BE DAI link is the entity
responsible for registering to the SOC USB layer.
It is expected for the USB SND offloading driver to add the kcontrol to the
sound card associated with the USB audio device. An example output would
look like:
tinymix -D 1 get 'USB Offload Playback Route PCM#0'
-1, -1 (range -1->255)
This example signifies that there is no mapped ASoC path available for the
USB SND device.
tinymix -D 1 get 'USB Offload Playback Route PCM#0'
0, 0 (range -1->255)
This example signifies that the offload path is available over ASoC sound
card index#0 and PCM device#0.
The USB offload kcontrol will be added in addition to the existing
kcontrols identified by the USB SND mixer. The kcontrols used to modify
the USB audio device specific parameters are still valid and expected to be
used. These parameters are not mirrored to the ASoC subsystem.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-31-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Add proper checks and updates to the USB substream once receiving a USB QMI
stream enable request. If the substream is already in use from the non
offload path, reject the stream enable request. In addition, update the
USB substream opened parameter when enabling the offload path, so the
non offload path can be blocked.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-30-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Several Qualcomm SoCs have a dedicated audio DSP, which has the ability to
support USB sound devices. This vendor driver will implement the required
handshaking with the DSP, in order to pass along required resources that
will be utilized by the DSP's USB SW. The communication channel used for
this handshaking will be using the QMI protocol. Required resources
include:
- Allocated secondary event ring address
- EP transfer ring address
- Interrupter number
The above information will allow for the audio DSP to execute USB transfers
over the USB bus. It will also be able to support devices that have an
implicit feedback and sync endpoint as well. Offloading these data
transfers will allow the main/applications processor to enter lower CPU
power modes, and sustain a longer duration in those modes.
Audio offloading is initiated with the following sequence:
1. Userspace configures to route audio playback to USB backend and starts
playback on the platform soundcard.
2. The Q6DSP AFE will communicate to the audio DSP to start the USB AFE
port.
3. This results in a QMI packet with a STREAM enable command.
4. The QC audio offload driver will fetch the required resources, and pass
this information as part of the QMI response to the STREAM enable command.
5. Once the QMI response is received the audio DSP will start queuing data
on the USB bus.
As part of step#2, the audio DSP is aware of the USB SND card and pcm
device index that is being selected, and is communicated as part of the QMI
request received by QC audio offload. These indices will be used to handle
the stream enable QMI request.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-29-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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The Qualcomm USB audio offload driver utilizes the QMI protocol to
communicate with the audio DSP. Add the necessary QMI header and field
definitions, so the QMI interface driver is able to route the QMI packet
received to the USB audio offload driver.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-28-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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The USB SND path may need to know how the USB offload path is routed, so
that applications can open the proper sound card and PCM device. The
implementation for the QC ASoC design has a "USB Mixer" kcontrol for each
possible FE (Q6ASM) DAI, which can be utilized to know which front end link
is enabled.
When an application/userspace queries for the mapped offload devices, the
logic will lookup the USB mixer status though the following path:
MultiMedia* <-> MM_DL* <-> USB Mixer*
The "USB Mixer" is a DAPM widget, and the q6routing entity will set the
DAPM connect status accordingly if the USB mixer is enabled. If enabled,
the Q6USB backend link can fetch the PCM device number from the FE DAI
link (Multimedia*). With respects to the card number, that is
straightforward, as the ASoC components have direct references to the ASoC
platform sound card.
An example output can be shown below:
Number of controls: 9
name value
Capture Channel Map 0, 0 (range 0->36)
Playback Channel Map 0, 0 (range 0->36)
Headset Capture Switch On
Headset Capture Volume 1 (range 0->4)
Sidetone Playback Switch On
Sidetone Playback Volume 4096 (range 0->8192)
Headset Playback Switch On
Headset Playback Volume 20, 20 (range 0->24)
USB Offload Playback Route PCM#0 0, 1 (range -1->255)
The "USB Offload Playback Route PCM#*" kcontrol will signify the
corresponding card and pcm device it is offload to. (card#0 pcm - device#1)
If the USB SND device supports multiple audio interfaces, then it will
contain several PCM streams, hence in those situations, it is expected
that there will be multiple playback route kcontrols created.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-27-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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The headphone jack framework has a well defined infrastructure for
notifying userspace entities through input devices. Expose a jack device
that carries information about if an offload capable device is connected.
Applications can further identify specific offloading information through
other SND kcontrols.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-26-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.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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For USB offloading situations, the AFE port start command will result in a
QMI handshake between the Q6DSP and the main processor. Depending on if
the USB bus is suspended, this routine would require more time to complete,
as resuming the USB bus has some overhead associated with it. Increase the
timeout to 3s to allow for sufficient time for the USB QMI stream enable
handshake to complete.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-24-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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The QC ADSP is able to support USB playback endpoints, so that the main
application processor can be placed into lower CPU power modes. This adds
the required AFE port configurations and port start command to start an
audio session.
Specifically, the QC ADSP can support all potential endpoints that are
exposed by the audio data interface. This includes isochronous data
endpoints, in either synchronous mode or asynchronous mode. In the latter
case both implicit or explicit feedback endpoints are supported. The size
of audio samples sent per USB frame (microframe) will be adjusted based on
information received on the feedback endpoint.
Some pre-requisites are needed before issuing the AFE port start command,
such as setting the USB AFE dev_token. This carries information about the
available USB SND cards and PCM devices that have been discovered on the
USB bus. The dev_token field is used by the audio DSP to notify the USB
offload driver of which card and PCM index to enable playback on.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-23-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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In case the USB backend device has not been initialized/probed, USB SND
device connections can still occur. When the USB backend is eventually
made available, previous USB SND device connections are not communicated to
the USB backend. Call snd_usb_rediscover_devices() to generate the connect
callbacks for all USB SND devices connected. This will allow for the USB
backend to be updated with the current set of devices available.
The chip array entries are all populated and removed while under the
register_mutex, so going over potential race conditions:
Thread#1:
q6usb_component_probe()
--> snd_soc_usb_add_port()
--> snd_usb_rediscover_devices()
--> mutex_lock(register_mutex)
Thread#2
--> usb_audio_disconnect()
--> mutex_lock(register_mutex)
So either thread#1 or thread#2 will complete first. If
Thread#1 completes before thread#2:
SOC USB will notify DPCM backend of the device connection. Shortly
after, once thread#2 runs, we will get a disconnect event for the
connected device.
Thread#2 completes before thread#1:
Then during snd_usb_rediscover_devices() it won't notify of any
connection for that particular chip index.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-19-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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In case of notifying SND platform drivers of connection events, some of
these use cases, such as offloading, require an ASoC USB backend device to
be initialized before the events can be handled. If the USB backend device
has not yet been probed, this leads to missing initial USB audio device
connection events.
Expose an API that traverses the usb_chip array for connected devices, and
to call the respective connection callback registered to the SND platform
driver.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-14-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Allow for different platforms to be notified on USB SND connect/disconnect
sequences. This allows for platform USB SND modules to properly initialize
and populate internal structures with references to the USB SND chip
device.
Tested-by: Puma Hsu <pumahsu@google.com>
Tested-by: Daehwan Jung <dh10.jung@samsung.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-13-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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With USB audio offloading, an audio session is started from the ASoC
platform sound card and PCM devices. Likewise, the USB SND path is still
readily available for use, in case the non-offload path is desired. In
order to prevent the two entities from attempting to use the USB bus,
introduce a flag that determines when either paths are in use.
If a PCM device is already in use, the check will return an error to
userspace notifying that the stream is currently busy. This ensures that
only one path is using the USB substream.
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/20250409194804.3773260-12-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Within the UAC descriptor, there is information describing the size of a
sample (bSubframeSize/bSubslotSize) and the number of relevant bits
(bBitResolution). Currently, fmt_bits carries only the bit resolution,
however, some offloading entities may also require the overall size of the
sample. Save this information in a separate parameter, as depending on the
UAC format type, the sample size can not easily be decoded from other
existing parameters.
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/20250409194804.3773260-11-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Allow for checks on a specific USB audio device to see if a requested PCM
format is supported. This is needed for support when playback is
initiated by the ASoC USB backend path.
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/20250409194804.3773260-10-quic_wcheng@quicinc.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Some vendor modules will utilize useful parsing and endpoint management
APIs to start audio playback/capture.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Puma Hsu <pumahsu@google.com>
Tested-by: Daehwan Jung <dh10.jung@samsung.com>
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20250409194804.3773260-9-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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Use vmemdup_user() to get a copy of the user buffer in wm_coeff_tlv_put().
Apart from simplifying the code and avoiding open-coding, it means we
also automatically benefit from any security enhancements in the code
behind vmemdup_user().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250410101812.1180539-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add of_match_table for aw88081 driver to make matching
between dts and driver more flexible
Signed-off-by: Weidong Wang <wangweidong.a@awinic.com>
Link: https://patch.msgid.link/20250410024953.26565-1-wangweidong.a@awinic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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On SNDRV_PCM_TRIGGER_START event, audio data pointers are not reset.
This leads to wrong data buffer usage when multiple TRIGGER_START are
received and ends to incorrect buffer usage between the user-space and
the driver. Indeed, the driver can read data that are not already set by
the user-space or the user-space and the driver are writing and reading
the same area.
Fix that resetting data pointers on each SNDRV_PCM_TRIGGER_START events.
Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio")
Cc: stable@vger.kernel.org
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20250410091643.535627-1-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some of the manually selected jack configurations will disable the
headphone clamp override. Restore this on jack removal, such that
the state is consistent for a new insert.
Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250409120717.1294528-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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ASUS platform Headset Mic was disable by default.
Assigned verb table for Mic pin will enable it.
Fixes: 7ab61d0a9a35 ("ALSA: hda/realtek: Add support for ASUS B3405 and B3605 Laptops using CS35L41 HDA")
Fixes: c86dd79a7c33 ("ALSA: hda/realtek: Add support for ASUS B5405 and B5605 Laptops using CS35L41 HDA")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/0fe3421a6850461fb0b7012cb28ef71d@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Depend on SND_HDA_CIRRUS_SCODEC and GPIOLIB instead of selecting them.
KUNIT_ALL_TESTS should only build tests that have satisfied dependencies
and test components that are already being built. It must not cause
other stuff to be added to the build.
Fixes: 2144833e7b41 ("ALSA: hda: cirrus_scodec: Add KUnit test")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250409114520.914079-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The code has been commented out ever since commit 1da177e4c3f4
("Linux-2.6.12-rc2"), remove it.
Signed-off-by: Thorsten Blum <thorsten.blum@linux.dev>
Link: https://patch.msgid.link/20250408083015.796638-2-thorsten.blum@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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strcpy() is deprecated, use strscpy() instead.
Link: https://github.com/KSPP/linux/issues/88
Cc: linux-hardening@vger.kernel.org
Signed-off-by: Thorsten Blum <thorsten.blum@linux.dev>
Link: https://patch.msgid.link/20250407090832.743255-1-thorsten.blum@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge series from srinivas.kandagatla@linaro.org:
On Lenovo ThinkPad T14s, the headset is connected via a HiFi Switch to
support CTIA and OMTP headsets. This switch is used to minimise pop and
click during headset type switching.
This patchset adds required bindings and changes to codec and dts to
tnable the regulator required to power this switch along with wiring up
gpio that control the headset switching.
Without this patchset, there will be lots of noise on headset and mic
will not we functional.
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Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
Add several improvements for the sai interface.
1.allow to set mclk rate with zero clk_id for master mode
2.add xlate_tdm_slot_mask() callback to avoid channel constrain
3.separate 'is_dsp_mode' for tx and rx
4.separate set_tdm_slot() for tx and rx
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Merge series from "Peng Fan (OSS)" <peng.fan@oss.nxp.com>:
of_gpio.h is deprecated, so update driver to use gpiod API.
The current driver use value 0 to assert reset and 1 to deassert reset.
The DTSes in tree that use the codec are using GPIO_ACTIVE_LOW.
So it is safe to use devm_gpiod_get to get GPIO descriptors and
use gpiod_set_value to configure output with value 1 means raw value
0, value 0 means raw value 1.
Note:
I not have devices to test, just my best pratice to do the convertion.
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Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Relatively small delta-wise patchset which raises max channels supported
from 8 to 16. The existing limitation is software-based, not hardware
based. The hardware, as per HDAudio specification, section 1.2.2,
(relevant register at SDnFMT, section 3.3.41) supports the
configurations for years. The avs-driver becomes the first consumer of
that configuration on the Linux kernel side.
Set starts off with update to string_helpers so that functionality added
with parse_int_array_user() can be utilized in kernel-kernel
interactions.
Follow up is rasing the cap on HDAudio-library side. The format
selection procedure found in the library is good-to-go as is.
Everything that follows these two patches is avs-driver specific:
- raise channels_max for every DAI-driver template
- provide i2s_test module parameter for testing purposes. When combined
with I2S loopback card, allows to test 16ch on most Intel hardware post
Broadwell era
- adjust TDM masks to reflect the 8 -> 16 channels change
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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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Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
As discussed in user bug report [1] & [2], it was identified that HDA
card provides wrong FE device numbers to be used with UCM, at the same
time it was requested that we change card names to better names.
This patchset does that, by adding option for going back to old
behaviour in first patch. Using existing way of enumerating FEs from
topology in second patch. Then setting new names in all cards while
providing fallback to old name. Only exception is probe card, which is
used for debug purposes only, where we just directly rename card.
Do note that patch 2, will require updated topologies if topology
exposes more than 1 FE on card. Old topologies didn't assign id field
and it defaulted to 0, however when we use this field by setting
snd_soc_component_driver::use_dai_pcm_id field, we need topologies with
distinct values in FEs. Necessary changes are provided in avsdk and
avs-topology-xml repositories ([4] & [5]). linux-firmware update will
follow as soon as this changeset is merged.
[1]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=219654
[2]
Link: https://github.com/alsa-project/alsa-ucm-conf/pull/499
[3]
Link: https://lore.kernel.org/linux-sound/20250127144445.2739017-1-amadeuszx.slawinski@linux.intel.com/
[4]
https://github.com/thesofproject/avsdk/commit/a879c8ae4ba7be53b8ed528da1361a8c62475b6e
[5]
https://github.com/thesofproject/avs-topology-xml/commit/9b94d52cbc5c1e07c8d9503c86329cd62ea4c9e7
https://github.com/thesofproject/avs-topology-xml/commit/bdbc8d6ba9ea6db67daed9cbbaed3c23ff112ecb
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Merge series from srinivas.kandagatla@linaro.org:
This two patches fixes below two issues with the VI setup.
1. Only one channel gets enabled on VI feedback patch instead of two
channels
2. recording rate is hardcoded to 8K instead dyamically setting it up.
Both of these issues are fixed in these patches.
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commit 0ecd24a6d8b2 ("ASoC: hdmi-codec: dump ELD through procfs") adds
"eld#%d" entry for sound proc. It is using DAI ID. But it is possible to
have duplicate DAI ID on same Sound Card. In such case, we will get below
error. To avoid duplicate entry name, use RTD ID instead of DAI ID.
proc_dir_entry 'card0/eld#0' already registered
WARNING: CPU: 3 PID: 74 at fs/proc/generic.c:377 proc_register+0x11c/0x1a4
Modules linked in:
CPU: 3 UID: 0 PID: 74 Comm: kworker/u33:5 Not tainted 6.14.0-rc1-next-20250206-arm64-renesas #174
Hardware name: Renesas Salvator-X 2nd version board based on r8a77951 (DT)
Workqueue: events_unbound deferred_probe_work_func
pstate: 60000005 (nZCv daif -PAN -UAO -TCO -DIT -SSBS BTYPE=--)
pc : proc_register+0x11c/0x1a4
ata1: SATA link down (SStatus 0 SControl 300)
lr : proc_register+0x11c/0x1a4
sp : ffff8000847db880
x29: ffff8000847db880 x28: 0000000000000000 x27: ffff0004c3403c98
x26: 0000000000000005 x25: ffff0004c14b03e4 x24: 0000000000000005
x23: ffff0004c361adb8 x22: ffff800082f24860 x21: ffff0004c361ad00
x20: ffff0004c14b0300 x19: ffff0004c14b02c0 x18: 00000000ffffffff
x17: 0000000000000000 x16: 00400034b5503510 x15: ffff8001047db447
x14: 0000000000000000 x13: 6465726574736967 x12: ffff800082e66d30
x11: 000000000000028e x10: ffff800082e66d30 x9 : 00000000ffffefff
x8 : ffff800082ebed30 x7 : 0000000000017fe8 x6 : 0000000000000000
x5 : 80000000fffff000 x4 : 0000000000000000 x3 : 0000000000000000
x2 : 0000000000000000 x1 : 0000000000000000 x0 : ffff0004c15b3600
Call trace:
proc_register+0x11c/0x1a4 (P)
proc_create_data+0x3c/0x60
snd_info_register+0xd0/0x130
snd_info_register+0x30/0x130
snd_info_card_register+0x1c/0xbc
snd_card_register+0x194/0x1ec
snd_soc_bind_card+0x7f8/0xad0
snd_soc_register_card+0xe8/0xfc
devm_snd_soc_register_card+0x48/0x98
audio_graph_parse_of+0x1c4/0x1f8
graph_probe+0x6c/0x80
...
Fixes: 0ecd24a6d8b2 ("ASoC: hdmi-codec: dump ELD through procfs")
Reported-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/87a58roatw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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For i2s and DMIC copiers constraint stream capabilities based on
available NHLT configuration. This allows topology to provide generic
configuration that handles more hardware, while filtering unavailable
ones at runtime.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20250407130851.1726800-1-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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release_firmware() checks for NULL pointers internally.
Remove unneeded NULL check for fmw here.
Signed-off-by: Chen Ni <nichen@iscas.ac.cn>
Link: https://patch.msgid.link/20250407062725.2771916-1-nichen@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
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