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Pull the 6.5-devel branch for upstreaming.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On HP EliteBook 835/845/845W G10, the audio LEDs can be enabled by
ALC285_FIXUP_HP_MUTE_LED. So use it accordingly.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Fixes: 3e10f6ca76c4 ("ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops")
Link: https://lore.kernel.org/r/20230626130301.301712-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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According to CTA 861 the channel/speaker allocation info in the
audio infoframe only applies to uncompressed (PCM) audio streams.
The channel count info should indicate the number of channels
in the transmitted audio, which usually won't match the number of
channels used to transmit the compressed bitstream.
Some devices (eg some Sony TVs) will refuse to decode compressed
audio if these values are not set correctly.
To fix this we can simply set the channel count to 0 (which means
"refer to stream header") and set the channel/speaker allocation to 0
as well (which would mean stereo FL/FR for PCM, a safe value all sinks
will support) when transmitting compressed audio.
Signed-off-by: Matthias Reichl <hias@horus.com>
Link: https://lore.kernel.org/r/20230624165232.5751-1-hias@horus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The SADs of compressed formats contain the channel and sample rate
info of the audio data inside the compressed stream, but when
building constraints we must use the rates and channels used to
transport the compressed streams.
eg 48kHz 6ch EAC3 needs to be transmitted as a 2ch 192kHz stream.
This patch fixes the constraints for the common AC3 and DTS formats,
the constraints for the less common MPEG, DSD etc formats are copied
directly from the info in the SADs as before as I don't have the specs
and equipment to test those.
Signed-off-by: Matthias Reichl <hias@horus.com>
Link: https://lore.kernel.org/r/20230624165216.5719-1-hias@horus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull one more API update for UMP core.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Yet more preliminary work for the upcoming USB gadget support.
Now export the helpers to convert between legacy MIDI1 and UMP data
for handling the MIDI 1.0 USB interface. The header file is moved to
include/sound.
The API functions are slightly changed, so that they can be used
without the direct access to snd_ump object. The allocation is done
in ump.c itself as it's a simple kcalloc().
Link: https://lore.kernel.org/r/20230623075530.10976-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This requires a patched ACPI table or a firmware from ASUS to work because
the system does not come with the _DSD field for the CSC3551.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217550
Signed-off-by: Matthew Anderson <ruinairas1992@gmail.com>
Tested-by: Philip Mueller <philm@manjaro.org>
Link: https://lore.kernel.org/r/20230621161714.9442-1-ruinairas1992@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fix for v6.4
A fix for a typoed iterator in the Intel Soundwire driver, fairly simple
on inspection though not reviewed by Intel.
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Adds the required quirk to enable the Cirrus amp and correct pins
on the ASUS ROG GV601V series.
While this works if the related _DSD properties are made available, these
aren't included in the ACPI of these laptops (yet).
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Link: https://lore.kernel.org/r/20230621085715.5382-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a small patch set to change the UMP core for the upcoming
gadget driver support. Basically exporting a couple of helper
functions and adding a flag to suppress the internal UMP handling.
No functional changes by those alone.
Link: https://lore.kernel.org/r/20230621110241.4751-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is another preliminary patch for USB MIDI 2.0 gadget driver.
Export the currently local snd_ump_receive_ump_val(). It can be used
by the gadget driver for processing the UMP data.
Link: https://lore.kernel.org/r/20230621110241.4751-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is another preliminary patch for USB MIDI 2.0 gadget driver.
Add a new flag, no_process_stream, to snd_ump for suppressing the UMP
Stream message handling in UMP core.
Link: https://lore.kernel.org/r/20230621110241.4751-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a preliminary patch for MIDI 2.0 USB gadget driver.
Export a new helper to allow changing the current MIDI protocol from
the outside.
Link: https://lore.kernel.org/r/20230621110241.4751-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now that the driver core allows for struct class to be in read-only
memory, making all 'class' structures to be declared at build time
placing them into read-only memory, instead of having to be dynamically
allocated at load time.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Ivan Orlov <ivan.orlov0322@gmail.com>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Geoff Levand <geoff@infradead.org>
Cc: Thierry Reding <treding@nvidia.com>
Cc: "Uwe Kleine-König" <u.kleine-koenig@pengutronix.de>
Cc: alsa-devel@alsa-project.org
Suggested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20230620175633.641141-2-gregkh@linuxfoundation.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For gapless playback it is possible that each track can have different
codec profile with same decoder, for example we have WMA album,
we may have different tracks as WMA v9, WMA v10 and so on
Or if DSP's like QDSP have abililty to switch decoders on single stream
for each track, then this call could be used to set new codec parameters.
Existing code does not allow to change this profile while doing gapless
playback.
Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along
some additional checks to enforce proper state machine.
With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS
anytime after setting next track and additional check in write should
also ensure that params are set before writing new data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20230619092805.21649-1-srinivas.kandagatla@linaro.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adds the required quirk to enable the Cirrus amp and correct pins
on the ASUS ROG G634Z series.
While this works if the related _DSD properties are made available, these
aren't included in the ACPI of these laptops (yet).
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Link: https://lore.kernel.org/r/20230619060320.1336455-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The loop checking for multiple different devices on a single sdw link
contains a typo accidentally using i twice instead of j. Correct to the
correct index variable.
Fixes: dc5a3e60a4b5 ("ASoC: Intel: sof_sdw: append codec type to dai link name")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230614142116.1059677-1-ckeepax@opensource.cirrus.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 v6.4
A couple more fixes for v6.4, one fixing a misleading error log and
another stopping us seeing spurious failures setting the master volume
on some Tegra systems introduced by a change to how we calculate delay
times.
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This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for
HEM devices which supports native DSD. Prior to this change Linux kernel
was not enabling native DSD playback for HEM devices, and as a result,
DSD audio was being converted to PCM "on the fly". HEM devices,
when connected to the system, would only play audio in PCM format,
even if the source material was in DSD format. With the addition of new
VENDOR_FLG in the quircks.c file, the devices are now correctly
recognized, and raw DSD data is transmitted to the device,
allowing for native DSD playback.
Signed-off-by: Lukasz Tyl <ltyl@hem-e.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230614122524.30271-1-ltyl@hem-e.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As reported in the bugzilla below, the PM resume of a UAC3 device may
fail due to the incomplete power state change, stuck at D1. The
reason is that the driver expects the full D0 power state change only
at hw_params, while the normal PCM resume procedure doesn't call
hw_params.
For fixing the bug, we add the same power state update to D0 at the
prepare callback, which is certainly called by the resume procedure.
Note that, with this change, the power state change in the hw_params
becomes almost redundant, since snd_usb_hw_params() doesn't touch the
parameters (at least it tires so). But dropping it is still a bit
risky (e.g. we have the media-driver binding), so I leave the D0 power
state change in snd_usb_hw_params() as is for now.
Fixes: a0a4959eb4e9 ("ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217539
Link: https://lore.kernel.org/r/20230612132818.29486-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be
called concurrently from different code paths, we have no proper data
protection against races. Introduce open_mutex to each seq_oss_midi
object for avoiding the races.
Reported-by: "Gong, Sishuai" <sishuai@purdue.edu>
Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu
Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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smatch error:
sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error:
we previously assumed 'rac97' could be null (see line 2072)
remove redundant assignment, return error if rac97 is NULL.
Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*")
Signed-off-by: Su Hui <suhui@nfschina.com>
Link: https://lore.kernel.org/r/20230615021732.1972194-1-suhui@nfschina.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 3ed2b549b39f ("ALSA: pcm: fix wait_time calculations") corrected
the PCM wait_time calculations and in doing so reduced the calculated
wait_time. This exposed an issue with the Tegra Master Volume Control
(MVC) device where the reduced wait_time caused the MVC to fail. For now
fix this by setting the default wait_time for Tegra to be 500ms.
Fixes: 3ed2b549b39f ("ALSA: pcm: fix wait_time calculations")
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20230613093453.13927-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The file is called spdif-in, but we abused it to show only sample rates
from various sources. Rectify it as far as possible (the FPGA doesn't
give us a lot of information).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-10-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixes a tentative FIXME. Because we can.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-9-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The timer was presuming a fixed 48 kHz word clock, like the rest of the
code.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some info-querying code still used hw.resolution directly instead of
calling snd_timer_hw_resolution(), thus missing a possible
hw.c_resolution callback. This patch rectifies that.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is only a very partial fix - the frequency-dependent envelope & LFO
register values aren't adjusted.
But I'm not sure they were even correct at 48 kHz to start with, as most
of them are precalculated by common code which assumes an EMU8K-specific
44.1 kHz word clock, and it seems somewhat unlikely that the hardware's
register interpretation was adjusted to compensate for the different
word clock.
In any case I'm not going to spend time on fixing that, as this code is
unlikely to be actually used by anyone today.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now that we know the actual word clock, we can:
- Put the resulting rate into the hardware info
- At 44.1 kHz word clock shift the rate for the pitch calculations,
which presume a 48 kHz word clock
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The value isn't used yet; the subsequent commits will do that.
This ignores the existence of rates above 48 kHz, which is fine, as the
hardware will just switch to the fallback clock source when fed with a
rate which is incompatible with the base clock multiplier, which
currently is always x1.
The sample rate display in /proc spdif-in is adjusted to reflect our
understanding of the input rates.
This is tested only with an 0404b card without sync card, so there is a
lot of room for improvement.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The actually available clock sources depend on the available audio input
ports and dedicated clock input ports.
This includes refactoring the code to be data-driven to remain
manageable.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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So far, we set the fallback as a side effect of setting the source. But
the fallback makes no sense at all when an internal clock is selected.
Defaulting to 48k for S/PDIF & ADAT makes sense, but as that is the
global default and we're not changing it automatically any more, it's
just fine to leave it entirely to the explicit setting.
This changes the name of the pre-existing control to something more
appropriate (regardless of the split), so users will need to adjust
their mixer settings.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As the updated MIDI 2.0 spec has been published freshly, this is a
catch up to add the support for new specs, especially UMP v1.1
features, on Linux kernel.
The new UMP v1.1 introduced the concept of Function Blocks (FB), which
is a kind of superset of USB MIDI 2.0 Group Terminal Blocks (GTB).
The patch set adds the support for FB as the primary information
source while keeping the parse of GTB as fallback. Also UMP v1.1
supports the groupless messages, the protocol switch, static FBs, and
other new fundamental features, and those are supported as well.
Link: https://www.midi.org/midi-articles/details-about-midi-2-0-midi-ci-profiles-and-property-exchange
Link: https://lore.kernel.org/r/20230612081054.17200-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There have been a few enhancements for the new UMP 1.1 features.
Update the documentation accordingly.
Link: https://lore.kernel.org/r/20230612081054.17200-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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UMP v1.1 spec allows to inform whether the function blocks are static
and not dynamically updated. Add a new flag bit to
snd_ump_endpoint_info to reflect that attribute, too.
The flag is set when a USB MIDI device is still in the old MIDI 2.0
without UMP 1.1 support. Then the driver falls back to GTBs, and they
are supposed to be static-only.
Link: https://lore.kernel.org/r/20230612081054.17200-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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UMP v1.1 supports the protocol switch via a UMP Stream message. When
it's received, we need to take care of the midi_version field in the
corresponding sequencer client, too.
This patch introduces a new ops to notify the protocol change to
snd_seq_ump_ops for handling it.
Link: https://lore.kernel.org/r/20230612081054.17200-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For allowing applications to track the FB active changes, this patch
adds the notification from the system port at each time a FB change is
handled and the active flag or re-grouping happens.
Link: https://lore.kernel.org/r/20230612081054.17200-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch implements the handling of the dynamic update of FB info.
When the FB info update is received after the initial parsing, it
means the dynamic FB info update. We compare the result, and if the
actual update is detected, it's notified via a new ops,
notify_fb_change, to the sequencer client, and the corresponding
sequencer ports are updated accordingly.
Link: https://lore.kernel.org/r/20230612081054.17200-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The UMP Utility and Stream messages are "groupless", i.e. an incoming
groupless packet should be sent only to the UMP EP port, and the event
with the groupless message is sent to UMP EP as is without the group
translation per port.
Also, the former reserved bit 0 for the client group filter is now
used for groupless events. When the bit 0 is set, the groupless
events are filtered out and skipped.
Link: https://lore.kernel.org/r/20230612081054.17200-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a new option to enable/disable the UMP Endpoint probing.
Some firmware seems screwed up when such a new command issued, and
this option allows user to suppress it.
Link: https://lore.kernel.org/r/20230612081054.17200-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Try to parse the UMP Endpoint and UMP Function Blocks for building the
topology at first. Only when those are missing (e.g. on an older USB
MIDI 2.0 spec or a unidirectional endpoint), the driver still creates
blocks based on USB group terminal block information as fallback.
Link: https://lore.kernel.org/r/20230612081054.17200-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds the basic support for UMP Endpoint and UMP Function
Block parsing, which are extended in the new UMP v1.1 spec.
The patch provides a new helper function to perform the query of the
UMP Endpoint information and builds up the UMP blocks based on UMP
Function Block information. For the communication over the UMP
Endpoint, it opens the rawmidi device once internally, inquiries the
UMP Endpoint and Function Block info by sending new UMP Stream
messages, and waits for the response for each query.
The new UMP spec allows to update the FB info and change its
associated groups or its activeness on the fly, too. For catching it,
the UMP core keeps watching the incoming UMP messages, and
snd_ump_receive() handles the incoming UMP Stream messages to refresh
the FB info.
Link: https://lore.kernel.org/r/20230612081054.17200-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a few more fields to snd_ump_endpoint_info and snd_ump_block_info
that are added in the new v1.1 spec. Those are filled by the UMP Stream
messages.
The rawmidi protocol version is bumped to 2.0.4 to indicate those
updates.
Also, update the proc outputs to show the newly introduced fields.
Link: https://lore.kernel.org/r/20230612081054.17200-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Ladisch
In MODULE_LICENSE macro, "GPL" string obsoletes "GPL v2" string by a
commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2"
bogosity").
This commit uses the preferable expression.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20230611144445.221529-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Sakamoto
In MODULE_LICENSE macro, "GPL" string obsoletes "GPL v2" string by a
commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2"
bogosity").
This commit uses the preferable expression.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20230611144445.221529-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HDA can only support single register read and write operations so does not
benefit from block writes. This means it gets no benefit from using the
rbtree register cache over the maple tree register cache so convert it to
use maple trees instead, it is more modern.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230609-alsa-hda-maple-v1-1-a2b725c8b8f5@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a quirk for Compaq N14JP6 to fixup ALC897 headset MIC no sound.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230609201058.523499-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This test covers the new Virtual PCM Test Driver, including the capturing,
playback and ioctl redefinition functionalities for both interleaved and
non-interleaved access modes. This test is also helpful as an usage example
of the 'pcmtest' driver.
We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Generate random or pattern-based capture data
- Check the playback stream for containing the looped pattern
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-3-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Generate random or pattern-based capture data
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-2-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
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Add documentation for the new Virtual PCM Test Driver. It covers all
possible usage cases: errors and delay injections, random and
pattern-based data generation, playback and ioctl redefinition
functionalities testing.
We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Check the playback stream for containing the looped pattern
- Generate random or pattern-based capture data
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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