Matthias Reichl [Sat, 24 Jun 2023 16:52:32 +0000 (18:52 +0200)]
ASoC: hdmi-codec: fix channel info for compressed formats
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.
Matthias Reichl [Sat, 24 Jun 2023 16:52:16 +0000 (18:52 +0200)]
ALSA: pcm: fix ELD constraints for (E)AC3, DTS(-HD) and MLP formats
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.
Takashi Iwai [Fri, 23 Jun 2023 07:55:30 +0000 (09:55 +0200)]
ALSA: ump: Export MIDI1 / UMP conversion helpers
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().
Takashi Iwai [Wed, 21 Jun 2023 11:06:39 +0000 (13:06 +0200)]
Merge branch 'topic/midi20' into for-next
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.
Takashi Iwai [Wed, 21 Jun 2023 11:02:41 +0000 (13:02 +0200)]
ALSA: ump: Export snd_ump_receive_ump_val()
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.
Takashi Iwai [Wed, 21 Jun 2023 11:02:40 +0000 (13:02 +0200)]
ALSA: ump: Add no_process_stream flag
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.
Ivan Orlov [Tue, 20 Jun 2023 17:56:34 +0000 (19:56 +0200)]
sound: make all 'class' structures const
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.
Srinivas Kandagatla [Mon, 19 Jun 2023 09:28:05 +0000 (10:28 +0100)]
ALSA: compress: allow setting codec params after next track
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.
Charles Keepax [Wed, 14 Jun 2023 14:21:16 +0000 (15:21 +0100)]
ASoC: intel: sof_sdw: Fixup typo in device link checking
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.
Takashi Iwai [Fri, 16 Jun 2023 07:28:27 +0000 (09:28 +0200)]
Merge tag 'asoc-fix-v6.4-rc6-2' of 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.
Lukasz Tyl [Wed, 14 Jun 2023 12:25:24 +0000 (14:25 +0200)]
ALSA: usb-audio: Add quirk flag for HEM devices to enable native DSD playback
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.
Takashi Iwai [Mon, 12 Jun 2023 13:28:18 +0000 (15:28 +0200)]
ALSA: usb-audio: Fix broken resume due to UAC3 power state
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.
Takashi Iwai [Mon, 12 Jun 2023 12:55:33 +0000 (14:55 +0200)]
ALSA: seq: oss: Fix racy open/close of MIDI devices
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.
Jon Hunter [Tue, 13 Jun 2023 09:34:53 +0000 (10:34 +0100)]
ASoC: tegra: Fix Master Volume Control
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.
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:25 +0000 (21:13 +0200)]
ALSA: emu10k1: actually show some S/PDIF status in /proc for E-MU cards
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).
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:22 +0000 (21:13 +0200)]
ALSA: timer: minimize open-coded access to hw.resolution
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.
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:21 +0000 (21:13 +0200)]
ALSA: emu10k1: fix synthesizer pitch for E-MU cards at 44.1 kHz
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.
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:20 +0000 (21:13 +0200)]
ALSA: emu10k1: fix sample rates for E-MU cards at 44.1 kHz word clock
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
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:19 +0000 (21:13 +0200)]
ALSA: emu10k1: query rate of external clock sources on E-MU cards
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.
Oswald Buddenhagen [Mon, 12 Jun 2023 19:13:17 +0000 (21:13 +0200)]
ALSA: emu10k1: split off E-MU fallback clock from clock source
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.
Takashi Iwai [Tue, 13 Jun 2023 05:36:39 +0000 (07:36 +0200)]
Merge branch 'topic/midi20' into for-next
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:53 +0000 (10:10 +0200)]
ALSA: ump: Add info flag bit for static blocks
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:52 +0000 (10:10 +0200)]
ALSA: seq: ump: Notify UMP protocol change to sequencer
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:51 +0000 (10:10 +0200)]
ALSA: seq: ump: Notify port changes to system port
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:50 +0000 (10:10 +0200)]
ALSA: seq: ump: Handle FB info update
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:49 +0000 (10:10 +0200)]
ALSA: seq: ump: Handle groupless messages
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:48 +0000 (10:10 +0200)]
ALSA: usb-audio: Add midi2_ump_probe option
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:47 +0000 (10:10 +0200)]
ALSA: usb-audio: Parse UMP Endpoint and Function Blocks at first
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:46 +0000 (10:10 +0200)]
ALSA: ump: Support UMP Endpoint and Function Block parsing
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.
Takashi Iwai [Mon, 12 Jun 2023 08:10:45 +0000 (10:10 +0200)]
ALSA: ump: Add more attributes to UMP EP and FB info
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.
Takashi Sakamoto [Sun, 11 Jun 2023 14:44:45 +0000 (23:44 +0900)]
ALSA: firewire: use 'GPL' string for module license contributed by Clemens 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").
Takashi Sakamoto [Sun, 11 Jun 2023 14:44:44 +0000 (23:44 +0900)]
ALSA: firewire: use 'GPL' string for module license contributed by Takashi 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").
Mark Brown [Sat, 10 Jun 2023 14:26:37 +0000 (15:26 +0100)]
ALSA: hda: Use maple tree register cache
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.
Ivan Orlov [Tue, 6 Jun 2023 19:32:54 +0000 (23:32 +0400)]
selftests: ALSA: Add test for the 'pcmtest' driver
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.
Ivan Orlov [Tue, 6 Jun 2023 19:32:53 +0000 (23:32 +0400)]
ALSA: Implement the new Virtual PCM Test 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
- 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.
Ivan Orlov [Tue, 6 Jun 2023 19:32:52 +0000 (23:32 +0400)]
docs: sound: add 'pcmtest' driver documentation
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.
Yanteng Si [Wed, 7 Jun 2023 09:21:51 +0000 (17:21 +0800)]
ALSA: hda: Workaround for SDnCTL register on loongson
On loongson controller, after calling snd_hdac_stream_updateb()
to enable DMA engine, the SDnCTL.STRM will become to zero. We
need to access SDnCTL in dword to keep SDnCTL.STRM is not changed.
Yanteng Si [Wed, 7 Jun 2023 09:21:50 +0000 (17:21 +0800)]
ALSA: hda: Using polling mode for loongson controller by default
On loongson controller, RIRBSTS.RINTFL cannot be cleared,
azx_interrupt() is called all the time. We disable RIRB
interrupt, and use polling mode by default.
Chris Chiu [Tue, 6 Jun 2023 14:57:47 +0000 (22:57 +0800)]
ALSA: hda/realtek: Enable 4 amplifiers instead of 2 on a HP platform
In the commit 7bb62340951a ("ALSA: hda/realtek: fix speaker, mute/micmute
LEDs not work on a HP platform"), speakers and LEDs are fixed but only 2
CS35L41 amplifiers on SPI bus connected to Realtek codec are enabled. Need
the ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED to get all amplifiers working.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com> Fixes: 7bb62340951a ("ALSA: hda/realtek: fix speaker, mute/micmute LEDs not work on a HP platform") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230606145747.135966-1-chris.chiu@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 13:09:42 +0000 (15:09 +0200)]
Merge tag 'asoc-fix-v6.4-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
A lot of routine driver specific fixes here, nothing in the core though
there are a couple of fixes for the generic cards. There's also a few
new quirks for x86 platforms.
Takashi Iwai [Tue, 6 Jun 2023 09:40:35 +0000 (11:40 +0200)]
ALSA: control: Keep the previous numid at snd_ctl_rename_id()
We don't need to change the numid at each time snd_ctl_rename_id() is
called, as the control element size itself doesn't change. Let's keep
the previous numid value.
Along with it, add a note about calling this function only in the
card init phase.
Takashi Iwai [Tue, 6 Jun 2023 09:38:55 +0000 (11:38 +0200)]
ALSA: hda: Fix kctl->id initialization
HD-audio core code replaces the kctl->id.index of SPDIF-related
controls after assigning via snd_ctl_add(). This doesn't work any
longer with the new Xarray lookup change. The change of the kctl->id
content has to be done via snd_ctl_rename_id() helper, instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:54 +0000 (11:38 +0200)]
ALSA: gus: Fix kctl->id initialization
GUS driver replaces the kctl->id.index after assigning the kctl via
snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:53 +0000 (11:38 +0200)]
ALSA: cmipci: Fix kctl->id initialization
cmipci driver replaces the kctl->id.device after assigning the kctl
via snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:52 +0000 (11:38 +0200)]
ALSA: ymfpci: Fix kctl->id initialization
ymfpci driver replaces the kctl->id.device after assigning the kctl
via snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 5 Jun 2023 14:47:58 +0000 (16:47 +0200)]
ALSA: seq: Avoid confusion of aligned read size
Currently the read event packet size in snd_seq_read() is defined by
client->midi_version value that is guaranteed to be zero if UMP isn't
enabled. But the static analyzer doesn't know of the fact, and it
still suspects as if it were leading to a potential overflow.
Add the more explicit check of CONFIG_SND_SEQ_UMP to determine the
aligned_size value for avoiding the confusion.
Richard Fitzgerald [Mon, 5 Jun 2023 14:32:38 +0000 (15:32 +0100)]
firmware: cs_dsp: Log correct region name in bin error messages
In cs_dsp_load_coeff() region_name should be set in the XM/YM/ZM
cases otherwise any errors will log the region as "Unknown".
While doing this also change one error message that logged the
region type ID to log the region_name instead. This makes it
consistent with other messages in the same function.
Oswald Buddenhagen [Fri, 26 May 2023 10:16:59 +0000 (12:16 +0200)]
ALSA: emu10k1: vastly improve usefulness of info in /proc
- Include the FX bus map, without which the already present send routing
info would require looking up the documentation.
- Include the physical I/O channels as known to the driver
- Make the multi-channel capture map actually name the mapped input
channels rather than "FXBUS" (Audigy) or even "???" (SbLive)
- The latter two are omitted for E-MU cards, as their physical I/O is
routed through the FPGA
- While at it, make the "Card" field somewhat more useful
This includes de-duplicating the label tables between emuproc and emufx,
updating/improving the FX bus label table, and making the SB Live! 5.1
multi-track capture channel mapping hack data-driven.
Oswald Buddenhagen [Fri, 26 May 2023 10:16:55 +0000 (12:16 +0200)]
ALSA: emu10k1: fix writing 1st pointer-offset register set through /proc
The limits were appropriate only for the 2nd set.
FWIW, the channel count 4 for the 2nd set is suspicious as well - at
least P17V_PLAYBACK_FIFO_PTR actually has 8 channels, and comments on
HCFG2 hint at that as well. But all bitmasks are documented only for 4
channels. Anyway, rectifying that is out of scope for this patch.
Oswald Buddenhagen [Fri, 26 May 2023 10:16:54 +0000 (12:16 +0200)]
ALSA: emu10k1: hide absent 2nd pointer-offset register set from /proc
The 2nd register set belongs to the P16V chip (or embedded P17V module),
so there is nothing to show when no such part is present. Gen2 E-MU
cards have a P17V, but it's entirely unused, so we hide it there as
well.
Uwe Kleine-König [Thu, 25 May 2023 20:36:40 +0000 (22:36 +0200)]
ALSA: Switch i2c drivers back to use .probe()
After commit b8a1a4cd5a98 ("i2c: Provide a temporary .probe_new()
call-back type"), all drivers being converted to .probe_new() and then 03c835f498b5 ("i2c: Switch .probe() to not take an id parameter") convert
back to (the new) .probe() to be able to eventually drop .probe_new() from
struct i2c_driver.
selftests: alsa: pcm-test: Fix compiler warnings about the format
GCC 11.3.0 issues warnings in this module about wrong sizes of format
specifiers:
pcm-test.c: In function ‘test_pcm_time’:
pcm-test.c:384:68: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 5 \
has type ‘unsigned int’ [-Wformat=]
384 | snprintf(msg, sizeof(msg), "rate mismatch %ld != %ld", rate, rrate);
pcm-test.c:455:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
455 | "expected %d, wrote %li", rate, frames);
pcm-test.c:462:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
462 | "expected %d, wrote %li", rate, frames);
pcm-test.c:467:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
467 | "expected %d, wrote %li", rate, frames);
Simple fix according to compiler's suggestion removed the warnings.
Chancel Liu [Tue, 30 May 2023 10:30:12 +0000 (18:30 +0800)]
ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted
There's an issue on SAI synchronous mode that TX/RX side can't get BCLK
from RX/TX it sync with if BYP bit is asserted. It's a workaround to
fix it that enable SION of IOMUX pad control and assert BCI.
For example if TX sync with RX which means both TX and RX are using clk
form RX and BYP=1. TX can get BCLK only if the following two conditions
are valid:
1. SION of RX BCLK IOMUX pad is set to 1
2. BCI of TX is set to 1
The code in asoc_simple_startup was treating any non-zero return from
snd_pcm_hw_constraint_minmax as an error, when this can return 1 in some
normal cases and only negative values indicate an error.
When this happened, it caused asoc_simple_startup to disable the clocks
it just enabled and return 1, which was not treated as an error by the
calling code which only checks for negative return values. Then when the
PCM is eventually shut down, it causes the clock framework to complain
about disabling clocks that were not enabled.
Fix the check for snd_pcm_hw_constraint_minmax return value to only
treat negative values as an error.
Mark Brown [Thu, 1 Jun 2023 15:43:38 +0000 (16:43 +0100)]
ASoC: mediatek: fix use-after-free in driver remove
Merge series from Trevor Wu <trevor.wu@mediatek.com>:
These patches concern modifications made in mt8186[1]. The clock
unregistration mechanism used in mt8188 and mt8195 is similar with
mt8186, resulting in the same problem existing within the driver.
Therefore, the solution has also been applied to these two platforms.
Trevor Wu [Thu, 1 Jun 2023 03:33:18 +0000 (11:33 +0800)]
ASoC: mediatek: mt8195: fix use-after-free in driver remove path
During mt8195_afe_init_clock(), mt8195_audsys_clk_register() was called
followed by several other devm functions. At mt8195_afe_deinit_clock()
located at mt8195_afe_pcm_dev_remove(), mt8195_audsys_clk_unregister()
was called.
However, there was an issue with the order in which these functions were
called. Specifically, the remove callback of platform_driver was called
before devres released the resource, resulting in a use-after-free issue
during remove time.
At probe time, the order of calls was:
1. mt8195_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8195_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, we can utilize devm_add_action_or_reset() in
mt8195_audsys_clk_register() so that the remove order can be changed to
3->2->1.
Fixes: 6746cc858259 ("ASoC: mediatek: mt8195: add platform driver") Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Reviewed-by: Douglas Anderson <dianders@chromium.org> Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> Link: https://lore.kernel.org/r/20230601033318.10408-3-trevor.wu@mediatek.com Signed-off-by: Mark Brown <broonie@kernel.org>
Trevor Wu [Thu, 1 Jun 2023 03:33:17 +0000 (11:33 +0800)]
ASoC: mediatek: mt8188: fix use-after-free in driver remove path
During mt8188_afe_init_clock(), mt8188_audsys_clk_register() was called
followed by several other devm functions. The caller of
mt8188_afe_init_clock() utilized devm_add_action_or_reset() to call
mt8188_afe_deinit_clock(). However, the order was incorrect, causing a
use-after-free issue during remove time.
At probe time, the order of calls was:
1. mt8188_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8188_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, it's necessary to move devm_add_action_or_reset()
to the appropriate position so that the remove order can be 3->2->1.
Fixes: f6b026479b13 ("ASoC: mediatek: mt8188: support audio clock control") Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Reviewed-by: Douglas Anderson <dianders@chromium.org> Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> Link: https://lore.kernel.org/r/20230601033318.10408-2-trevor.wu@mediatek.com Signed-off-by: Mark Brown <broonie@kernel.org>
Vijendar Mukunda [Thu, 25 May 2023 11:29:55 +0000 (16:59 +0530)]
ASoC: amd: ps: fix for acp_lock access in pdm driver
Sending the mutex address(acp_lock) as platform
data during ACP PDM platform driver register sequence,
its creating copy of the platform data.
Referencing this platform data in ACP PDM driver results
incorrect reference to the common lock usage.
Instead of directly passing the lock address as platform
data, retrieve it from parent driver data structure
and use the same lock reference in ACP PDM driver.
Lenovo M70/M90 Gen4 are equipped with ALC897, and they need
ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work.
The previous quirk for M70/M90 is for Gen3.
Srinivas Kandagatla [Tue, 23 May 2023 16:54:14 +0000 (17:54 +0100)]
ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
Srinivas Kandagatla [Tue, 23 May 2023 15:46:05 +0000 (16:46 +0100)]
ASoC: codecs: wsa881x: do not set can_multi_write flag
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
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