Jeremy Szu [Fri, 25 Jun 2021 13:34:13 +0000 (21:34 +0800)]
ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 830 G8 Notebook PC
The HP EliteBook 830 G8 Notebook PC using ALC285 codec which using 0x04 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Takashi Iwai [Thu, 24 Jun 2021 08:02:45 +0000 (10:02 +0200)]
Merge tag 'asoc-fix-v5.13-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Fixes for v5.13
A final batch of fixes for v5.13, this is larger than I'd like due to
the fixes for a series of suspend issues that Intel turned up in their
testing this week.
Imre Deak [Wed, 23 Jun 2021 13:46:00 +0000 (16:46 +0300)]
ALSA: hda: Release controller display power during shutdown/reboot
Make sure the HDA driver's display power reference is released during
shutdown/reboot.
During the shutdown/reboot sequence the pci device core calls the
pm_runtime_resume handler for all devices before calling the driver's
shutdown callback and so the HDA driver's runtime resume callback will
acquire a display power reference (on HSW/BDW). This triggers a power
reference held WARN on HSW/BDW in the i915 driver's subsequent shutdown
handler, which expects all display power references to be released by
that time.
Since the HDA controller is stopped in the shutdown handler in any case,
let's follow here the same sequence as the one during runtime suspend.
This will also reset the HDA link and drop the display power reference,
getting rid of the above WARN.
Tested on HSW.
v2:
- Fix the build for CONFIG_PM=n (Takashi)
- s/__azx_runtime_suspend/azx_shutdown_chip/
Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/3618
References: https://lore.kernel.org/lkml/cea1f9a-52e0-b83-593d-52997fe1aaf6@er-systems.de Reported-and-tested-by: Thomas Voegtle <tv@lio96.de> Signed-off-by: Imre Deak <imre.deak@intel.com> Link: https://lore.kernel.org/r/20210623134601.2128663-1-imre.deak@intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Wed, 23 Jun 2021 07:59:38 +0000 (16:59 +0900)]
ALSA: firewire-motu: code refactoring for source detection of sampling clock in v3 protocol
Current implementation of driver has two similar helper functions for
source detection of sampling clock. This commit merges them as a code
refactoring.
Takashi Sakamoto [Wed, 23 Jun 2021 07:59:34 +0000 (16:59 +0900)]
ALSA: firewire-motu: code refactoring for detection of clock source in v2 protocol
Current implementation of driver has two similar helper functions for
source detection of sampling clock. This commit merges them as a code
refactoring.
Takashi Sakamoto [Wed, 23 Jun 2021 07:59:33 +0000 (16:59 +0900)]
ALSA: firewire-motu: fix detection for S/PDIF source on optical interface in v2 protocol
The devices in protocol version 2 has a register with flag for IEC 60958
signal detection as source of sampling clock without discrimination
between coaxial and optical interfaces. On the other hand, current
implementation of driver manage to interpret type of signal on optical
interface instead.
This commit fixes the detection of optical/coaxial interface for S/PDIF
signal.
ALSA: usb-audio: scarlett2: Update get_config to do endian conversion
For configuration items with a size of 16, scarlett2_usb_get_config()
was filling *buf with little-endian data. Update it to convert to CPU
endian. This function is not currently used so affects nothing yet;
will be used by the upcoming talkback feature.
ALSA: usb-audio: scarlett2: Add speaker switching support
The 18i8 and 18i20 Gen 3 support "speaker switching". Add a Speaker
Switch control which can be set to Off/Main/Alt.
When speaker switching is enabled or disabled, the interface may
change the state of the Analog Outputs 3 and 4 routing and the global
mute button, so use a flag private->speaker_switching_switched to note
that those should be checked when the next "monitor other"
notification is received.
ALSA: usb-audio: scarlett2: Update mux controls to allow updates
Enabling/disabling speaker switching will update the mux
configuration. To prepare for this, add a private->mux_updated flag
and update the scarlett2_mux_src_enum_ctl_get() callback to check it.
ALSA: usb-audio: scarlett2: Add sw_hw_ctls and mux_ctls
Save the struct snd_kcontrol pointers for the sw_hw and mux controls.
This is in preparation for speaker switching support which needs to be
able to update those controls.
ALSA: usb-audio: scarlett2: Split up sw_hw_enum_ctl_put()
Split part of scarlett2_sw_hw_enum_ctl_put() out into
scarlett2_sw_hw_change() so that the code which actually makes the
change is available in its own function. This will be used by the
speaker switching support which needs to set the SW/HW switch to HW
when speaker switching is enabled.
ALSA: usb-audio: scarlett2: Label 18i8 Gen 3 line outputs correctly
The 18i8 Gen 3 analogue 7/8 outputs are identified as line 3/4 on the
rear of the unit. Add support for remapping the channel numbers to
match the labelling.
ALSA: usb-audio: scarlett2: Add phantom power switch support
Some inputs on Gen 3 models support software-selectable phantom power.
Add support for getting and setting the state of those switches and
the "Phantom Power Persistence" switch.
ALSA: usb-audio: scarlett2: Add "air" switch support
Some inputs on Gen 3 models have an "air" feature which can be enabled
from the driver or (model-dependent) from the front panel. Add support
for getting and setting the state of those switches.
ALSA: usb-audio: scarlett2: Add support for Solo and 2i2 Gen 3
Add initial support for the Focusrite Scarlett Solo and 2i2 devices:
- They have no mixer
- They don't support reporting sync status or levels
- The configuration space is laid out differently to the other models
- There is no level (line/inst) switch on input 1 of the Solo
ALSA: usb-audio: scarlett2: Move get config above set config
Move scarlett2_usb_get() and scarlett2_usb_get_config() above the
functions relating to updating the configuration so that
scarlett2_usb_set_config() can call scarlett2_usb_get() in a
subsequent patch.
ALSA: usb-audio: scarlett2: Add Gen 3 MSD mode switch
Add a control to disable the Gen 3 MSD mode so that the full
functionality of the device is available. Don't create the other
controls until MSD mode is disabled.
ALSA: usb-audio: scarlett2: Add support for "input-other" notify
Some models allow the level and pad settings to be controlled from the
front-panel of the device. For these, the device will send an
"input-other" notification to prompt the driver to re-read the status
of those settings.
Takashi Iwai [Tue, 22 Jun 2021 17:00:49 +0000 (02:30 +0930)]
ALSA: usb-audio: scarlett2: Fix wrong resume call
The current way of the scarlett2 mixer code managing the
usb_mixer_elem_info object is wrong in two ways: it passes its
internal index to the head.id field, and the val_type field is
uninitialized. This ended up with the wrong execution at the resume
because a bogus unit id is passed wrongly. Also, in the later code
extensions, we'll have more mixer elements, and passing the index will
overflow the unit id size (of 256).
This patch corrects those issues. It introduces a new value type,
USB_MIXER_BESPOKEN, which indicates a non-standard mixer element, and
use this type for all scarlett2 mixer elements, as well as
initializing the fixed unit id 0 for avoiding the overflow.
ALSA: usb-audio: scarlett2: Fix Level Meter control
The Level Meter control had a fixed number of channels and therefore
only worked with the 18i20 Gen 2. Fix the control to contain the
correct number of channels.
The scarlett2_ports struct contains both generic (hardware IDs and
descriptions) and model-specific (port count) data. Remove the generic
data from the scarlett2_device_info struct so it is not repeated for
every model.
ALSA: usb-audio: scarlett2: Allow arbitrary ordering of mux entries
Some Gen 3 devices do not put all of the mux entries for the same port
types together in order in the "set mux" message data. To prepare for
this, replace the struct scarlett2_ports num[] array and the
assignment_order[] array with mux_assignment[], a list of port types
and ranges that is defined in the struct scarlett2_device_info.
For each analogue output, in addition to the output volume (gain)
control, the hardware also has a mute control. Add ALSA mute controls
for each analogue output.
If the device has the line_out_hw_vol feature, then the mute control
is disabled along with the output volume control when the switch is
set to HW.
ALSA: usb-audio: scarlett2: Move info lookup out of init function
The info variable is not used by snd_scarlett_gen2_init() except to
pass it to snd_scarlett_gen2_controls_create(), so move the lookup
into that function.
ALSA: usb-audio: scarlett2: Improve device info lookup
Add the USB device ID to the scarlett2_device_info struct so that the
switch statement which finds the appropriate struct can be replaced
with a loop that looks through an array of pointers to those structs.
Jiajun Cao [Tue, 22 Jun 2021 13:19:42 +0000 (21:19 +0800)]
ALSA: hda: Add IRQ check for platform_get_irq()
The function hda_tegra_first_init() neglects to check the return
value after executing platform_get_irq().
hda_tegra_first_init() should check the return value (if negative
error number) for errors so as to not pass a negative value to
the devm_request_irq().
Fix it by adding a check for the return value irq_id.
Takashi Iwai [Tue, 22 Jun 2021 09:06:47 +0000 (11:06 +0200)]
ALSA: usb-audio: Fix OOB access at proc output
At extending the available mixer values for 32bit types, we forgot to
add the corresponding entries for the format dump in the proc output.
This may result in OOB access. Here adds the missing entries.
ASoC: atmel-i2s: Fix usage of capture and playback at the same time
For both capture and playback streams to work at the same time, only the
needed values from a register need to be updated. Also, clocks should be
enabled only when the first stream is started and stopped when there is no
running stream.
Timur Tabi [Sun, 20 Jun 2021 16:01:35 +0000 (11:01 -0500)]
MAINTAINERS: remove Timur Tabi from Freescale SOC sound drivers
I haven't touched these drivers in seven years, and none of the
patches sent to me these days affect code that I wrote. The
other maintainers are doing a very good job without me.
ASoC: rt711-sdca-sdw: fix race condition on system suspend
In the initial driver we cancelled deferred work, but there is still a
window of time where a new interrupt could result in new deferred work
executed after the link is disabled, leading to an IO error. While we
did not see this IO error on RT711-sdca-based platforms, the code pattern
is similar to the RT700 case where the IO error was noted, so the fix
is added for consistency.
This patch uses an 'disable_irq_lock' mutex to prevent new interrupts
from happening after the start of the system suspend. The choice of a
mutex v. a spinlock is mainly due to the time required to clear
interrupts, which requires a command to be transmitted by the
SoundWire host IP and acknowledged with an interrupt. The
'interrupt_callback' routine is also not meant to be called from an
interrupt context.
An additional 'disable_irq' flag prevents race conditions where the
status changes before the interrupts are disabled, but the workqueue
handling status changes is scheduled after the completion of the
system suspend. On resume the interrupts are re-enabled already by the
io_init routine so we only clear the flag.
The code is slightly different from the other codecs since the
interrupt callback deals with the SDCA interrupts, leading to a much
larger section that's protected by the mutex. The SoundWire interrupt
scheme requires a read after clearing a status, it's not clear from
the specifications what would happen if SDCA interrupts are disabled
in the middle of the sequence, so the entire interrupt status
read/write is kept as is, even if in the end we discard the
information.
ASoC: rt5682-sdw: fix race condition on system suspend
In the initial driver we cancelled deferred work, but there is still a
window of time where a new interrupt could result in new deferred work
executed after the link is disabled, leading to an IO error. While we
did not see this IO error on RT5682-based platforms, the code pattern
is similar to the RT700 case where the IO error was noted, so the fix
is added for consistency.
This patch uses an 'disable_irq_lock' mutex to prevent new interrupts
from happening after the start of the system suspend. The choice of a
mutex v. a spinlock is mainly due to the time required to clear
interrupts, which requires a command to be transmitted by the
SoundWire host IP and acknowledged with an interrupt. The
'interrupt_callback' routine is also not meant to be called from an
interrupt context.
An additional 'disable_irq' flag prevents race conditions where the
status changes before the interrupts are disabled, but the workqueue
handling status changes is scheduled after the completion of the
system suspend. On resume the interrupts are re-enabled already by the
io_init routine so we only clear the flag.
The Fixes tag points to a 5.10 commit, there's no need to propagate
this change to earlier upstream versions.
ASoC: rt711-sdw: fix race condition on system suspend
In previous commits we cancelled deferred work, but there is still a
window of time where a new interrupt could result in new deferred work
executed after the link is disabled, leading to an IO error. While we
did not see this IO error on RT711-based platforms, the code pattern
is similar to the RT700 case where the IO error was noted, so the fix
is added for consistency.
This patch uses an 'disable_irq_lock' mutex to prevent new interrupts
from happening after the start of the system suspend. The choice of a
mutex v. a spinlock is mainly due to the time required to clear
interrupts, which requires a command to be transmitted by the
SoundWire host IP and acknowledged with an interrupt. The
'interrupt_callback' routine is also not meant to be called from an
interrupt context.
An additional 'disable_irq' flag prevents race conditions where the
status changes before the interrupts are disabled, but the workqueue
handling status changes is scheduled after the completion of the
system suspend. On resume the interrupts are re-enabled already by the
io_init routine so we only clear the flag.
ASoC: rt700-sdw: fix race condition on system suspend
In previous commits we cancelled deferred work, but there is still a
window of time where a new interrupt could result in new deferred work
executed after the link is disabled, leading to an IO error.
This patch uses an 'disable_irq_lock' mutex to prevent new interrupts
from happening after the start of the system suspend. The choice of a
mutex v. a spinlock is mainly due to the time required to clear
interrupts, which requires a command to be transmitted by the
SoundWire host IP and acknowledged with an interrupt. The
'interrupt_callback' routine is also not meant to be called from an
interrupt context.
An additional 'disable_irq' flag prevents race conditions where the
status changes before the interrupts are disabled, but the workqueue
handling status changes is scheduled after the completion of the
system suspend. On resume the interrupts are re-enabled already by the
io_init routine so we only clear the flag.
soundwire: export sdw_update() and sdw_update_no_pm()
We currently export sdw_read() and sdw_write() but the sdw_update()
and sdw_update_no_pm() are currently available only to the bus
code. This was missed in an earlier contribution.
Export both functions so that codec drivers can perform
read-modify-write operations without duplicating the code.
ALSA: usb-audio: scarlett2: Fix union usage in mixer control callbacks
Fix mixer control callbacks to use the correct members of the struct
snd_ctl_elem_value. The use of value.integer and value.enumerated were
swapped in a few places.
Update scarlett2_mux_src_enum_ctl_put() to use min() instead of
clamp() as value.enumerated.item is unsigned.
The private->vol_updated flag was being checked outside of the
mutex_lock/unlock() of private->data_mutex leading to the volume data
being fetched twice from the device unnecessarily or old volume data
being returned.
Update scarlett2_*_ctl_get() and include the private->vol_updated flag
check inside the critical region.
Rename struct scarlett2_mixer_data to struct scarlett2_data. A
less-wordy name is better because it is used everywhere, and although
this is a mixer driver, it also controls other vendor-specific
features.
To match the vendor's terminology, change #defines, identifiers, and
comments:
- mute/dim/hardware buttons are now called dim/mute
- mixer status/interrupt is now notify
- vol is now monitor
The per-model button_count value was used to determine whether
dim/mute controls should be added, but these are present iff
line_out_hw_vol is true. Remove button_count and replace with
SCARLETT2_BUTTON_MAX and a check for line_out_hw_vol true.
Just ignore instead of printing an error if the interrupt data is not
the expected length. This check was for development and the condition
has not been observed.
Improve alignment and readability with:
- Whitespace fixes
- Add leading zeros to 32-bit flag values
- Rename SCARLETT2_USB_GET_METER_LEVELS to SCARLETT2_USB_GET_METER
- Rename SCARLETT2_PORT_DIRECTIONS to SCARLETT2_PORT_DIRNS
Takashi Iwai [Sun, 20 Jun 2021 06:59:52 +0000 (08:59 +0200)]
ALSA: hda/realtek: Fix bass speaker DAC mapping for Asus UM431D
Asus Zenbook 14 UM431D has two speaker pins and a headphone pin, and
the auto-parser ends up assigning the bass to the third DAC 0x06.
Although the tone comes out, it's inconvenient because this DAC has no
volume control unlike two other DACs.
For obtaining the volume control for the bass speaker, this patch
enforces the mapping to let both front and bass speaker pins sharing
the same DAC. It's not ideal but a little bit of improvement.
Since we've already applied the same workaround for another ASUS
machine, we just need to hook the chain to the existing quirk.
Elia Devito [Sat, 19 Jun 2021 20:41:04 +0000 (22:41 +0200)]
ALSA: hda/realtek: Improve fixup for HP Spectre x360 15-df0xxx
On HP Spectre x360 15-df0xxx, after system boot with plugged headset, the
headset mic are not detected.
Moving pincfg and DAC's config to single fixup function fix this.
[ The actual bug in the original code was that it used a chain to
ALC286_FIXUP_SPEAKER2_TO_DAC1, and it contains not only the DAC1
route fix but also another chain to ALC269_FIXUP_THINKPAD_ACPI.
I thought the latter one is harmless for non-Thinkpad, but it
doesn't seem so; it contains again yet another chain to
ALC269_FIXUP_SKI_IGNORE, and this might be bad for some machines,
including this HP machine. -- tiwai ]
Takashi Sakamoto [Sat, 19 Jun 2021 08:39:22 +0000 (17:39 +0900)]
ALSA: bebob: add support for ToneWeal FW66
A user of FFADO project reported the issue of ToneWeal FW66. As a result,
the device is identified as one of applications of BeBoB solution.
I note that in the report the device returns contradictory result in plug
discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't
perform it thus it's likely to handle the device without issues.
I receive no reaction to test request for this patch yet, however it would
be worth to add support for it.
Takashi Sakamoto [Fri, 18 Jun 2021 04:07:13 +0000 (13:07 +0900)]
ALSA: firewire-motu: fix rx packet format at higher rate for MOTU 828 mk3 Hybrid
I assumed that the combination of packet formats for MOTU 828 mk3 Hybrid
is the same as MOTU 828 mk3 FireWire. However at higher sampling rate, it
is different. MOTU 828 mk3 Hybrid has additional 4 dummy data chunks for
rx packet.
This commit fixes the issue to which I address at a commit f2ac3b839540
("ALSA: firewire-motu: sequence replay for source packet header").
Jeremy Szu [Thu, 17 Jun 2021 17:14:20 +0000 (01:14 +0800)]
ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook x360 830 G8
The HP EliteBook x360 830 G8 using ALC285 codec which using 0x04 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Takashi Iwai [Thu, 17 Jun 2021 13:47:42 +0000 (15:47 +0200)]
ALSA: seq: oss: Fix error check at system port creation
The system port creation in ALSA OSS sequencer was wrongly checked
against to the port number that can be never negative. The error code
should be checked rather against the ioctl call.
Takashi Sakamoto [Wed, 16 Jun 2021 08:28:47 +0000 (17:28 +0900)]
ALSA: firewire-motu: add support for MOTU 896
MOTU 896 is a second model in MOTU FireWire series, produced in 2001. This
model consists of three chips:
* Texas Instruments TSB41AB2 (Physical layer for IEEE 1394 bus)
* Philips Semiconductors PDI 1394L21BE (Link layer for IEEE 1394 bus and
packet processing layer)
* QuickLogic QuickRAM QL4016 (Data block processing layer and digital
signal processing)
This commit adds a support for the model, with its unique protocol as
version 1. The features of this protocol are:
* no MIDI support.
* Rx packets have no data chunks for control and status messages.
* Tx packets have 2 bytes for control and status messages in the end of
each data block.
* The most of settings are represented in bit flag in one quadlet address
(0x'ffff'f000'0b14).
* It's selectable to use signal on optical interface, however the device
has no register specific to it. The state has effect just to whether
to exclude differed data chunks.
* The internal multiplexer is not configured by software.
Just after powering on, the device has a quirk to fail handling
transaction. I recommend users to connect the device enough after powering
on.
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 0003ab34 directory_length 3, crc 43828
42c 120001f2 specifier id
430 13000002 version
434 17102801 model
Takashi Sakamoto [Wed, 16 Jun 2021 08:28:46 +0000 (17:28 +0900)]
ALSA: firewire-motu: add support for MOTU 828
MOTU 828 is a first model in MOTU FireWire series, produced in 2001. This
model consists of three chips:
* Texas Instruments TSB41AB1 (Physical layer for IEEE 1394 bus)
* Philips Semiconductors 1394L21BE (Link layer for IEEE 1394 bus and
packet processing layer)
* QuickLogic QuickRAM QL4016 (Data block processing layer and digital
signal processing)
This commit adds a support for this model, with its unique protocol as
version 1. The features of this protocol are:
* no MIDI support.
* Rx packets have no data chunks for control and status messages.
* Tx packets have 2 data chunks for control and status messages in the
end of each data block. The chunks consist of data block counter
(4 byte) and message (2 byte).
* All of settings are represented in bit flag in one quadlet address
(0x'ffff'f000'0b00).
* When optical interface is configured as S/PDIF, signals of the interface
is multiplexed for packets, instead of signals of coaxial interface.
* The internal multiplexer is not configured by software.
I note that the device has a quirk to mute output voluntarily during
receiving batch of packets in the beginning of packet streaming. The
operation to unmute should be done by software enough after the device
shifts the state, however it's not deterministic. Furthermore, just
after switching rate of sampling clock, the device keeps the state longer.
This patch manages to sleep 100 msec before unmute operation, but it may
fail to release the mute in the case that the rate is changed. As a
workaround, users can restart packet streaming at the same rate, or write
to specific register from userspace.
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 00035052 directory_length 3, crc 20562
42c 120001f2 specifier id
430 13000001 version
434 17101800 model
Colin Ian King [Tue, 15 Jun 2021 14:20:48 +0000 (15:20 +0100)]
ALSA: bebob: Fix bit flag quirk constants
The quirking bit-flags are currently set as contiguous integer enum values
and so currently SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC is zero and so
he quirking never getting set or tested correctly for this quirk. Fix this
by setting the quirking constants as shifted bit values.
Addresses-Coverity: ("Bitwise-and with zero") Fixes: 93cd12d6e88a ("ALSA: bebob: code refactoring for model-dependent quirks") Signed-off-by: Colin Ian King <colin.king@canonical.com> Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20210615142048.59900-1-colin.king@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jaroslav Kysela [Mon, 14 Jun 2021 07:17:10 +0000 (09:17 +0200)]
ALSA: control_led - fix initialization in the mode show callback
The str variable should be always initialized before use even if
the switch covers all cases. This is a minimalistic fix: Assign NULL,
the sprintf() may print '(null)' if something is corrupted.
Takashi Sakamoto [Fri, 11 Jun 2021 09:37:30 +0000 (18:37 +0900)]
ALSA: bebob: correct device entries for Phonic Helix Board and FireFly series
Phonic shipped Helix board and FireFly series with IEEE 1394
functionality. Regarding to the parameters in unit directory, these
series have two cases below:
1. the same parameters in unit directory
* Firefly 202
* Firefly 302
* Firefly 808 Universal
* HelixBoard 12 FireWire, 12 Universal
* HelixBoard 18 FireWire, 18 Universal
* HelixBoard 24 FireWire, 24 Universal
Takashi Sakamoto [Fri, 11 Jun 2021 09:37:29 +0000 (18:37 +0900)]
ALSA: bebob: code refactoring for M-Audio models
For M-Audio FireWire 410, the value of immediate entry for vendor in unit
directory is the value for BridgeCo. AG OUI. It seems that M-Audio uses
initial settings as is for its product.
For Mackie D.2 FireWire option card, 0x00000f is used for the value of
immediate entry for vendor in unit directory. The value comes from report
by FFADO user in below page:
However, it seems to be wrong. There are two causes; vendor's mistake to
decide value for GUID field in configuration ROM against standard, as
Stefan Richter mentioned in below post:
Another is implementation of libffado. The library doesn't print out the
value from immediate entry for vendor in unit directory. It just print out
the first 6 bytes of GUID as vendor ID.
Takashi Sakamoto [Fri, 11 Jun 2021 09:37:25 +0000 (18:37 +0900)]
ALSA: bebob: fulfil device entries
Although unit directory in root directory of configuration ROM has the
same value (0x00a02d) for its specifier_id entry to express AV/C device,
it has two cases for the value (0x010001/0x014001) to version entry.
projects / thirdparty / linux.git / log
