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.
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.
YueHaibing [Sat, 12 Jun 2021 03:34:58 +0000 (11:34 +0800)]
ALSA: trident: Fix build error
sound/pci/trident/trident_memory.c: In function ‘set_tlb_bus’:
sound/pci/trident/trident_memory.c:85:35: error: ‘pagetr’ undeclared (first use in this function); did you mean ‘page’?
for (i = 0; i < UNIT_PAGES; i++, pagetr++) {
^~~~~~
page
Takashi Sakamoto [Fri, 11 Jun 2021 03:50:02 +0000 (12:50 +0900)]
ALSA: bebob: delete workaround for protocol version 3
In a commit c4d860a0d256 ("ALSA: bebob: loosen up severity of checking
continuity for BeBoB v3 quirk"), a workaround was added for the quirk in
BeBoB protocol version 3 against the discontinuity of data block counter.
As long as seeing with sequence replay for media clock recovery, such
quirk disappears.
Takashi Sakamoto [Fri, 11 Jun 2021 03:50:01 +0000 (12:50 +0900)]
ALSA: bebob: dismiss sleep after breaking connections
In a commit d3eabe939aee ("ALSA: bebob: expand sleep just after breaking
connections for protocol version 1"), a workaround was added for a quirk
of freeze in BeBoB protocol version 1. As long as seeing with sequence
replay for media clock recovery, the quirk disappears.
Takashi Iwai [Thu, 10 Jun 2021 11:09:35 +0000 (13:09 +0200)]
ALSA: core: Fix build error due to missing PAGE_SIZE
The recent refactoring of memalloc stuff removed the inclusion of
asm/page.h for simplicity, but it turned out this caused a compile
error due the lack of PAGE_SIZE definition on some architectures.
Do a partial revert for recovering from that.
Damien Zammit [Thu, 10 Jun 2021 08:35:28 +0000 (18:35 +1000)]
ALSA: usb-audio: Add support for Denon DN-X1600
This provides support for Denon DN-X1600 hardware mixer.
The device itself supports 44100, 48000 and 96000 (Hz)
sample rates, but switching rates via software is currently not working.
Therefore, this patch hardcodes the sample rate to 48000Hz which
enables all 8 channels to function correctly when the correct
sample rate is selected on the hardware itself.
Takashi Iwai [Wed, 9 Jun 2021 16:25:51 +0000 (18:25 +0200)]
ALSA: core: Add continuous and vmalloc mmap ops
The mmap of continuous pages and vmalloc'ed pages are relatively
easily done in a shot with the existing helper functions.
Implement the mmap ops for those types, so that the mmap works without
relying on the page fault handling.
Takashi Iwai [Wed, 9 Jun 2021 16:25:49 +0000 (18:25 +0200)]
ALSA: core: Abstract memory alloc helpers
This patch introduces the ops table to each memory allocation type
(SNDRV_DMA_TYPE_XXX) and abstract the handling for the better code
management. Then we get separate the page allocation, release and
other tasks for each type, especially for the SG buffer.
Each buffer type has now callbacks in the struct snd_malloc_ops, and
the common helper functions call those ops accordingly. The former
inline code that is specific to SG-buffer is moved into the local
sgbuf.c, and we can simplify the PCM code without details of memory
handling.
Takashi Sakamoto [Thu, 10 Jun 2021 03:17:32 +0000 (12:17 +0900)]
ALSA: firewire-lib: operate for period elapse event in process context
All of drivers in ALSA firewire stack processes two chances to process
isochronous packets in any isochronous context; in software IRQ context
for 1394 OHCI, and in process context of ALSA PCM application.
In the process context, callbacks of .pointer and .ack are utilized. The
callbacks are done by ALSA PCM core under acquiring lock of PCM substream,
In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for
drivers to awaken user processes from waiting for available frames. The
function voluntarily acquires lock of PCM substream, therefore it is not
called in the process context since it causes dead lock.
As a workaround to avoid the dead lock, all of drivers in ALSA firewire
stack uses workqueue to delegate the call. A variant of
snd_pcm_period_elapsed() without lock acquisition can obsolete the
workqueue.
An extra care is needed for the callback of .pointer since it's called
from snd_pcm_period_elapsed(). The isochronous context in Linux FireWire
subsystem is safe mostly for nested call except in software IRQ context.
Takashi Sakamoto [Thu, 10 Jun 2021 03:17:31 +0000 (12:17 +0900)]
ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
Current implementation of ALSA PCM core has a kernel API,
snd_pcm_period_elapsed(), for drivers to queue event to awaken processes
from waiting for available frames. The function voluntarily acquires lock
of PCM substream, therefore it is not called in process context for any
PCM operation since the lock is already acquired.
It is convenient for packet-oriented driver, at least for drivers to audio
and music unit in IEEE 1394 bus. The drivers are allowed by Linux
FireWire subsystem to process isochronous packets queued till recent
isochronous cycle in process context in any time.
This commit adds snd_pcm_period_elapsed() variant,
snd_pcm_period_elapsed_without_lock(), for drivers to queue the event in
the process context.
Takashi Iwai [Tue, 8 Jun 2021 14:05:38 +0000 (16:05 +0200)]
ALSA: poewrmac: Fix assignment in if condition
PPC powermac driver code contains a few assignments in if condition,
which is a bad coding style that may confuse readers and occasionally
lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:37 +0000 (16:05 +0200)]
ALSA: synth: Fix assignment in if condition
EMUx synth driver code contains lots of assignments in if condition,
which is a bad coding style that may confuse readers and occasionally
lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:34 +0000 (16:05 +0200)]
ALSA: vx: Fix assignment in if condition
VX driver helper code contains lots of assignments in if condition,
which is a bad coding style that may confuse readers and occasionally
lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:31 +0000 (16:05 +0200)]
ALSA: pcmcia: Fix assignment in if condition
PCMCIA VX222 and PDAudioCF drivers contain a few assignments in if
condition, which is a bad coding style that may confuse readers and
occasionally lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:30 +0000 (16:05 +0200)]
ALSA: seq: Fix assignment in if condition
There are lots of places doing assignments in if condition in ALSA
sequencer core, which is a bad coding style that may confuse readers
and occasionally lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:29 +0000 (16:05 +0200)]
ALSA: oss: Fix assignment in if condition
There are a few places doing assignments in if condition in ALSA PCM
and OSS emulation layers, which is a bad coding style that may confuse
readers and occasionally lead to bugs.
This patch is merely for coding-style fixes, no functional changes.
Takashi Iwai [Tue, 8 Jun 2021 14:05:28 +0000 (16:05 +0200)]
ALSA: pcm: Fix assignment in if condition
There are a few places doing assignments in if condition in ALSA PCM
core code, which is a bad coding style that may confuse readers and
occasionally lead to bugs.
This patch is merely for coding-style fixes, no functional changes.