Takashi Iwai [Tue, 26 May 2020 08:28:10 +0000 (10:28 +0200)]
ALSA: usb-audio: Quirks for Gigabyte TRX40 Aorus Master onboard audio
Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Chris Chiu [Tue, 26 May 2020 06:26:13 +0000 (14:26 +0800)]
ALSA: usb-audio: mixer: volume quirk for ESS Technology Asus USB DAC
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Takashi Iwai [Tue, 26 May 2020 06:24:06 +0000 (08:24 +0200)]
ALSA: hda/realtek - Add a model for Thinkpad T570 without DAC workaround
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 54947cd64c1b ("ALSA: hda/realtek - Fix
speaker output regression on Thinkpad T570"). Essentially it fixes
the DAC / pin pairing by a static table. It was confirmed and merged
to stable kernel later.
Now, interestingly, we got another regression report for the very same
model (T570) about the similar problem, and the commit above was the
culprit. That is, by some reason, there are devices that prefer the
DAC1, and another device DAC2!
Unfortunately those have the same ID and we have no idea what can
differentiate, in this patch, a new fixup model "tpt470-dock-fix" is
provided, so that users with such a machine can apply it manually.
When model=tpt470-dock-fix option is passed to snd-hda-intel module,
it avoids the fixed DAC pairing and the DAC1 is assigned to the
speaker like the earlier versions.
Takashi Sakamoto [Sat, 23 May 2020 07:17:33 +0000 (16:17 +0900)]
ALSA: firewire-motu: add support for MOTU UltraLite-mk3 (FireWire only model)
UltraLite-mk3 was shipped 2008 by MOTU. This model has two lineups;
FireWire-only and FireWire/USB2.0 Hybrid model. Additionally, the latter
has two variants in respect of the type of IEEE 1394 connector; alpha
and beta connector.
This commit adds support for the FireWire-only model, which is already
discontinued.
root directory
-----------------------------------------------------------------
414 0004ef04 directory_length 4, crc 61188
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 d1000002 --> unit directory at 428
424 8d000005 --> eui-64 leaf at 438
unit directory at 428
-----------------------------------------------------------------
428 00035556 directory_length 3, crc 21846
42c 120001f2 specifier id
430 13000019 version
434 17100800 model
Takashi Sakamoto [Tue, 19 May 2020 11:16:33 +0000 (20:16 +0900)]
ALSA: firewire-motu: add model-specific table of chunk count
In MOTU protocol, data block consists of SPH and 24-bit chunks
aligned to quadlet. The number of chunks per data block is specific
to model. For models with optical interface, the number differs
depending on I/O settings for the interface (ADAT, TOSLINK).
Currently the number is calculated from flags in model-specific
data. However this is weak in the case that the model has quirks.
Actually, for quirks of some models, flags are used against their
original meanings.
This commit adds model-specific table of chunk count. For future
integration, this table is based on the calculation.
Scott Bahling [Mon, 18 May 2020 17:57:28 +0000 (19:57 +0200)]
ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") we
enabled the ADCIII ports using the model=staudio option but forgot
this part to ensure the STDSP24 card is initialized properly.
The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec
requires a similar workaround for Clevo laptops to enforce the
DAC/mixer connection path. Set up a quirk entry for that.
Brent Lu [Mon, 18 May 2020 04:30:38 +0000 (12:30 +0800)]
ALSA: pcm: fix incorrect hw_base increase
There is a corner case that ALSA keeps increasing the hw_ptr but DMA
already stop working/updating the position for a long time.
In following log we can see the position returned from DMA driver does
not move at all but the hw_ptr got increased at some point of time so
snd_pcm_avail() will return a large number which seems to be a buffer
underrun event from user space program point of view. The program
thinks there is space in the buffer and fill more data.
This is because the hw_base will be increased by runtime->buffer_size
frames unconditionally if the hw_ptr is not updated for over half of
buffer time. As the hw_base increases, so does the hw_ptr increased
by the same number.
The avail value returned from snd_pcm_avail() could exceed the limit
(buffer_size) easily becase the hw_ptr itself got increased by same
buffer_size samples when the corner case happens. In following log,
the buffer_size is 16368 samples but the avail is 21810 samples so
CRAS server complains about it.
cras_server[1907]: pcm_avail returned frames larger than buf_size:
sof-glkda7219max: :0,5: 21810 > 16368
By updating runtime->hw_ptr_jiffies each time the HWSYNC is called,
the hw_base will keep the same when buffer stall happens at long as
the interval between each HWSYNC call is shorter than half of buffer
time.
Following is a log captured by a patched kernel. The hw_base/hw_ptr
value is fixed in this corner case and user space program should be
aware of the buffer stall and handle it.
Takashi Iwai [Sat, 16 May 2020 06:28:54 +0000 (08:28 +0200)]
ALSA: hda: Unexport some local helper functions
snd_hdac_bus_queue_event() and snd_hdac_bus_exec_verb() are used only
internally in HD-audio core. Let's drop the exports and move the
declarations into local.h.
Takashi Iwai [Sat, 16 May 2020 06:25:56 +0000 (08:25 +0200)]
ALSA: hda: Fix potential race in unsol event handler
The unsol event handling code has a loop retrieving the read/write
indices and the arrays without locking while the append to the array
may happen concurrently. This may lead to some inconsistency.
Although there hasn't been any proof of this bad results, it's still
safer to protect the racy accesses.
This patch adds the spinlock protection around the unsol handling loop
for addressing it. Here we take bus->reg_lock as the writer side
snd_hdac_bus_queue_event() is also protected by that lock.
Erwin Burema [Sun, 10 May 2020 18:29:11 +0000 (20:29 +0200)]
ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback
For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.
This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")
Takashi Iwai [Thu, 14 May 2020 16:05:33 +0000 (18:05 +0200)]
ALSA: hda/realtek - Limit int mic boost for Thinkpad T530
Lenovo Thinkpad T530 seems to have a sensitive internal mic capture
that needs to limit the mic boost like a few other Thinkpad models.
Although we may change the quirk for ALC269_FIXUP_LENOVO_DOCK, this
hits way too many other laptop models, so let's add a new fixup model
that limits the internal mic boost on top of the existing quirk and
apply to only T530.
Takashi Iwai [Tue, 12 May 2020 07:32:03 +0000 (09:32 +0200)]
ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA
ASUS ZenBook UX431DA requires an additional COEF setup when booted
from the recent Windows 10, otherwise it produces the noisy output.
The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly
due to the pop noise reduction.
Takashi Sakamoto [Sun, 10 May 2020 07:43:01 +0000 (16:43 +0900)]
ALSA: fireface: add support for RME Fireface UFX (untested)
Fireface UFX was shipped by RME GmbH in 2010, and now discontinued.
Although this model has some enhanced feature which Fireface 802
doesn't have (e.g. on-board USB mass storage device class, configuration
interface with color display), the functionality relevant to
packet communication on IEEE 1394 bus seems to be the same as
Fireface 802 (e.g. available number of channels for PCM frame in
each sampling transfer frequency).
With the assumption, this commit adds support for Fireface UFX. In ALSA
fireface driver, these two models are handled as the same one.
Takashi Sakamoto [Sun, 10 May 2020 07:43:00 +0000 (16:43 +0900)]
ALSA: fireface: add support for RME FireFace 802
Fireface 802 was shipped by RME GmbH in 2014. This model supports later
protocol for management of isochronous communication and synchronization
of sampling transmission frequency.
This model consists of below ICs:
* TI TSB41AB2
* Xilinx Spartan-6 FPGA XC6SLX16
* TI TMS320 C6747
* SMSC USB3250
Especially, this model just supports IEEE 1394a, against its name which
evokes Fireface 800.
This commit adds support for Fireface 802 (tested). Userspace applications
can transfer PCM frames and MIDI messages via ALSA PCM/Rawmidi interface.
I note that 4 channels for ADAt1 and ADAT2 are disabled at higher sampling
transfer frequency since isochronous resources reservation fails due to
bandwidth limitation of IEEE 1394a.
The value read from LATTER_SYNC_STATUS register is slightly different
from the one of Fireface UCX. The higher 4 bits and lower 4 bits are
swapped within the same byte.
Without any assist of userspace application, transmitted MIDI messages
from the device are not going to be processed. For detail, please refer
to my comment in code of latter protocol.
root directory
-----------------------------------------------------------------
414 0005ffff directory_length 5, crc 65535 (should be 9514)
418 0c0083c0 node capabilities per IEEE 1394
41c 03000a35 vendor
420 8100000b --> descriptor leaf at 44c
424 8d000007 --> eui-64 leaf at 440
428 d1000001 --> unit directory at 42c
unit directory at 42c
-----------------------------------------------------------------
42c 0004ffff directory_length 4, crc 65535 (should be 45134)
430 12000a35 specifier id
434 13000005 version
438 17101800 model
43c 81000008 --> descriptor leaf at 45c
Takashi Sakamoto [Sun, 10 May 2020 07:42:58 +0000 (16:42 +0900)]
ALSA: fireface: code refactoring to add enumeration constants for model identification
In RME fireface series, version field of unit directory in configuration
ROM is used to distinguish each model. The value of field is known and
it's better to use enumeration constants for code representation.
This commit adds enumeration constants for model identification.
Takashi Sakamoto [Sun, 10 May 2020 07:42:57 +0000 (16:42 +0900)]
ALSA: fireface: start IR context immediately
In the latter models of RME Fireface series, device start to transfer
packets several dozens of milliseconds. On the other hand, ALSA fireface
driver starts IR context 2 milliseconds after the start. This results
in loss to handle incoming packets on the context.
This commit changes to start IR context immediately instead of
postponement. For Fireface 800, this affects nothing because the device
transfer packets 100 milliseconds or so after the start and this is
within wait timeout.
Mike Pozulp [Sun, 10 May 2020 03:28:37 +0000 (20:28 -0700)]
ALSA: hda/realtek: Add quirk for Samsung Notebook
Some models of the Samsung Notebook 9 have very quiet and distorted
headphone output. This quirk changes the VREF value of the ALC298
codec NID 0x1a from default HIZ to new 100.
[ adjusted to 5.7-base and rearranged in SSID order -- tiwai ]
Takashi Sakamoto [Fri, 8 May 2020 04:36:35 +0000 (13:36 +0900)]
ALSA: firewire-lib: use sequence of syt offset and data block on pool in AMDTP domain
In previous commit, the sequence of syt offset and the number of data
blocks per packet is calculated for pool in AMDTP domain structure in
advance of processing outgoing packets.
This commit uses the sequence for outgoing packet processing to obsolete
per-stream processing of the sequence.
Takashi Sakamoto [Fri, 8 May 2020 04:36:33 +0000 (13:36 +0900)]
ALSA: firewire-lib: add cache for packet sequence to AMDTP domain structure
For future extension, storage is required to store packet sequence in
incoming AMDTP stream to recover media clock for outgoing AMDTP stream.
This commit adds the storage to AMDTP domain for this purpose. The
packet sequence is represented by 'struct seq_desc' which has two
members; syt_offset and the number of data blocks. The size of storage
is decided according to the size of packet queue.
Takashi Sakamoto [Fri, 8 May 2020 04:36:32 +0000 (13:36 +0900)]
ALSA: firewire-lib: code refactoring for data block calculation
When calculating the number of data blocks per packet, some states are
stored in AMDTP stream structure. This is inconvenient when reuse the
calculation from non-stream structure.
This commit applies refactoring to helper function for the calculation
so that the function doesn't touch AMDTP stream structure.
Takashi Sakamoto [Fri, 8 May 2020 04:36:31 +0000 (13:36 +0900)]
ALSA: firewire-lib: code refactoring for syt offset calculation
When calculating syt offset, some states are stored in AMDTP stream
structure. This is inconvenient when reuse the calculation from
non-stream structure.
This commit applies refactoring to helper function for the calculation
so that the function doesn't touch AMDTP stream structure.
Takashi Sakamoto [Fri, 8 May 2020 04:36:30 +0000 (13:36 +0900)]
ALSA: firewire-lib: code refactoring for syt computation
In current implementation for outgoing AMDTP packet, the value of syt
field in CIP header is computed when calculating syt offset. For
future extension, it's convenient to split the computation and
calculation.
Takashi Sakamoto [Fri, 8 May 2020 04:36:29 +0000 (13:36 +0900)]
ALSA: firewire-lib: code refactoring for parameters of packet queue and IRQ timing
Although the parameter for packet queue and IRQ timing is calculated when
AMDTP stream starts, the calculated parameters are the same between
streams in AMDTP domain.
This commit moves the calculation and decide the parameters when AMDTP
domain starts.
Takashi Sakamoto [Fri, 8 May 2020 04:36:28 +0000 (13:36 +0900)]
ALSA: firewire-lib: add reference to domain structure from stream structure
In current implementation, AMDTP domain structure and AMDTP stream
structure has one way of reference from the former to the latter. For
future extension, bidirectional reference is needed.
This commit adds a member into stream structure to refer to domain
structure to which the stream belongs.
Takashi Sakamoto [Fri, 8 May 2020 04:36:27 +0000 (13:36 +0900)]
ALSA: firewire-lib: use macro for maximum value of second in 1394 OHCI isoc descriptor
In descriptor of isochronous context in 1394 OHCI, the field of second
has 3 bit, thus the maximum value is 8. The value is used for correct
cycle calculation.
This commit replaces hard-coded value with macro to obsolete magic
number.
Takashi Sakamoto [Fri, 8 May 2020 04:36:26 +0000 (13:36 +0900)]
ALSA: firewire-lib: fix invalid assignment to union data for directional parameter
Although the value of FDF is used just for outgoing stream, the assignment
to union member is done for both directions of stream. At present this
causes no issue because the value of same position is reassigned later for
opposite stream. However, it's better to add if statement.
Gustavo A. R. Silva [Thu, 7 May 2020 18:52:45 +0000 (13:52 -0500)]
ALSA: fireworks: Replace zero-length array with flexible-array
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertently introduced[3] to the codebase from now on.
Also, notice that, dynamic memory allocations won't be affected by
this change:
"Flexible array members have incomplete type, and so the sizeof operator
may not be applied. As a quirk of the original implementation of
zero-length arrays, sizeof evaluates to zero."[1]
sizeof(flexible-array-member) triggers a warning because flexible array
members have incomplete type[1]. There are some instances of code in
which the sizeof operator is being incorrectly/erroneously applied to
zero-length arrays and the result is zero. Such instances may be hiding
some bugs. So, this work (flexible-array member conversions) will also
help to get completely rid of those sorts of issues.
Gustavo A. R. Silva [Thu, 7 May 2020 19:22:23 +0000 (14:22 -0500)]
ALSA: Replace zero-length array with flexible-array
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertently introduced[3] to the codebase from now on.
Also, notice that, dynamic memory allocations won't be affected by
this change:
"Flexible array members have incomplete type, and so the sizeof operator
may not be applied. As a quirk of the original implementation of
zero-length arrays, sizeof evaluates to zero."[1]
sizeof(flexible-array-member) triggers a warning because flexible array
members have incomplete type[1]. There are some instances of code in
which the sizeof operator is being incorrectly/erroneously applied to
zero-length arrays and the result is zero. Such instances may be hiding
some bugs. So, this work (flexible-array member conversions) will also
help to get completely rid of those sorts of issues.
Takashi Iwai [Thu, 7 May 2020 11:44:56 +0000 (13:44 +0200)]
ALSA: rawmidi: Fix racy buffer resize under concurrent accesses
The rawmidi core allows user to resize the runtime buffer via ioctl,
and this may lead to UAF when performed during concurrent reads or
writes: the read/write functions unlock the runtime lock temporarily
during copying form/to user-space, and that's the race window.
This patch fixes the hole by introducing a reference counter for the
runtime buffer read/write access and returns -EBUSY error when the
resize is performed concurrently against read/write.
Note that the ref count field is a simple integer instead of
refcount_t here, since the all contexts accessing the buffer is
basically protected with a spinlock, hence we need no expensive atomic
ops. Also, note that this busy check is needed only against read /
write functions, and not in receive/transmit callbacks; the race can
happen only at the spinlock hole mentioned in the above, while the
whole function is protected for receive / transmit callbacks.
Jason Yan [Wed, 6 May 2020 06:17:16 +0000 (14:17 +0800)]
ALSA: hda: Return true,false for return type bool
Fix the following coccicheck warning:
include/sound/hdaudio.h:210:73-74: WARNING: return of 0/1 in function
'snd_hdac_is_in_pm' with return type bool
include/sound/hdaudio.h:211:76-77: WARNING: return of 0/1 in function
'snd_hdac_is_power_on' with return type bool
Sameer Pujar [Mon, 4 May 2020 08:16:16 +0000 (13:46 +0530)]
ALSA: hda/tegra: workaround playback failure on Tegra194
Tegra194 has 4 SDO lines and with this configuration playback fails
for 44.1K/48K, 2-channel and 16-bps. It results in below print,
"aplay: pcm_write:2011: write error: Input/output error"
Below relation is used to derive stripe control and is referenced
from HD Audio Specification: Revision 1.0a.
{ ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
Due to a legacy HW design problem, playback issue is hit while using
a stripe value resulting from above formula when ratio is '8'. Thus
it is recommended that the ratio must be greater than '8'. Since the
number of SDO lines is in powers of 2, next available ratio '16' is
used as a limiting factor on Tegra194 to workaround the problem.
Sameer Pujar [Mon, 4 May 2020 08:16:15 +0000 (13:46 +0530)]
ALSA: hda: add member to store ratio for stripe control
Stripe control programming is governed by following formula, which is
referenced from the HD Audio specification(Revision 1.0a).
{ ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
Currently above is implemented in snd_hdac_get_stream_stripe_ctl().
This patch introduces a structure member to store the default factor
of '8'. If any HW wants to use a different value, this member can be
easily updated.
Sameer Pujar [Mon, 4 May 2020 08:16:14 +0000 (13:46 +0530)]
ALSA: hda/tegra: correct number of SDO lines for Tegra194
Tegra194 supports 4 SDO lines but GCAP register indicates 2 lines. Thus it
does not reflect the true capability of the HW. This patch presents a
workaround by updating NSDO value accordingly in T_AZA_DBG_CFG_2 register.
Andrew Oakley [Sun, 3 May 2020 14:16:39 +0000 (15:16 +0100)]
ALSA: usb-audio: add mapping for ASRock TRX40 Creator
This is another TRX40 based motherboard with ALC1220-VB USB-audio
that requires a static mapping table.
This motherboard also has a PCI device which advertises no codecs. The
PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD
vendor ID, don't blacklist for now in case other boards have a working
audio device with the same ssid.
Takashi Sakamoto [Sun, 3 May 2020 04:57:18 +0000 (13:57 +0900)]
ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints format
The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current
printk format for the event includes 'sizeof(u8)' macro expected to be
extended in compilation time. However, this is not done. As a result,
perf tools cannot parse the event for printing:
Vasily Khoruzhick [Sat, 2 May 2020 19:31:20 +0000 (12:31 -0700)]
ALSA: line6: Add poll callback for hwdep
At least POD HD500 uses message-based communication, both sides can
send messages. Add poll callback so application can wait for device
messages without using busy loop.
Cover with a proper ifdef around the variable declaration for fixing
the following compilation warning without CONFIG_LEDS_TRIGGER_AUDIO:
sound/pci/hda/patch_realtek.c: In function 'alc_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4134:6: warning: unused variable 'err' [-Wunused-variable]
gcc-10 points out a few instances of suspicious integer arithmetic
leading to value truncation:
sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure':
sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask'
351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c: In function 'snd_miro_configure':
sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask'
1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~
These are all harmless here as only the low 8 bit are passed down
anyway. Change the macros to inline functions to make the code
more readable and also avoid the warning.
Strictly speaking those functions also need locking to make the
read/write pair atomic, but it seems unlikely that anyone would
still run into that issue.
Kai Vehmanen [Tue, 28 Apr 2020 12:38:36 +0000 (15:38 +0300)]
ALSA: hda/hdmi: fix race in monitor detection during probe
A race exists between build_pcms() and build_controls() phases of codec
setup. Build_pcms() sets up notifier for jack events. If a monitor event
is received before build_controls() is run, the initial jack state is
lost and never reported via mixer controls.
The problem can be hit at least with SOF as the controller driver. SOF
calls snd_hda_codec_build_controls() in its workqueue-based probe and
this can be delayed enough to hit the race condition.
Fix the issue by invalidating the per-pin ELD information when
build_controls() is called. The existing call to hdmi_present_sense()
will update the ELD contents. This ensures initial monitor state is
correctly reflected via mixer controls.
Hui Wang [Mon, 27 Apr 2020 03:00:39 +0000 (11:00 +0800)]
ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter
This new Lenovo ThinkCenter has two front mics which can't be handled
by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change
the location for one of the mics.
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)
[ This is again a forward-port of the fix applied for 5.6-base code
(commit 4285de0725b1) to 5.7-base, hence neither Fixes nor
Cc-to-stable tags are included here -- tiwai ]
The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow")
are put in the wrong places mistakenly, which leads to the expected
(repeated) sound when the rate plugin is involved. Fix in the right
places.
Also, at those right places, the zero check is needed for the
termination node, so added there as well, and let's get it done,
finally.
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly
The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow")
are put in the wrong places mistakenly, which leads to the expected
(repeated) sound when the rate plugin is involved. Fix in the right
places.
Also, at those right places, the zero check is needed for the
termination node, so added there as well, and let's get it done,
finally.
ALSA: usb-audio: Fix racy list management in output queue
The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.
Do delete the link at the right place inside the spinlock.
Alexander Tsoy [Fri, 24 Apr 2020 02:24:48 +0000 (05:24 +0300)]
ALSA: usb-audio: Improve frames size computation
For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.
But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).
This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.
ALSA: hda: Match both PCI ID and SSID for driver blacklist
The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
Since the empty codec problem appear on the certain AMD platform (PCI
ID 1022:1487), this patch changes the blacklist matching to both PCI
ID and SSID using pci_match_id(). Also, the entry that was removed by
the previous fix for ASUS ROG Zenigh II is re-added.
NHLT fetch based on _DSM prevents ACPI table override mechanism from
being utilized. Make use of acpi_get_table to enable it and get rid of
redundant code. In consequence, NHLT can be overridden just like any
other ACPI table, e.g.: DSDT or SSDT.
Change has been verified on all Intel AVS architecture platforms, RVP
and production laptops both.
Change possible due to addition of NHLT signature to the list of
standard ACPI tables:
https://patchwork.kernel.org/patch/11463235/
Override helps not only with debug purposes but also allows user for
table adjustment when one found on their production hardware is invalid.
Shared official NHLT spec is now available to community at:
https://01.org/blogs/intel-smart-sound-technology-audio-dsp
NHLT support for iASL is still ongoing subject but should be available
in nearest future.
ALSA: hda: Always use jackpoll helper for jack update after resume
HD-audio codec driver applies a tricky procedure to forcibly perform
the runtime resume by mimicking the usage count even if the device has
been runtime-suspended beforehand. This was needed to assure to
trigger the jack detection update after the system resume.
And recently we also applied the similar logic to the HD-audio
controller side. However this seems leading to some inconsistency,
and eventually PCI controller gets screwed up.
This patch is an attempt to fix and clean up those behavior: instead
of the tricky runtime resume procedure, the existing jackpoll work is
scheduled when such a forced codec resume is required. The jackpoll
work will power up the codec, and this alone should suffice for the
jack status update in usual cases. If the extra polling is requested
(by checking codec->jackpoll_interval), the manual update is invoked
after that, and the codec is powered down again.
Also, we filter the spurious wake up of the codec from the controller
runtime resume by checking codec->relaxed_resume flag. If this flag
is set, basically we don't need to wake up explicitly, but it's
supposed to be done via the audio component notifier.
Xiyu Yang [Thu, 23 Apr 2020 04:54:19 +0000 (12:54 +0800)]
ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".
When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.
The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.
Fix this issue by jumping to "end" label when those error scenarios
occur.
Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses") Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.
My first thought was that this could be easily deduced from the
associated terminals, but some of them have even no associate terminal
ID, hence it's not too trivial to figure out.
Since the number of such terminals are small and limited, this patch
implements another quirk table for the simple mapping of the
connectors. It's not really scalable, but let's hope that there will
be not many such funky devices in future.
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