snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the T+A VID to the generic check in order to enable
native DSD support for T+A devices. This works with the new T+A USB
DAC model SD3100HV and will also work with future devices which
support the XMOS/Thesycon style DSD format.
Signed-off-by: Udo Eberhardt <udo.eberhardt@thesycon.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
XMOS/Thesycon family of USB Audio Class firmware flags DSD altsetting
separate from the PCM ones. Thus the DSD altsetting can be auto-detected
based on the flag and doesn't need maintaining specific altsetting
whitelist.
In addition, static VID:PID-to-altsetting whitelisting causes problems
when firmware update changes the altsetting, or same VID:PID is reused
for another device that has different kind of firmware.
This patch removes existing explicit whitelist mappings for XMOS VID
(0x20b1) and Thesycon VID (0x152a).
Also corrects placement of Hegel HD12 and NuPrime DAC-10 to keep list
sorted based on VID.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Signed-off-by: Olek Poplavsky <woodenbits@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/usb/mixer.c: In function 'parse_audio_feature_unit':
sound/usb/mixer.c:1838:28: warning:
variable 'first_ch_bits' set but not used [-Wunused-but-set-variable]
It never used since 2.6
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix typo after a recent commit causing headphones to have no sound
Fixes: ad43d528a7 (ALSA: usb-audio: Define registers for CM6206)
Signed-off-by: Amadeusz Sławiński <amade@asmblr.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix checks if snd_card_register() fails, and if so logs the error
via dev_err() consistent with other patches.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Reported-by: Hui Peng <benquike@163.com>
Tested-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had some sanity checks of the mixer unit descriptors but they
are too loose and some corner cases are overlooked. Add more strict
checks in uac_mixer_unit_get_channels() for avoiding possible OOB
accesses by malformed descriptors.
This also changes the semantics of uac_mixer_unit_get_channels()
slightly. Now it returns zero for the cases where the descriptor
lacks of bmControls instead of -EINVAL. Then the caller side skips
the mixer creation for such unit while it keeps parsing it.
This corresponds to the case like Maya44.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser for the processing unit reads bNrInPins field before the
bLength sanity check, which may lead to an out-of-bound access when a
malformed descriptor is given. Fix it by assignment after the bLength
check.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49ab ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide
useful string for the vendor and product names too. In order to share
the UCM with WD15, here we keep the profile_name same as the WD15.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The register map for CM6206 is known and we can define what
the values written to the different registers actually mean.
I tested to print the hex values before/after this change,
there is no functional change.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in caiaq driver.
Note that this seems necessary to be put exactly before the next
label, so it's outside the ifdef block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds several vendor specific mixer quirks for RME's Class Compliant
USB devices. These provide extra status information from the device
otherwise not available.
These include AES/SPDIF rate and status information, current system
sampling rate and measured frequency. This information is especially
useful in cases where device's clock is slaved to external clock
source.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quirk in snd-usb-audio was added to automate setting sample rate to
4800k and remove the previously exposed nonfunctional microphone for
the Bowers & Wilkins PX:
commit 240a8af929https://lore.kernel.org/patchwork/patch/919689/
However the headphones where updated shortly after that to remove the
unintentional microphone functionality. I guess because of this the
headphones now crash when connecting them via USB while the quirk is
active. Dmesg:
snd-usb-audio: probe of 2-3:1.0 failed with error -22
usb 2-3: 2:1: cannot get min/max values for control 2 (id 2)
This patch removes the microfone and allows the headphones to connect
and work out of the box. It is based on the current mainline kernel
and successfully applied an tested on my machine (4.18.10.arch1-1).
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Nicolas Huaman <nicolas@herochao.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For discarding the pending bytes on rawmidi, we process with a loop of
snd_rawmidi_transmit() which is just a waste of CPU power.
Implement a lightweight API function to discard the pending bytes and
the proceed the ring buffer instantly, and use it instead of open
codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 1357413 ("Missing break in switch")
Addresses-Coverity-ID: 114917 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sizeof() when applied to a pointer typed expression gives the
size of the pointer, not that of the pointed data.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 115084 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer 'urb' is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'urb' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of UAC3 Power Domains associated to an Audio Streaming
path within the PCM's logic. This means, when there is no audio
being transferred (pcm is closed), the host will set the Power Domain
associated to that substream to state D1. When audio is being transferred
(from hw_params onwards), the Power Domain will be set to D0 state.
This is the way the host lets the device know which Terminal
is going to be actively used and it is for the device to
manage its own internal resources on that UAC3 Power Domain.
Note the resume method now sets the Power Domain to D1 state as
resuming the device doesn't mean audio streaming will occur.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the UAC3 Power Domain state for an Audio Streaming interface
to D2 state before suspending the device (usb_driver callback).
This lets the device know there is no intention to use any of the
Units in the Audio Function and that the host is not going to
even listen for wake-up events (interrupts) on the units.
When the usb_driver gets resumed, the state D0 (fully powered) will
be set. This ties up the UAC3 Power Domains to the runtime PM.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power Domains in the UAC3 spec are mainly intended to be
associated to an Input or Output Terminal so the host
changes the power state of the entire capture or playback
path within the topology.
This patch adds support for finding Power Domains associated
to an Audio Streaming Interface (bTerminalLink) and adds a
reference to them in the usb audio substreams (snd_usb_substream).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Thee USB Audio Class 3 (UAC3) introduces Power Domains as a new
feature to let a host turn individual parts of an audio function
to different power states via USB requests. This lets the device
get to know a bit amore about what the host is up to in order to
optimize power consumption efficiently.
The Power Domains are optional for UAC3 configuration but all
UAC3 devices shall include at least one BADD configuration where
the support for Power Domains is compulsory.
This patch adds a set of features/helpers to parse these power
domains and change their status.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and
snd_usb_ctl_msg_quirk() are never called in atomic context.
They call mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep() and usleep_range().
This is found by a static analysis tool named DCNS written by myself.
Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_usb_feature_unit_ctl properly in mixer.h. Otherwise it's
error-prone.
This fixes the sparse warning:
sound/usb/mixer.c:1464:25: warning: symbol 'snd_usb_feature_unit_ctl' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AU0828_DEVICE() macro in quirks-table.h uses USB_DEVICE_VENDOR_SPEC()
for expanding idVendor and idProduct fields. However, the latter
macro adds also match_flags and bInterfaceClass, which are different
from the values AU0828_DEVICE() macro sets after that.
For fixing them, just expand idVendor and idProduct fields manually in
AU0828_DEVICE().
This fixes sparse warnings like:
sound/usb/quirks-table.h:2892:1: warning: Initializer entry defined twice
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD playback support for the Encore mDSD USB DAC by
specifying the vendor and product ID's
Signed-off-by: Jeff Crukley <jcrukley@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Thesycon provides solutions to XMOS chips, and has its own device
vendor id.
In this patch, we use generic method to detect DSD capability of
Thesycon-based UAC2 implementations in order to support a wide range
of current and future devices.
The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC
hence enable native DSD playback up to DSD512 format.
Signed-off-by: Yue Wang <yuleopen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the audio device is externally clocked and set to a rate that does
not match the external clock, the clock will never be valid and we cannot
set the rate successfully. To fix this, allow a rate change even if
the clock is initially invalid, and validate again after the rate is
changed.
This fixes problems with MOTU UltraLite AVB hardware over USB.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL. As the situation has never
changed over decades, let's rip it off.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch refactors the processing units min/max calculation logic
for the mixer controls and fixes an issue where the Mode Select
checking of the Up/Down mixers doesn't differentiate between the
UAC1 and UAC2 Control Selector (0x02) and the UAC3 one which is
different (0x01).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Audio Control interface descriptor subtypes do not match
across all the UAC versions. That makes reusability of the
"virtual type" (Mixer, Processors, Selectors, etc) terminals
difficult. It also makes the mixer get the default names for
the virtual terminals wrong due to the overlap.
This patch proposes an unified approach by always using the most
comprehensive spec version to define them all (in this case UAC3).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Processig Units defined in
the UAC3 spec. The main difference with the previous specs
is the lack of on/off switches in the controls for these
units and the addiction of the new Multi Function Processing
Unit.
The current version of the UAC3 spec doesn't define any
useful controls for the new Multi Function Processing Unit
so no control will get created once this unit is parsed.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current support for UAC2 Processing Units does the parsing
as one control per bit in the bitmap. However, the UAC2 spec
defines the controls as bit pairs where b01 means read-only
and b11 means read/write control.
This patch fixes that and uses the helper functions for checking
controls readability/writability when the control is defined as
bit pairs (UAC2 and UAC3).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support for Selector Units and Clock Selector Units
defined in the new UAC3 spec.
Selector Units play a really important role in the new UAC3 spec as
Processing Units do not define an on/off switch control anymore.
This forces topology designers to add bypass paths in the topology
to enable/dissable the Processing Units.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The get_seconds() function is deprecated because it truncates the
timestamp to 32 bits, so all users should change to ktime_get_seconds()
or ktime_get_real_seconds().
The firmware interface for passing the timestamp is also limited to
32 bits, so this patch only has the cosmetic effect of avoiding the
old interface.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here is a collection of small fixes on top of the previous update.
All small and obvious fixes. Mostly for usual suspects, USB-audio and
HD-audio, but a few trivial error handling fixes for misc drivers as
well.
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Merge tag 'sound-fix-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here is a collection of small fixes on top of the previous update.
All small and obvious fixes. Mostly for usual suspects, USB-audio and
HD-audio, but a few trivial error handling fixes for misc drivers as
well"
* tag 'sound-fix-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Always create the interrupt pipe for the mixer
ALSA: usb-audio: Add insertion control for UAC3 BADD
ALSA: usb-audio: Change in connectors control creation interface
ALSA: usb-audio: Add bi-directional terminal types
ALSA: lx6464es: add error handling for pci_ioremap_bar
ALSA: sonicvibes: add error handling for snd_ctl_add
ALSA: usb-audio: Remove explicitly listed Mytek devices
ALSA: usb-audio: Generic DSD detection for XMOS-based implementations
ALSA: usb-audio: Add native DSD support for Mytek DACs
ALSA: hda/realtek - Add shutup hint
ALSA: usb-audio: Disable the quirk for Nura headset
ALSA: hda: add dock and led support for HP ProBook 640 G4
ALSA: hda: add dock and led support for HP EliteBook 830 G5
ALSA: emu10k1: add error handling for snd_ctl_add
ALSA: fm801: add error handling for snd_ctl_add
An UAC3 BADD device may also include an interrupt status pipe
to report changes on the HEADSET ADAPTER terminals. The creation
of the status pipe is dependent on the device reporting that it
has it.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HEADSET ADAPTER profile for BADD devices is meant to support
Insertion Control for the Input and Output Terminals of the headset.
This patch defines the BADD inferred input and output terminals and
builds the connector controls.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change build_connector_control() and get_connector_control_name()
so they take `struct usb_mixer_interface` as input argument instead
of `struct mixer_build`.
This is preliminary work to add support for connectors control
for UAC3 BADD devices. No functional change.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use more generic method to detect DSD capability of XMOS-based UAC2
implementations in order to support future devices without having to
explicitly list every device separately.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new mostly generic code with Mytek VID to support native DSD mode.
This implementation should be easier to maintain when manufacturers
release new products.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 33193dca67 ("ALSA: usb-audio: Add a quirk for Nura's
first gen headset") added a quirk for Nura headset with USB ID
0a12:1243, with a hope that it doesn't conflict with others.
Unfortunately, other devices (e.g. Philips Wecall) with the very same
ID got broken by this change, spewing an error like:
usb 2-1.8.2: 2:1: cannot set freq 48000 to ep 0x3
Until we find a proper solution, fix the regression at first by
disabling the added quirk entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199905
Fixes: 33193dca67 ("ALSA: usb-audio: Add a quirk for Nura's first gen headset")
Reviewed-by: Martin Peres <martin.peres@free.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check on err is redundant as both the true and false paths
end up on a break statement. Remove the redundant check.
Detected by CoverityScan, CID#1268773 ("Identical code for different
branches")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as
this serves merely as an intermediate buffer that is copied to each
URB transfer buffer. This works well in general on x86, but on some
archs this may result in cache coherency issues when mmap is used.
OTOH, it works also on such arch unless mmap is used.
This patch is a step for mitigating the inconvenience; a new module
option "use_vmalloc" is provided so that user can choose to allocate
the DMA coherent buffer instead of the existing vmalloc buffer.
The drawback is that it'd be the standard dma_alloc_coherent() calls
and the system would require contiguous pages on non-x86 archs.
Note that it's a global option and not dynamically switchable since
the buffer is pre-allocated at the probe time. In theory, it's
possible to be switchable, but it'd be trickier and racier.
As default use_vmalloc option is set to true, so that the old behavior
is kept. For allowing the coherent mmap on ARM or MIPS, pass
use_vmalloc=0 option explicitly.
Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the device management by replacing the lowlevel device object
allocation with the card->private_data. Nowadays there is almost no
advantage by the lowlevel device, and with card->private_data, the
code becomes cleaner.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid if ((err = ...) style and expand to multiple lines instead.
No change in the end result, but just the beautification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that we can avoid the extra goto lines.
Also beautify the code to follow the standard codex.
No functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stream direction in open and close callbacks can be retrieved from
substream->direction, hence we don't have to stick with the unique PCM
ops hard-coded for each direction. Rewrite the common open/close
callback functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the superfluous #ifndef checks that had been put just for
allowing building the alsa-driver kernel modules externally.
Since the external build was discontinued years ago, let's clean up
the old kludges.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC3 channel map is created during interface parsing,
and in some cases was not freed in failure paths.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
retire_capture_urb() may print warning messages when the given URB
doesn't align, and this may flood the system log easily.
Put the rate limit to the message for avoiding it.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Obtain the number of channels for the Input Terminal from the
Logical Cluster Descriptor. This achieves a useful minimal parsing
of this unit so it can be used in other units in the topology.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the UAC3 insertion controls. The status
is reported as a boolean value in the same way it used to do
for UAC2. Hence, the presence of any connector in the response
will make the control saying the jack is connected.
The UAC2 support for this control has been moved to a dedicated
control for connectors as both UAC2 and UAC3 follow a specific
Control Request Parameter Block for this control. This parameter
block for UAC3 could not be read in the same simplistic
manner as in UAC2.
This implementation is not requesting additional information
from the HIGH CAPABILITY Connectors descriptor.
Tested with an UAC3 device with UAC2 as legacy configuration.
The connector status can be read with `amixer` and the interrupt
is also caught with `alsactl monitor`.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the MIXER UNIT in UAC3. All the information
is obtained from the (HIGH CAPABILITY) Cluster's header. We don't
read the rest of the logical cluster to obtain the channel config
as that wont make any difference in the current mixer behaviour.
The name of the mixer unit is not yet requested as there is not
support for the UAC3 Class Specific String requests.
Tested in an UAC3 device working as a HEADSET with a basic mixer
unit (same as the one in the BADD spec) with no controls.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
bmAtributes offset doesn't exist in the UAC3 CS_EP descriptor.
Hence, checking for pitch control as if it was UAC2 doesn't make
any sense. Use the defined UAC3 offsets instead.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently it's not possible to set volume lower than 26% (it just mutes).
Also fixes this warning:
Warning! Unlikely big volume range (=9472), cval->res is probably wrong.
[13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1
, and volume works fine for full range.
Signed-off-by: Federico Cuello <fedux@fedux.com.ar>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Wait for Axe-Fx III to fully bootup before initializing card.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently released USB Audio Class 3.0 specification
contains BADD (Basic Audio Device Definition) document
which describes pre-defined UAC3 configurations.
BADD support is mandatory for UAC3 devices, it should be
implemented as a separate USB device configuration.
As per BADD document, class-specific descriptors
shall not be included in the Device’s Configuration
descriptor ("inferred"), but host can guess them
from BADD profile number, number of endpoints and
their max packed sizes.
This patch adds support of all BADD profiles from the spec
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Functions like snd_usb_copy_string_desc() or
get_term_name() don't actually need mixer_build
state but can use snd_usb_audio structure instead
to get usb device.
This patch has no functional change but prepares
to future UAC3 BADD profiles support which don't
have class-specific descriptors so won't have
mixer_build state.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The control header needs to be read from buffer at this point only
in the case of UAC1 protocol. Move it inside the switch case as other
protocols such as the Basic Audio Device spec will have an empty buffer
that is latter filled as inferred.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
[Ruslan: updated with recently added sanity checks]
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.
Move class-specific parts of uac3 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.
Move class-specific parts of uac1/2 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().
This also broke Blue Microphones workaround (which
relies on audioformat decoded from previous altsetting)
into two parts: prepare quirk flag analyzing previous
altsetting then use it with current altsetting.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function which
became quite long after adding UAC3 spec support.
Move audioformat allocation and initialization
into separate function, this will make easier
future refactoring.
Attributes left in the original func because it'll
be used for UAC3 BADD profiles suport in the future
There is no functional change.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new macro for iterating over mixer element list for
avoiding the open codes in many places. Also the open-coded
container_of() and the forced cast to struct usb_mixer_elem_info are
replaced with another simple macro, too.
No functional changes but just readability improvement.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell WD15 Dock with 0bda:4014 doesn't give any useful strings for the
vendor and the product names. Name them more specifically via quirk,
as well as the UCM profile name.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Historically USB-audio driver sets the card's longname field with the
details of the device and the bus information. It's good per se, but
not preferable when it's referred as the identifier for UCM profile.
This patch adds a quirk profile_name field to override the card's
longname string to a pre-defined one, so that one can create a unique
and consistent ID string for the specific USB device via a quirk table
to be used as a UCM profile name.
The patch does a slight code refactoring to split out the functions to
set shortname and longname fields as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds "Keep Interface" control for each USB-audio device.
The control element is with SND_CTL_IFACE_CARD, so that it won't
appear on any sane mixer applications. For a device that is confirmed
to work well with "keep-interface" mode, user can flip the control via
amixer, e.g.
% amixer -c1 cset iface=CARD,name='Keep Interface' on
and save/restore the state via alsactl.
The reason to provide this via control API is that the behavior must
be pretty depending on the device (and the firmware in it), so it's
not ideal to apply via module option.
For a device that certainly works, we may set it statically via a
quirk table entry. But a device like Dell WD15 dock behaves so
differently depending on the firmware, and we can't set it
statically. So leave this as a dynamic switch each user can adjust
freely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new flag to struct snd_usb_audio for allowing the device
to skip usb_set_interface() calls at changing or closing the stream.
As of this patch, the flag is nowhere set, so it's just a place
holder. The dynamic switching will be added in the following patch.
A background information for this change:
Dell WD15 dock with Realtek chip gives a very long pause at each time
the driver changes the altset, which eventually happens at every PCM
stream open/close and parameter change. As the long pause happens in
each usb_set_interface() call, there is nothing we can do as long as
it's called. The workaround is to reduce calling it as much as
possible, and this flag indicates that behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary change for the upcoming quirk implementation.
Currently USB-audio driver tries to call usb_set_interface() whenever
the format change with interface/altset modification happens. In this
patch, the check is replaced with the comparison of cur_altsetting and
the targeted altsetting pointer, so that the driver may skip the
unnecessary function calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the early commit adcdd0d5a1 ("ALSA: usb-audio: Skip volume
controls triggers hangup on Dell USB Dock"), we add the mixer quirks
for Dell dock to skip two mixer FU's for playback. This supposed that
the device has always the proper initial volume, but it doesn't seem
always correct.
This patch adds the explicit initialization of the volumes to the
fixed 0dB at the device probe time. Also, such a fixup is needed
after the resume, so a new function is hooked to the resume callback
as well.
Bugzilla: http://bugzilla.suse.com/show_bug.cgi?id=1089467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use new return type vm_fault_t for fault handler. For
now, this is just documenting that the function returns
a VM_FAULT value rather than an errno. Once all instances
are converted, vm_fault_t will become a distinct type.
Commit 1c8f422059 ("mm: change return type to vm_fault_t")
Signed-off-by: Souptick Joarder <jrdr.linux@gmail.com>
Reviewed-by: Matthew Wilcox <mawilcox@microsoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The channel mapping is defined by bChRelationship, not bChPurpose.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Michael Drake <michael.drake@codethink.co.uk>
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Dell Dock USB-audio device with 0bda:4014 is behaving notoriously
bad, and we have already applied some workaround to avoid the firmware
hiccup. Yet we still need to skip one thing, the Extension Unit at ID
4, which doesn't react correctly to the mixer ctl access.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1090658
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2 jack detection support introduced the bmControls checks in a
couple of places, but they forgot the endian conversion; the
bmControls of UAC2 terminal descriptor is __le16, not a byte like in
UAC1.
Fixes: 5a222e8494 ("ALSA: usb-audio: UAC2 jack detection")
Tested-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent code refactoring made the argument for some helper
functions to be the explicit UAC_CS_* and UAC2_CS_* value instead of
0-based offset. However, there was one place left forgotten, and it
caused a regression on some devices appearing as the inconsistent
mixer setup.
This patch corrects the forgotten conversion.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199449
Fixes: 21e9b3e931 ("ALSA: usb-audio: fix uac control query argument")
Tested-by: Nazar Mokrynskyi <nazar@mokrynskyi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Based on a downstream patch from Harry ten Berge.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-and-tested-by: wenyi@tianyu-wool.com
Original-by: Harry ten Berge <htenberge@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx III implicit feedback end point and the data sink endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sending MIDI messages to a PODxt through the USB connection shows
"usb_submit_urb failed" in dmesg and the message is not received by
the POD.
The error is caused because in the funcion send_midi_async() in midi.c
there is a call to usb_sndbulkpipe() for endpoint 3 OUT, but the PODxt
USB descriptor shows that this endpoint it's an interrupt endpoint.
Patch tested with PODxt only.
[ The bug has been present from the very beginning in the staging
driver time, but Fixes below points to the commit moving to sound/
directory so that the fix can be cleanly applied -- tiwai ]
Fixes: 61864d844c ("ALSA: move line6 usb driver into sound/usb")
Signed-off-by: Fabián Inostroza <fabianinostroza@udec.cl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC3 clock parser codes lack of the sanity checks for malformed
descriptors like UAC2 parser does. Without it, the driver may lead to
a potential crash.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sanity checks introduced for malformed descriptors loosely check
the given descriptor size, although the size greater than the defined
description is invalid. It was due to a concern of any funky firmware
in the actual products. But this doesn't look hitting, and any sane
products must have the defined descriptors.
So in this patch, we make the validators more strict, allowing only
with the defined descriptor sizes. The value in clock selector
validator is corrected from 5 to 7 to count the two unlisted fields
after baCSourceID[].
Suggested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are lots of open-coded functions to find a clock source,
selector and multiplier. Now there are both v2 and v3, so six
variants.
This patch refactors the code to use a common helper for the main
loop, and define each validator function for each target.
There is no functional change.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We recently made "format" a u64 variable so now static checkers complain
that this shift will wrap around if format is more than 31. I don't
think it makes a difference for runtime, but it's simple to silence the
warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the current exit return path of the ctl_info allocation failure
cval is not being freed resulting in a memory leak. Fix this by kfree'ing
it on the return.
Detected by CoverityScan, CID#1466878 ("Resource Leak")
Fixes: 21e9b3e931 ("ALSA: usb-audio: fix uac control query argument")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the "clock valid" control a global control instead of a mixer
so that it doesn't appear in mixer applications.
Additionally, remove the check for writeability prohibited by spec, and
Use common code to read the control value.
Tested with a UAC2 Audio device that presents a clock validity
control. The control still shows up in /proc usbmixer but not
in alsamixer.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This implements UAC2 jack detection support, presenting
jack status as a boolean read-only mono mixer.
The presence of any channel in the UAC2_TE_CONNECTOR
control for a terminal will result in the mixer saying
the jack is connected.
Mixer naming follows the convention in sound/core/ctljack.c,
terminating the mixer with " Jack".
For additional clues as to which jack is being presented,
the name is prefixed with " - Input Jack" or " - Output Jack"
depending on if it's an input or output terminal.
This is required because terminal names are ambiguous
between inputs and outputs and often duplicated -
Bidirectional terminal types (0x400 -> 0x4FF)
"... may be used separately for input only or output only.
These types require two Terminal descriptors. Both have the same type."
(quote from "USB Device Class Definition for Terminal Types")
Since bidirectional terminal types are common for headphone adapters,
this distinguishes between two otherwise identically-named
jack controls.
Tested with a UAC2 audio device with connector control capability.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Integrate the native DSD support quirk codes of "ITF-USB DSD" based DACs.
Now, "is_itf_usb_dsd_2alts_dac()" and "is_itf_usb_dsd_3alts_dac()" is
integrated into one function "is_itf_usb_dsd_dac()".
So, remove the logic to distinguish UD-501 and UD-501V2 by the
"Product Name".
The integration is possible by changing the following two functions.
- snd_usb_select_mode_quirk():
Change the determination condition of the DSD mode switch command,
from the altset number being used, to the audio format being played.
Actually, this operation is same as playback using ASIO driver in
Windows environment.
- snd_usb_interface_dsd_format_quirk():
To which altset supports native DSD is determined by the number of altsets.
Previously, it's a constant "2" or "3".
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two versions of TEAC UD-501, the normal version and
the vendor updated version(UD-501V2).
They have the same VID/PID, but the num of the altsetting is different,
UD-501 has 2 altsets for stream, and UD-501V2 has 3.
So, add the logic to distinguish them by the Product Name, not by the PID.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add native DSD support quirk for Luxman DA-06 DAC, by adding the
PID/VID 1852:5065.
Rename "is_marantz_denon_dac()" function to "is_itf_usb_dsd_2alts_dac()"
to cover broader device family sharing the same USB audio
implementation(*).
For the same reason, rename "is_teac_dsd_dac()" function to
"is_itf_usb_dsd_3alts_dac()".
(*)
These devices have the same USB controller "ITF-USB DSD", supplied by
INTERFACE Co., Ltd.
"ITF-USB DSD" USB controller has two patterns,
Pattern 1. (2 altsets version)
- Altset 0: for control
- Altset 1: for stream (S32)
- Altset 2: for stream (S32, DSD_U32)
Pattern 2. (3 altsets version)
- Altset 0: for control
- Altset 1: for stream (S16)
- Altset 2: for stream (S32)
- Altset 3: for stream (S32, DSD_U32)
"is_itf_usb_dsd_2alts_dac()" returns true, if the DAC has "Pattern 1"
USB controller, and "is_itf_usb_dsd_3alts_dac()" returns true, if
"Pattern2".
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes code readability and should have no functional change.
Correct uac control query functions to account for the 1-based indexing
of USB Audio Class control identifiers.
The function parameter, u8 control, should be the
constant defined in audio-v2.h to identify the control to be checked for
readability or writeability.
This patch fixes all callers that had adjusted, and makes explicit
the mapping between audio_feature_info[] array index and the associated
control identifier.
Signed-off-by: Andrew Chant <achant@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently released USB Audio Class 3.0 specification
introduces many significant changes comparing to
previous versions, like
- new Power Domains, support for LPM/L1
- new Cluster descriptor
- changed layout of all class-specific descriptors
- new High Capability descriptors
- New class-specific String descriptors
- new and removed units
- additional sources for interrupts
- removed Type II Audio Data Formats
- ... and many other things (check spec)
It also provides backward compatibility through
multiple configurations, as well as requires
mandatory support for BADD (Basic Audio Device
Definition) on each ADC3.0 compliant device
This patch adds initial support of UAC3 specification
that is enough for Generic I/O Profile (BAOF, BAIF)
device support from BADD document.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload USB audio interface parsing function by
moving quirks to a specially designed location (quirks.c)
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture interface doesn't work and the playback interface only
supports 48 kHz sampling rate even though it advertises more rates.
Signed-off-by: Erik Veijola <erik.veijola@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some more devices that need quirks to handle DSD modes correctly.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-and-tested-by: Thomas Gresens <tgresens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The layout of the UAC2 Control request and response varies depending on
the request type. With the current implementation, only the Layout 2
Parameter Block (with the 2-byte sized RANGE attribute) is handled
properly. For the Control requests with the 1-byte sized RANGE attribute
(Bass Control, Mid Control, Tremble Control), the response is parsed
incorrectly.
This commit:
* fixes the wLength field value in the request
* fixes parsing the range values from the response
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirk to ensure a sync endpoint is properly configured.
This patch is a fix for same symptoms on Behringer UFX1204 as patch
from Albertto Aquirre on Dec 8 2016 for Axe-Fx II.
Signed-off-by: Lassi Ylikojola <lassi.ylikojola@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the mindless scripted replacement of kernel use of POLL*
variables as described by Al, done by this script:
for V in IN OUT PRI ERR RDNORM RDBAND WRNORM WRBAND HUP RDHUP NVAL MSG; do
L=`git grep -l -w POLL$V | grep -v '^t' | grep -v /um/ | grep -v '^sa' | grep -v '/poll.h$'|grep -v '^D'`
for f in $L; do sed -i "-es/^\([^\"]*\)\(\<POLL$V\>\)/\\1E\\2/" $f; done
done
with de-mangling cleanups yet to come.
NOTE! On almost all architectures, the EPOLL* constants have the same
values as the POLL* constants do. But they keyword here is "almost".
For various bad reasons they aren't the same, and epoll() doesn't
actually work quite correctly in some cases due to this on Sparc et al.
The next patch from Al will sort out the final differences, and we
should be all done.
Scripted-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull misc vfs updates from Al Viro:
"All kinds of misc stuff, without any unifying topic, from various
people.
Neil's d_anon patch, several bugfixes, introduction of kvmalloc
analogue of kmemdup_user(), extending bitfield.h to deal with
fixed-endians, assorted cleanups all over the place..."
* 'work.misc' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (28 commits)
alpha: osf_sys.c: use timespec64 where appropriate
alpha: osf_sys.c: fix put_tv32 regression
jffs2: Fix use-after-free bug in jffs2_iget()'s error handling path
dcache: delete unused d_hash_mask
dcache: subtract d_hash_shift from 32 in advance
fs/buffer.c: fold init_buffer() into init_page_buffers()
fs: fold __inode_permission() into inode_permission()
fs: add RWF_APPEND
sctp: use vmemdup_user() rather than badly open-coding memdup_user()
snd_ctl_elem_init_enum_names(): switch to vmemdup_user()
replace_user_tlv(): switch to vmemdup_user()
new primitive: vmemdup_user()
memdup_user(): switch to GFP_USER
eventfd: fold eventfd_ctx_get() into eventfd_ctx_fileget()
eventfd: fold eventfd_ctx_read() into eventfd_read()
eventfd: convert to use anon_inode_getfd()
nfs4file: get rid of pointless include of btrfs.h
uvc_v4l2: clean copyin/copyout up
vme_user: don't use __copy_..._user()
usx2y: don't bother with memdup_user() for 16-byte structure
...
Pull poll annotations from Al Viro:
"This introduces a __bitwise type for POLL### bitmap, and propagates
the annotations through the tree. Most of that stuff is as simple as
'make ->poll() instances return __poll_t and do the same to local
variables used to hold the future return value'.
Some of the obvious brainos found in process are fixed (e.g. POLLIN
misspelled as POLL_IN). At that point the amount of sparse warnings is
low and most of them are for genuine bugs - e.g. ->poll() instance
deciding to return -EINVAL instead of a bitmap. I hadn't touched those
in this series - it's large enough as it is.
Another problem it has caught was eventpoll() ABI mess; select.c and
eventpoll.c assumed that corresponding POLL### and EPOLL### were
equal. That's true for some, but not all of them - EPOLL### are
arch-independent, but POLL### are not.
The last commit in this series separates userland POLL### values from
the (now arch-independent) kernel-side ones, converting between them
in the few places where they are copied to/from userland. AFAICS, this
is the least disruptive fix preserving poll(2) ABI and making epoll()
work on all architectures.
As it is, it's simply broken on sparc - try to give it EPOLLWRNORM and
it will trigger only on what would've triggered EPOLLWRBAND on other
architectures. EPOLLWRBAND and EPOLLRDHUP, OTOH, are never triggered
at all on sparc. With this patch they should work consistently on all
architectures"
* 'misc.poll' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (37 commits)
make kernel-side POLL... arch-independent
eventpoll: no need to mask the result of epi_item_poll() again
eventpoll: constify struct epoll_event pointers
debugging printk in sg_poll() uses %x to print POLL... bitmap
annotate poll(2) guts
9p: untangle ->poll() mess
->si_band gets POLL... bitmap stored into a user-visible long field
ring_buffer_poll_wait() return value used as return value of ->poll()
the rest of drivers/*: annotate ->poll() instances
media: annotate ->poll() instances
fs: annotate ->poll() instances
ipc, kernel, mm: annotate ->poll() instances
net: annotate ->poll() instances
apparmor: annotate ->poll() instances
tomoyo: annotate ->poll() instances
sound: annotate ->poll() instances
acpi: annotate ->poll() instances
crypto: annotate ->poll() instances
block: annotate ->poll() instances
x86: annotate ->poll() instances
...
The E1 has two headphone jacks, one of which can be set as a microphone
input. In the default mode, it uses the built-in microphone as an input.
By sending a special command, the second headphone jack is instead used
as an input.
This might work with the E3 as well, but I don't have one of those to
test it.
Signed-off-by: Ian Douglas Scott <ian@iandouglasscott.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture interface does not work, and the playback interface
actually supports only 48kHz unlike what is advertised (44.1, 32, 22,
16, 8).
The only unknown here is if there are other devices that use the same
product ID, but given that this ID is currently unknown, I would assume
it is specially allocated for the nura headset.
Signed-off-by: Martin Peres <martin.peres@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
memdup_user() checks it, so the only effect would be failing with
-EINVAL instead of -EFAULT in case when access_ok() is false.
However, the caller has already checked access_ok() itself (and
would have buggered off with -EFAULT), so the check is completely
pointless. Removing it both simplifies the only instance
of ->dsp_load() and allows to get rid of the check in caller -
its sole effect used to be in preventing a bogus error value
from access_ok() in the instance. Let memdup_user() do the
right thing instead...
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
get_term_name() calls snd_usb_copy_string_desc() for retrieving the
name when a specific ID (name field) is given. When this returns an
error (zero), however, it simply returns as is. This will end up in a
fixed name string in the caller side, which often is meaningless.
For giving a bit more useful name string depending on the terminal
type, change the get_term_name() function to go through the fallback
mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 89b89d121f ("ALSA: usb-audio: Add check return value for
usb_string()") added the check of the return value from
snd_usb_copy_string_desc(), which is correct per se, but it introduced
a regression. In the original code, either the "Clock Source",
"Playback Source" or "Capture Source" suffix is added after the
terminal string, while the commit changed it to add the suffix only
when get_term_name() is failing. It ended up with an incorrect ctl
name like "PCM" instead of "PCM Capture Source".
Also, even the original code has a similar bug: when the ctl name is
generated from snd_usb_copy_string_desc() for the given iSelector, it
also doesn't put the suffix.
This patch addresses these issues: the suffix is added always when no
static mapping is found. Also the patch tries to put more comments
and cleans up the if/else block for better readability in order to
avoid the same pitfall again.
Fixes: 89b89d121f ("ALSA: usb-audio: Add check return value for usb_string()")
Reported-and-tested-by: Mauro Santos <registo.mailling@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds VID:PID of Esoteric D-05X to the TEAC device id's.
Renames the is_teac_50X_dac() function to is_teac_dsd_dac() to cover
broader device family from the same corporation sharing the same USB
audio implementation.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_copy_string_desc() returns zero if usb_string() fails.
In case of failure, we need to check the snd_usb_copy_string_desc()'s
return value and add an exception case
Signed-off-by: Jaejoong Kim <climbbb.kim@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_usb_copy_string_desc() retrieves the usb string corresponding to
the index number through the usb_string(). The problem is that the
usb_string() returns the length of the string (>= 0) when successful, but
it can also return a negative value about the error case or status of
usb_control_msg().
If iClockSource is '0' as shown below, usb_string() will returns -EINVAL.
This will result in '0' being inserted into buf[-22], and the following
KASAN out-of-bound error message will be output.
AudioControl Interface Descriptor:
bLength 8
bDescriptorType 36
bDescriptorSubtype 10 (CLOCK_SOURCE)
bClockID 1
bmAttributes 0x07 Internal programmable Clock (synced to SOF)
bmControls 0x07
Clock Frequency Control (read/write)
Clock Validity Control (read-only)
bAssocTerminal 0
iClockSource 0
To fix it, check usb_string()'return value and bail out.
==================================================================
BUG: KASAN: stack-out-of-bounds in parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
Write of size 1 at addr ffff88007e66735a by task systemd-udevd/18376
CPU: 0 PID: 18376 Comm: systemd-udevd Not tainted 4.13.0+ #3
Hardware name: LG Electronics 15N540-RFLGL/White Tip Mountain, BIOS 15N5
Call Trace:
dump_stack+0x63/0x8d
print_address_description+0x70/0x290
? parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
kasan_report+0x265/0x350
__asan_store1+0x4a/0x50
parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
? save_stack+0xb5/0xd0
? save_stack_trace+0x1b/0x20
? save_stack+0x46/0xd0
? kasan_kmalloc+0xad/0xe0
? kmem_cache_alloc_trace+0xff/0x230
? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio]
? usb_audio_probe+0x4de/0xf40 [snd_usb_audio]
? usb_probe_interface+0x1f5/0x440
? driver_probe_device+0x3ed/0x660
? build_feature_ctl+0xb10/0xb10 [snd_usb_audio]
? save_stack_trace+0x1b/0x20
? init_object+0x69/0xa0
? snd_usb_find_csint_desc+0xa8/0xf0 [snd_usb_audio]
snd_usb_mixer_controls+0x1dc/0x370 [snd_usb_audio]
? build_audio_procunit+0x890/0x890 [snd_usb_audio]
? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio]
? kmem_cache_alloc_trace+0xff/0x230
? usb_ifnum_to_if+0xbd/0xf0
snd_usb_create_mixer+0x25b/0x4b0 [snd_usb_audio]
? snd_usb_create_stream+0x255/0x2c0 [snd_usb_audio]
usb_audio_probe+0x4de/0xf40 [snd_usb_audio]
? snd_usb_autosuspend.part.7+0x30/0x30 [snd_usb_audio]
? __pm_runtime_idle+0x90/0x90
? kernfs_activate+0xa6/0xc0
? usb_match_one_id_intf+0xdc/0x130
? __pm_runtime_set_status+0x2d4/0x450
usb_probe_interface+0x1f5/0x440
Cc: <stable@vger.kernel.org>
Signed-off-by: Jaejoong Kim <climbbb.kim@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the semantics of the enable option for snd-usb-audio
in order to allow users to disable a device specified by either or both
of the vendor id and the product id.
Signed-off-by: Tamaki Nishino <otamachan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 8428a8ebde ("ALSA: usb-audio: Fix potential zero-division
at parsing FU") is utterly bogus and breaks the case with csize=1
instead of fixing anything. Just take it back again.
Reported-by: Jörg Otte <jrg.otte@gmail.com>
Fixes: 8428a8ebde ("ALSA: usb-audio: Fix potential zero-division at parsing FU"
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull timer updates from Thomas Gleixner:
- The final conversion of timer wheel timers to timer_setup().
A few manual conversions and a large coccinelle assisted sweep and
the removal of the old initialization mechanisms and the related
code.
- Remove the now unused VSYSCALL update code
- Fix permissions of /proc/timer_list. I still need to get rid of that
file completely
- Rename a misnomed clocksource function and remove a stale declaration
* 'timers-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (27 commits)
m68k/macboing: Fix missed timer callback assignment
treewide: Remove TIMER_FUNC_TYPE and TIMER_DATA_TYPE casts
timer: Remove redundant __setup_timer*() macros
timer: Pass function down to initialization routines
timer: Remove unused data arguments from macros
timer: Switch callback prototype to take struct timer_list * argument
timer: Pass timer_list pointer to callbacks unconditionally
Coccinelle: Remove setup_timer.cocci
timer: Remove setup_*timer() interface
timer: Remove init_timer() interface
treewide: setup_timer() -> timer_setup() (2 field)
treewide: setup_timer() -> timer_setup()
treewide: init_timer() -> setup_timer()
treewide: Switch DEFINE_TIMER callbacks to struct timer_list *
s390: cmm: Convert timers to use timer_setup()
lightnvm: Convert timers to use timer_setup()
drivers/net: cris: Convert timers to use timer_setup()
drm/vc4: Convert timers to use timer_setup()
block/laptop_mode: Convert timers to use timer_setup()
net/atm/mpc: Avoid open-coded assignment of timer callback function
...
All commits found here are small fixes for regression or stable:
- PCM timestamp behavior fix that could be seen as a regression
- Remove spurious WARN_ON() from ALSA timer 32bit compat ioctl
- HD-audio HDMI/DP channel mapping fix for 32bit archs
- Fix the previous fix for HD-audio initialization code
- More hardening USB-audio against malicious USB descriptors
- HD-audio quirks/fixes (Realtek codec, AMD controller)
- Missing help text for the recent Intel SST kconfig change
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Merge tag 'sound-fix-4.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All commits found here are small fixes for regression or stable:
- PCM timestamp behavior fix that could be seen as a regression
- Remove spurious WARN_ON() from ALSA timer 32bit compat ioctl
- HD-audio HDMI/DP channel mapping fix for 32bit archs
- Fix the previous fix for HD-audio initialization code
- More hardening USB-audio against malicious USB descriptors
- HD-audio quirks/fixes (Realtek codec, AMD controller)
- Missing help text for the recent Intel SST kconfig change"
* tag 'sound-fix-4.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: Add Raven PCI ID
ALSA: hda/realtek - Fix ALC700 family no sound issue
ALSA: hda - Fix yet remaining issue with vmaster 0dB initialization
ALSA: usb-audio: Add sanity checks in v2 clock parsers
ALSA: usb-audio: Fix potential zero-division at parsing FU
ALSA: usb-audio: Fix potential out-of-bound access at parsing SU
ALSA: usb-audio: Add sanity checks to FE parser
ALSA: timer: Remove kernel warning at compat ioctl error paths
ALSA: pcm: update tstamp only if audio_tstamp changed
ALSA: hda/realtek: Add headset mic support for Intel NUC Skull Canyon
ALSA: hda: Fix too short HDMI/DP chmap reporting
ALSA: usb-audio: uac1: Invalidate ctl on interrupt
ALSA: hda/realtek - Fix ALC275 no sound issue
ASoC: Intel: Add help text for SND_SOC_INTEL_SST_TOPLEVEL
With all callbacks converted, and the timer callback prototype
switched over, the TIMER_FUNC_TYPE cast is no longer needed,
so remove it. Conversion was done with the following scripts:
perl -pi -e 's|\(TIMER_FUNC_TYPE\)||g' \
$(git grep TIMER_FUNC_TYPE | cut -d: -f1 | sort -u)
perl -pi -e 's|\(TIMER_DATA_TYPE\)||g' \
$(git grep TIMER_DATA_TYPE | cut -d: -f1 | sort -u)
The now unused macros are also dropped from include/linux/timer.h.
Signed-off-by: Kees Cook <keescook@chromium.org>
The helper functions to parse and look for the clock source, selector
and multiplier unit may return the descriptor with a too short length
than required, while there is no sanity check in the caller side.
Add some sanity checks in the parsers, at least, to guarantee the
given descriptor size, for avoiding the potential crashes.
Fixes: 79f920fbff ("ALSA: usb-audio: parse clock topology of UAC2 devices")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
parse_audio_feature_unit() contains a code dividing potentially with
zero when a malformed FU descriptor is passed. Although there is
already a sanity check, it checks only the value zero, hence it can
still lead to a zero-division when a value 1 is passed there.
Fix it by correcting the sanity check (and the error message
thereof).
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver may trigger an out-of-bound access at parsing a
malformed selector unit, as it checks the header length only after
evaluating bNrInPins field, which can be already above the given
length. Fix it by adding the length check beforehand.
Fixes: 99fc86450c ("ALSA: usb-mixer: parse descriptors with structs")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the usb-audio descriptor contains the malformed feature unit
description with a too short length, the driver may access
out-of-bounds. Add a sanity check of the header size at the beginning
of parse_audio_feature_unit().
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an interrupt occurs, the value of at least one of the belonging
controls should have changed. To make sure they get re-read from device
on the next read, invalidate the cache. This was correctly implemented
for uac2 already, but missing for uac1.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no big surprising changes in this cycle, yet not too
boring, either. The biggest change from diffstat POV is the removal
of the legacy OSS driver codes that have been already disabled for a
long time. This will bring a few trivial merge conflicts.
As new features in ASoC side, there are two things: a new AC97 bus
implementation and AMD Stony platform support. Both include the
relevant changes shared with other subsystems, e.g. AC97 MFD changes
and DRM AMD changes.
Some other highlighted topics are:
- A bunch of USB-audio drivers got the hardening against the malicious
device accesses with a new helper code for endpoint sanity check.
- Lots of cleanups for ASoC Intel platform code, including support for
their open source audio firmware.
- Continued ASoC core componentization works.
- Support for scaling MCLK with sample rate in ASoC simple-card.
- Stabler PCM hot-unplug capability, especially for ASoC usages.
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Merge tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are no big surprising changes in this cycle, yet not too boring,
either. The biggest change from diffstat POV is the removal of the
legacy OSS driver codes that have been already disabled for a long
time. This will bring a few trivial merge conflicts.
As new features in ASoC side, there are two things: a new AC97 bus
implementation and AMD Stony platform support. Both include the
relevant changes shared with other subsystems, e.g. AC97 MFD changes
and DRM AMD changes.
Some other highlighted topics are:
- A bunch of USB-audio drivers got the hardening against the
malicious device accesses with a new helper code for endpoint
sanity check
- Lots of cleanups for ASoC Intel platform code, including support
for their open source audio firmware
- Continued ASoC core componentization works
- Support for scaling MCLK with sample rate in ASoC simple-card
- Stabler PCM hot-unplug capability, especially for ASoC usages"
* tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (302 commits)
Documentation: sound: hd-audio: notes.rst
ASoC: bcm2835: Support left/right justified and DSP modes
ASoC: bcm2835: Enforce full symmetry
ASoC: bcm2835: Support additional samplerates up to 384kHz
ASoC: bcm2835: Add support for TDM modes
ASoC: add mclk-fs support to audio graph card
ASoC: add mclk-fs to audio graph card binding
ASoC: rt5514: work around link error
ASoC: rt5514: mark PM functions as __maybe_unused
ASoC: rt5663: Check the JD status in the button pushing
ASoC: amd: Modified DMA transfer Mechanism for Playback
ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+
ASoC: da7213: add support for DSP modes
ASoC: sun8i-codec: Add a comment on the LRCK inversion
ASoC: sun8i-codec: Set the BCLK divider
ASoC: rt5663: Delay and retry reading rt5663 ID register
ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds
ASoC: cs42l56: Fix reset GPIO name in example DT binding
ASoC: rt5514-spi: check irq status to schedule data copy in resume function
...
Pull core locking updates from Ingo Molnar:
"The main changes in this cycle are:
- Another attempt at enabling cross-release lockdep dependency
tracking (automatically part of CONFIG_PROVE_LOCKING=y), this time
with better performance and fewer false positives. (Byungchul Park)
- Introduce lockdep_assert_irqs_enabled()/disabled() and convert
open-coded equivalents to lockdep variants. (Frederic Weisbecker)
- Add down_read_killable() and use it in the VFS's iterate_dir()
method. (Kirill Tkhai)
- Convert remaining uses of ACCESS_ONCE() to
READ_ONCE()/WRITE_ONCE(). Most of the conversion was Coccinelle
driven. (Mark Rutland, Paul E. McKenney)
- Get rid of lockless_dereference(), by strengthening Alpha atomics,
strengthening READ_ONCE() with smp_read_barrier_depends() and thus
being able to convert users of lockless_dereference() to
READ_ONCE(). (Will Deacon)
- Various micro-optimizations:
- better PV qspinlocks (Waiman Long),
- better x86 barriers (Michael S. Tsirkin)
- better x86 refcounts (Kees Cook)
- ... plus other fixes and enhancements. (Borislav Petkov, Juergen
Gross, Miguel Bernal Marin)"
* 'locking-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (70 commits)
locking/x86: Use LOCK ADD for smp_mb() instead of MFENCE
rcu: Use lockdep to assert IRQs are disabled/enabled
netpoll: Use lockdep to assert IRQs are disabled/enabled
timers/posix-cpu-timers: Use lockdep to assert IRQs are disabled/enabled
sched/clock, sched/cputime: Use lockdep to assert IRQs are disabled/enabled
irq_work: Use lockdep to assert IRQs are disabled/enabled
irq/timings: Use lockdep to assert IRQs are disabled/enabled
perf/core: Use lockdep to assert IRQs are disabled/enabled
x86: Use lockdep to assert IRQs are disabled/enabled
smp/core: Use lockdep to assert IRQs are disabled/enabled
timers/hrtimer: Use lockdep to assert IRQs are disabled/enabled
timers/nohz: Use lockdep to assert IRQs are disabled/enabled
workqueue: Use lockdep to assert IRQs are disabled/enabled
irq/softirqs: Use lockdep to assert IRQs are disabled/enabled
locking/lockdep: Add IRQs disabled/enabled assertion APIs: lockdep_assert_irqs_enabled()/disabled()
locking/pvqspinlock: Implement hybrid PV queued/unfair locks
locking/rwlocks: Fix comments
x86/paravirt: Set up the virt_spin_lock_key after static keys get initialized
block, locking/lockdep: Assign a lock_class per gendisk used for wait_for_completion()
workqueue: Remove now redundant lock acquisitions wrt. workqueue flushes
...
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
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Merge tag 'asoc-v4.15' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.15
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
The amount of the changes isn't as quite small as wished, nevertheless
they are straight fixes that deserve merging to 4.14 final.
Most of fixes are about ALSA core bugs spotted by fuzzer: a follow-up
fix for the previous nested rwsem patch, a fix to avoid the resource
hogs due to too many concurrent ALSA timer invocations, and a fix for
a crash with SYSEX MIDI transfer over OSS sequencer emulation that is
used by none but fuzzer.
The rest are usual HD-audio and USB-audio device-specific quirks,
which are safe to apply.
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Merge tag 'sound-4.14' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The amount of the changes isn't as quite small as wished, nevertheless
they are straight fixes that deserve merging to 4.14 final.
Most of fixes are about ALSA core bugs spotted by fuzzer: a follow-up
fix for the previous nested rwsem patch, a fix to avoid the resource
hogs due to too many concurrent ALSA timer invocations, and a fix for
a crash with SYSEX MIDI transfer over OSS sequencer emulation that is
used by none but fuzzer.
The rest are usual HD-audio and USB-audio device-specific quirks,
which are safe to apply"
* tag 'sound-4.14' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - fix headset mic problem for Dell machines with alc274
ALSA: seq: Fix OSS sysex delivery in OSS emulation
ALSA: seq: Avoid invalid lockdep class warning
ALSA: timer: Limit max instances per timer
ALSA: usb-audio: support new Amanero Combo384 firmware version
The us122l driver creates URBs per the fixed endpoints, and this may
end up with URBs with inconsistent pipes when a fuzzer or a malicious
program deals with the manipulated endpoints. It ends up with a
kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 0 != type 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:471
usb_submit_urb+0x113e/0x1400
Call Trace:
usb_stream_start+0x48a/0x9f0 sound/usb/usx2y/usb_stream.c:690
us122l_start+0x116/0x290 sound/usb/usx2y/us122l.c:365
us122l_create_card sound/usb/usx2y/us122l.c:502
us122l_usb_probe sound/usb/usx2y/us122l.c:588
....
For avoiding the bad access, this patch adds a few sanity checks of
the validity of created URBs like previous similar fixes using the new
usb_urb_ep_type_check() helper function.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many source files in the tree are missing licensing information, which
makes it harder for compliance tools to determine the correct license.
By default all files without license information are under the default
license of the kernel, which is GPL version 2.
Update the files which contain no license information with the 'GPL-2.0'
SPDX license identifier. The SPDX identifier is a legally binding
shorthand, which can be used instead of the full boiler plate text.
This patch is based on work done by Thomas Gleixner and Kate Stewart and
Philippe Ombredanne.
How this work was done:
Patches were generated and checked against linux-4.14-rc6 for a subset of
the use cases:
- file had no licensing information it it.
- file was a */uapi/* one with no licensing information in it,
- file was a */uapi/* one with existing licensing information,
Further patches will be generated in subsequent months to fix up cases
where non-standard license headers were used, and references to license
had to be inferred by heuristics based on keywords.
The analysis to determine which SPDX License Identifier to be applied to
a file was done in a spreadsheet of side by side results from of the
output of two independent scanners (ScanCode & Windriver) producing SPDX
tag:value files created by Philippe Ombredanne. Philippe prepared the
base worksheet, and did an initial spot review of a few 1000 files.
The 4.13 kernel was the starting point of the analysis with 60,537 files
assessed. Kate Stewart did a file by file comparison of the scanner
results in the spreadsheet to determine which SPDX license identifier(s)
to be applied to the file. She confirmed any determination that was not
immediately clear with lawyers working with the Linux Foundation.
Criteria used to select files for SPDX license identifier tagging was:
- Files considered eligible had to be source code files.
- Make and config files were included as candidates if they contained >5
lines of source
- File already had some variant of a license header in it (even if <5
lines).
All documentation files were explicitly excluded.
The following heuristics were used to determine which SPDX license
identifiers to apply.
- when both scanners couldn't find any license traces, file was
considered to have no license information in it, and the top level
COPYING file license applied.
For non */uapi/* files that summary was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 11139
and resulted in the first patch in this series.
If that file was a */uapi/* path one, it was "GPL-2.0 WITH
Linux-syscall-note" otherwise it was "GPL-2.0". Results of that was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 WITH Linux-syscall-note 930
and resulted in the second patch in this series.
- if a file had some form of licensing information in it, and was one
of the */uapi/* ones, it was denoted with the Linux-syscall-note if
any GPL family license was found in the file or had no licensing in
it (per prior point). Results summary:
SPDX license identifier # files
---------------------------------------------------|------
GPL-2.0 WITH Linux-syscall-note 270
GPL-2.0+ WITH Linux-syscall-note 169
((GPL-2.0 WITH Linux-syscall-note) OR BSD-2-Clause) 21
((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) 17
LGPL-2.1+ WITH Linux-syscall-note 15
GPL-1.0+ WITH Linux-syscall-note 14
((GPL-2.0+ WITH Linux-syscall-note) OR BSD-3-Clause) 5
LGPL-2.0+ WITH Linux-syscall-note 4
LGPL-2.1 WITH Linux-syscall-note 3
((GPL-2.0 WITH Linux-syscall-note) OR MIT) 3
((GPL-2.0 WITH Linux-syscall-note) AND MIT) 1
and that resulted in the third patch in this series.
- when the two scanners agreed on the detected license(s), that became
the concluded license(s).
- when there was disagreement between the two scanners (one detected a
license but the other didn't, or they both detected different
licenses) a manual inspection of the file occurred.
- In most cases a manual inspection of the information in the file
resulted in a clear resolution of the license that should apply (and
which scanner probably needed to revisit its heuristics).
- When it was not immediately clear, the license identifier was
confirmed with lawyers working with the Linux Foundation.
- If there was any question as to the appropriate license identifier,
the file was flagged for further research and to be revisited later
in time.
In total, over 70 hours of logged manual review was done on the
spreadsheet to determine the SPDX license identifiers to apply to the
source files by Kate, Philippe, Thomas and, in some cases, confirmation
by lawyers working with the Linux Foundation.
Kate also obtained a third independent scan of the 4.13 code base from
FOSSology, and compared selected files where the other two scanners
disagreed against that SPDX file, to see if there was new insights. The
Windriver scanner is based on an older version of FOSSology in part, so
they are related.
Thomas did random spot checks in about 500 files from the spreadsheets
for the uapi headers and agreed with SPDX license identifier in the
files he inspected. For the non-uapi files Thomas did random spot checks
in about 15000 files.
In initial set of patches against 4.14-rc6, 3 files were found to have
copy/paste license identifier errors, and have been fixed to reflect the
correct identifier.
Additionally Philippe spent 10 hours this week doing a detailed manual
inspection and review of the 12,461 patched files from the initial patch
version early this week with:
- a full scancode scan run, collecting the matched texts, detected
license ids and scores
- reviewing anything where there was a license detected (about 500+
files) to ensure that the applied SPDX license was correct
- reviewing anything where there was no detection but the patch license
was not GPL-2.0 WITH Linux-syscall-note to ensure that the applied
SPDX license was correct
This produced a worksheet with 20 files needing minor correction. This
worksheet was then exported into 3 different .csv files for the
different types of files to be modified.
These .csv files were then reviewed by Greg. Thomas wrote a script to
parse the csv files and add the proper SPDX tag to the file, in the
format that the file expected. This script was further refined by Greg
based on the output to detect more types of files automatically and to
distinguish between header and source .c files (which need different
comment types.) Finally Greg ran the script using the .csv files to
generate the patches.
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Reviewed-by: Philippe Ombredanne <pombredanne@nexb.com>
Reviewed-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Support DSD_U32_BE sample format on new Amanero Combo384 firmware
version on older VID/PID.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Please do not apply this to mainline directly, instead please re-run the
coccinelle script shown below and apply its output.
For several reasons, it is desirable to use {READ,WRITE}_ONCE() in
preference to ACCESS_ONCE(), and new code is expected to use one of the
former. So far, there's been no reason to change most existing uses of
ACCESS_ONCE(), as these aren't harmful, and changing them results in
churn.
However, for some features, the read/write distinction is critical to
correct operation. To distinguish these cases, separate read/write
accessors must be used. This patch migrates (most) remaining
ACCESS_ONCE() instances to {READ,WRITE}_ONCE(), using the following
coccinelle script:
----
// Convert trivial ACCESS_ONCE() uses to equivalent READ_ONCE() and
// WRITE_ONCE()
// $ make coccicheck COCCI=/home/mark/once.cocci SPFLAGS="--include-headers" MODE=patch
virtual patch
@ depends on patch @
expression E1, E2;
@@
- ACCESS_ONCE(E1) = E2
+ WRITE_ONCE(E1, E2)
@ depends on patch @
expression E;
@@
- ACCESS_ONCE(E)
+ READ_ONCE(E)
----
Signed-off-by: Mark Rutland <mark.rutland@arm.com>
Signed-off-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Cc: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: davem@davemloft.net
Cc: linux-arch@vger.kernel.org
Cc: mpe@ellerman.id.au
Cc: shuah@kernel.org
Cc: snitzer@redhat.com
Cc: thor.thayer@linux.intel.com
Cc: tj@kernel.org
Cc: viro@zeniv.linux.org.uk
Cc: will.deacon@arm.com
Link: http://lkml.kernel.org/r/1508792849-3115-19-git-send-email-paulmck@linux.vnet.ibm.com
Signed-off-by: Ingo Molnar <mingo@kernel.org>
In preparation for unconditionally passing the struct timer_list pointer to
all timer callbacks, switch to using the new timer_setup() and from_timer()
to pass the timer pointer explicitly.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Back-merge for applying the timer API conversion patch for line6
driver that conflicts with the recent fix in upstream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer card is being assigned a value but it is never used.
Remove this redundant variable. Cleans up clang warning:
Value stored to 'card' is never read
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add native DSD support quirk for Pro-Ject Pre Box S2 Digital USB id
2772:0230.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
caiaq driver doesn't kill the URB properly at its error path during
the probe, which may lead to a use-after-free error later. This patch
addresses it.
Reported-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Johan Hovold <johan@kernel.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pulling the EP validity checks in USB audio drivers.
It also adds a new helper in USB core, which was acked by Greg.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few other places calling usb_submit_urb() with the URB
composed from the fixed endpoint without validation. For avoiding the
spurious kernel warnings, add the sanity checks to appropriate
places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few other places in caiaq driver have the URB handling with the
fixed endpoints without checking the validity, too. Add the sanity
check with the new helper function at each appropriate place for
avoiding the spurious kernel warnings due to invalid EPs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hiface usb-audio driver sets up URBs containing the fixed endpoints
without validation. This may end up with an oops-like kernel warning
when submitted.
For avoiding it, this patch adds the calls of the new sanity-check
helper for URBs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usx2y driver sets up URBs containing the fixed endpoints without
validation. This may end up with an oops-like kernel warning when
submitted.
For avoiding it, this patch adds the calls of the new sanity-check
helper for URBs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver may set up a URB containing the fixed EP without
validating its presence for some non-class-compliant devices. This
may end up with an oops-like kernel warning when submitted.
For avoiding it, this patch adds the call of the new sanity-check
helper for URBs. The checks are needed only for MIDI I/O as the other
places have already some other checks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently line6 drivers submit a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 3 != type 1
------------[ cut here ]------------
WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 0 PID: 24 Comm: kworker/0:1 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
line6_start_listen+0x55f/0x9e0 sound/usb/line6/driver.c:82
line6_init_cap_control sound/usb/line6/driver.c:690
line6_probe+0x7c9/0x1310 sound/usb/line6/driver.c:764
podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently caiaq driver submits a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 3 != type 1
------------[ cut here ]------------
WARNING: CPU: 1 PID: 1150 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 1 PID: 1150 Comm: kworker/1:1 Not tainted
4.14.0-rc2-42660-g24b7bd59eec0 #277
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
init_card sound/usb/caiaq/device.c:467
snd_probe+0x81c/0x1150 sound/usb/caiaq/device.c:525
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently bcd2000 driver submits a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 1 != type 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 1846 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 0 PID: 1846 Comm: kworker/0:2 Not tainted
4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
bcd2000_init_device sound/usb/bcd2000/bcd2000.c:289
bcd2000_init_midi sound/usb/bcd2000/bcd2000.c:345
bcd2000_probe+0xe64/0x19e0 sound/usb/bcd2000/bcd2000.c:406
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver may leave a stray URB for the mixer interrupt when it
exits by some error during probe. This leads to a use-after-free
error as spotted by syzkaller like:
==================================================================
BUG: KASAN: use-after-free in snd_usb_mixer_interrupt+0x604/0x6f0
Call Trace:
<IRQ>
__dump_stack lib/dump_stack.c:16
dump_stack+0x292/0x395 lib/dump_stack.c:52
print_address_description+0x78/0x280 mm/kasan/report.c:252
kasan_report_error mm/kasan/report.c:351
kasan_report+0x23d/0x350 mm/kasan/report.c:409
__asan_report_load8_noabort+0x19/0x20 mm/kasan/report.c:430
snd_usb_mixer_interrupt+0x604/0x6f0 sound/usb/mixer.c:2490
__usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779
....
Allocated by task 1484:
save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59
save_stack+0x43/0xd0 mm/kasan/kasan.c:447
set_track mm/kasan/kasan.c:459
kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551
kmem_cache_alloc_trace+0x11e/0x2d0 mm/slub.c:2772
kmalloc ./include/linux/slab.h:493
kzalloc ./include/linux/slab.h:666
snd_usb_create_mixer+0x145/0x1010 sound/usb/mixer.c:2540
create_standard_mixer_quirk+0x58/0x80 sound/usb/quirks.c:516
snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560
create_composite_quirk+0x1c4/0x3e0 sound/usb/quirks.c:59
snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560
usb_audio_probe+0x1040/0x2c10 sound/usb/card.c:618
....
Freed by task 1484:
save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59
save_stack+0x43/0xd0 mm/kasan/kasan.c:447
set_track mm/kasan/kasan.c:459
kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524
slab_free_hook mm/slub.c:1390
slab_free_freelist_hook mm/slub.c:1412
slab_free mm/slub.c:2988
kfree+0xf6/0x2f0 mm/slub.c:3919
snd_usb_mixer_free+0x11a/0x160 sound/usb/mixer.c:2244
snd_usb_mixer_dev_free+0x36/0x50 sound/usb/mixer.c:2250
__snd_device_free+0x1ff/0x380 sound/core/device.c:91
snd_device_free_all+0x8f/0xe0 sound/core/device.c:244
snd_card_do_free sound/core/init.c:461
release_card_device+0x47/0x170 sound/core/init.c:181
device_release+0x13f/0x210 drivers/base/core.c:814
....
Actually such a URB is killed properly at disconnection when the
device gets probed successfully, and what we need is to apply it for
the error-path, too.
In this patch, we apply snd_usb_mixer_disconnect() at releasing.
Also introduce a new flag, disconnected, to struct usb_mixer_interface
for not performing the disconnection procedure twice.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While line6_probe() may kick off URB for a control MIDI endpoint, the
function doesn't clean up it properly at its error path. This results
in a leftover URB action that is eventually triggered later and causes
an Oops like:
general protection fault: 0000 [#1] PREEMPT SMP KASAN
CPU: 1 PID: 0 Comm: swapper/1 Not tainted
RIP: 0010:usb_fill_bulk_urb ./include/linux/usb.h:1619
RIP: 0010:line6_start_listen+0x3fe/0x9e0 sound/usb/line6/driver.c:76
Call Trace:
<IRQ>
line6_data_received+0x1f7/0x470 sound/usb/line6/driver.c:326
__usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779
usb_hcd_giveback_urb+0x337/0x420 drivers/usb/core/hcd.c:1845
dummy_timer+0xba9/0x39f0 drivers/usb/gadget/udc/dummy_hcd.c:1965
call_timer_fn+0x2a2/0x940 kernel/time/timer.c:1281
....
Since the whole clean-up procedure is done in line6_disconnect()
callback, we can simply call it in the error path instead of
open-coding the whole again. It'll fix such an issue automagically.
The bug was spotted by syzkaller.
Fixes: eedd0e95d3 ("ALSA: line6: Don't forget to call driver's destructor at error path")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When podhd_init() failed with the acquiring a ctrl i/f, the line6
helper still calls the disconnect callback that eventually calls again
usb_driver_release_interface() with the NULL intf.
Put the proper NULL check before calling it for avoiding an Oops.
Fixes: fc90172ba2 ("ALSA: line6: Claim pod x3 usb data interface")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error path in podhd_init() tries to clear the pending timer, while
the timer object is initialized at the end of init sequence, thus it
may hit the uninitialized object, as spotted by syzkaller:
INFO: trying to register non-static key.
the code is fine but needs lockdep annotation.
turning off the locking correctness validator.
CPU: 1 PID: 1845 Comm: kworker/1:2 Not tainted
4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
__dump_stack lib/dump_stack.c:16
dump_stack+0x292/0x395 lib/dump_stack.c:52
register_lock_class+0x6c4/0x1a00 kernel/locking/lockdep.c:769
__lock_acquire+0x27e/0x4550 kernel/locking/lockdep.c:3385
lock_acquire+0x259/0x620 kernel/locking/lockdep.c:4002
del_timer_sync+0x12c/0x280 kernel/time/timer.c:1237
podhd_disconnect+0x8c/0x160 sound/usb/line6/podhd.c:299
line6_probe+0x844/0x1310 sound/usb/line6/driver.c:783
podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474
....
For addressing it, assure the initializations of timer and work by
moving them to the beginning of podhd_init().
Fixes: 790869dacc ("ALSA: line6: Add support for POD X3")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, lots of Plantronics, MS and Phoenix Audio devices need the
quirk not to read sample rate back, and the list just grows.
In this patch, instead of adding each device, apply the quirk by
matching with these vendors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like other Plantronics devices, P610 does not support sample
rate reading. Apply sample rate quirk to it.
BugLink: https://bugs.launchpad.net/bugs/1719853
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usx2y driver allocates the stream read/write buffers in continuous
pages depending on the stream setup, and this may spew the kernel
warning messages with a stack trace like:
WARNING: CPU: 1 PID: 1846 at mm/page_alloc.c:3883
__alloc_pages_slowpath+0x1ef2/0x2d70
Modules linked in:
CPU: 1 PID: 1846 Comm: kworker/1:2 Not tainted
....
It may confuse user as if it were any serious error, although this is
no fatal error and the driver handles the error case gracefully.
Since the driver has already some sanity check of the given size (128
and 256 pages), it can't pass any crazy value. So it's merely page
fragmentation.
This patch adds __GFP_NOWARN to each caller for suppressing such
kernel warnings. The original issue was spotted by syzkaller.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make these const as they are only passed to a const argument of the
function snd_pcm_set_ops in the file referencing them. Also, add const
to the declaration in the headers.
Structures found using Coccinelle and changes done by hand.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like other Plantronics devices, C310 and C520-M do not support sample
rate reading. Add them to the sample rate quirk accordingly.
BugLink: https://bugs.launchpad.net/bugs/1708499
BugLink: https://bugs.launchpad.net/bugs/1709282
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Line6 POD HD500X multi effect processor for playback
and capture (in/out audio) through USB.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usx2y driver has a debug printk code without proper KERN_ prefix.
On recent kernels, KERN_CONT prefix is mandatory for continued output
lines. Put it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver has a debug printk code without proper KERN_
prefix. On recent kernels, KERN_CONT prefix is mandatory for
continued output lines. Put it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add jump targets so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These headsets reports a lot of: cannot set freq 44100 to ep 0x81
and need a small delay between sample rate settings, just like
Zoom R16/24. Add both headsets to the Zoom R16/24 quirk for
a 1 ms delay between control msgs.
Signed-off-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/pcm.h> work with
const snd_pcm_ops. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for new Amanero Combo384 firmware version with a new
PID. This firmware uses DSD_U32_BE.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
C-Media devices (at least some models) mute the playback stream when
volumes are set to the minimum value. But this isn't informed via TLV
and the user-space, typically PulseAudio, gets confused as if it's
still played in a low volume.
This patch adds the new flag, min_mute, to struct usb_mixer_elem_info
for indicating that the mixer element is with the minimum-mute volume.
This flag is set for known C-Media devices in
snd_usb_mixer_fu_apply_quirk() in turn.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196669
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make this const as it is only used in a copy operation.
Done using Coccinelle.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Senheisser headset requires the typical sample-rate quirk for
avoiding spurious errors from inquiring the current sample rate like:
usb 1-1: 2:1: cannot get freq at ep 0x4
usb 1-1: 3:1: cannot get freq at ep 0x83
The USB ID 1395:740a has to be added to the entries in
snd_usb_get_sample_rate_quirk().
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1052580
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Omit extra messages for a memory allocation failure in this function.
This issue was detected by using the Coccinelle software.
[Fixed the unused variable warning by this change as well -- tiwai]
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove Initialization from POD HD500X because it's not needed.
Every time the device is connected dmesg gives the following output:
"receive length failed (error -11)".
To solve this problem, another flags is introduced
(LINE6_CAP_CONTROL_INFO) and it is only used for PODX3 in: sysfs
entries, call podhd_startup_finalize(pod) and disconnection.
With this patch the error disappear.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Line6 POD HD500X multi effect processor for
playback and capture (in/out audio) through USB.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another attempt to work around the VLA used in
mixer_us16x08.c. Basically the temporary array is used individually
for two cases, and we can declare locally in each block, instead of
hackish max() usage.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A mixer element created in a quirk for Tascam US-16x08 contains a
typo: it should be "EQ MidLow Q" instead of "EQ MidQLow Q".
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Cc: <stable@vger.kernel.org> # v4.11+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 89b593c30e ("ALSA: usb-audio: purge needless
variable length array"). The patch turned out to cause a severe
regression, triggering an Oops at snd_usb_ctl_msg(). It was overseen
that snd_usb_ctl_msg() writes back the response to the given buffer,
while the patch changed it to a read-only const buffer. (One should
always double-check when an extra pointer cast is present...)
As a simple fix, just revert the affected commit. It was merely a
cleanup. Although it brings VLA again, it's clearer as a fix. We'll
address the VLA later in another patch.
Fixes: 89b593c30e ("ALSA: usb-audio: purge needless variable length array")
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875
Cc: <stable@vger.kernel.org> # v4.11+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new structures as const as they are only passed an
argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Cross compiled these files:
sound/aoa/codecs/tas.c - powerpc
sound/mips/{hal2.c/sgio2audio.c} - mips
sound/ppc/{awacs.c/beep.c/tumbler.c} - powerpc
sound/soc/sh/siu_dai.c - sh
Could not find an architecture to compile sound/sh/aica.c.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This USB driver currently depends on X86 for some undocumented reason.
Since it compiles just fine for arm, we can at least enable compile
testing.
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the device-id table and a private flag to determine the device type
(US122 or US144) rather than spreading product-id conditionals
throughout the driver.
This USB driver currently depends on X86 (why?), but we should still add
the missing endianness conversions when accessing the USB
device-descriptor fields.
Compile-tested only.
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add missing endianness conversion when using the USB device-descriptor
bcdDevice field when applying the Amanero Combo384 (endianness!) quirk.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Cc: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The /proc/bus/usb devices don't exist anymore, since when we
got rid of usbfs. Those devices are now seen at
/dev/bus/usb.
Signed-off-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In kernel APIs of ALSA control interface, drivers can create a control
element set by a call of snd_ctl_new1() with a template. This template
is known to have const qualifier in general cases.
This commit adds the qualifier to template array, for safer program and
runtime. Application of this change moves the symbol from .data section
to .rodata section.
Cc: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently snd-usb-audio driver received a new option, quirk_alias, to
allow user to apply the existing quirk for a different device. This
works for many quirks as is, but some still need more tune-ups:
namely, some quirks check the USB vendor/device IDs in various places,
thus it doesn't work as long as the ID is different from the expected
one.
With this patch, the driver stores the aliased USB ID, so that these
rest quirks per device ID are applied. The transition to use the
cached USB ID was already done in the past, so what we needed now is
only to overwrite chip->usb_id.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
->fault(), ->page_mkwrite(), and ->pfn_mkwrite() calls do not need to
take a vma and vmf parameter when the vma already resides in vmf.
Remove the vma parameter to simplify things.
[arnd@arndb.de: fix ARM build]
Link: http://lkml.kernel.org/r/20170125223558.1451224-1-arnd@arndb.de
Link: http://lkml.kernel.org/r/148521301778.19116.10840599906674778980.stgit@djiang5-desk3.ch.intel.com
Signed-off-by: Dave Jiang <dave.jiang@intel.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Ross Zwisler <ross.zwisler@linux.intel.com>
Cc: Theodore Ts'o <tytso@mit.edu>
Cc: Darrick J. Wong <darrick.wong@oracle.com>
Cc: Matthew Wilcox <mawilcox@microsoft.com>
Cc: Dave Hansen <dave.hansen@intel.com>
Cc: Christoph Hellwig <hch@lst.de>
Cc: Jan Kara <jack@suse.com>
Cc: Dan Williams <dan.j.williams@intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
A few more cleanups and improvements that have been overlooked:
- Use ARRAY_SIZE() macro appropriately
- Code shuffling for minor optimization
- Omit superfluous variable initializations
- Get rid of superfluous NULL checks
- Add const to snd_us16x08_control_params definitions
No functional changes.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places leaking memory and doing double-free in
mixer_us16x08.c.
The driver allocates a usb_mixer_elem_info object at each
add_new_ctl() call. This has to be freed via kctl->private_free, but
currently this is done properly only for some controls.
Also, the driver allocates three external objects (comp_store,
eq_store, meter_store), and these are referred in elem->private_data
(it's not kctl->private_data). And these have to be released, but
there are none doing it. Moreover, these extra objects have to be
released only once. Thus the release should be done only by the first
kctl element that refers to it.
For fixing these, we call either snd_usb_mixer_elem_free() (only for
kctl->private_data) or elem_private_free() (for both
kctl->private_data and elem->private_data) via kctl->private_free
appropriately.
Last but not least, snd_us16x08_controls_create() may return in the
middle without releasing the allocated *_store objects due to an
error. For fixing this, we shuffle the allocation code so that it's
called just before its reference.
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Reported-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable length array is used in 'snd_us16x08_meter_get()', while there
is no need. It's better to purge it because variable length array has
overhead for stack handling.
This commit replaces the array with static length. Sparse generated below
warning.
sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used.
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new structures as const as they are only passed as
an argument to the function add_new_ctl. This agrument is of type const,
so snd_kcontrol_new structures having this property can be made const
too.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed inner a file, functions should have static qualifier for
local-linkage.
This commit fixes the bug. Sparse generated below warning.
sound/usb/mixer_us16x08.c:1043:32: warning: symbol 'snd_us16x08_create_meter_store' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed by one referrer inner a file, variables should have static
qualifier to declare local-linkage.
This commit fixes the bug. Sparse generated below warnings.
sound/usb/mixer_us16x08.c:156:13: warning: duplicate const
sound/usb/mixer_us16x08.c:156:18: warning: symbol 'route_names' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for Tascam US-16x08 usb interface.
Even that this is an usb compliant device,
the input channels and DSP functions (EQ/Compressor) aren't accessible
by default.
Signed-off-by: Detlef Urban <onkel@paraair.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While not all line6 devices currently support PCM, it causes no
harm to 'have it prepared'.
This also fixes toneport, which only has PCM - in which case
we previously skipped the USB transfer properties detection completely.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit f6a0dd107a.
The commit caused a regression on LINE6 Transport that has no control
caps. Although reverting the commit may result back in a spurious
error message for some device again, it's the simplest regression fix,
hence it's taken as is at first. The further code fix will follow
later.
Fixes: f6a0dd107a ("ALSA: line6: Only determine control port properties if needed")
Reported-by: Igor Zinovev <zinigor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for both little endian (DSD_U32_LE) and big endian
(DSD_U32_BE) version of the Amanero firmware.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 16200948d8.
The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.
As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.
Fixes: 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.
Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).
- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track
[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.
[Solution]
This patch adds DSD silence pattern to the endpoint settings.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the following devices.
- TEAC NT-503
- TEAC UD-503
- TEAC UD-501
(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
DSD mode, same as Denon/Marantz devices.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the
following warning.
[ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072),
cval->res is probably wrong.
[ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val =
4608/7680/1
Adding it to the list of devices in volume_control_quirks makes it work
properly, fixing related typo.
Signed-off-by: Con Kolivas <kernel@kolivas.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
since commit 57e6dae108 ("ALSA: usb-audio: do not trust too-big
wMaxPacketSize values"), the expected packetsize is always limited
to nominal + 25%. It was discovered, that some devices (Android audio
accessory) have a much higher jitter in used packetsizes than 25%
which would result in BABBLE condition and dropping of packets.
A better solution is so assume the jitter to be the nominal packetsize:
-one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape <apape@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a kernel crash report showing like:
Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000
[00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000
Internal error: Oops: 17 [#1] PREEMPT SMP ARM
CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1
task: a3ae61c0 ti: a08c8000 task.ti: a08c8000
PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio]
LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio]
pc : [<7f0eb22c>] lr : [<7f0e57fc>] psr: 200e0193
sp : a08c9c98 ip : a08c9ce8 fp : a08c9ce4
r10: 0000000a r9 : 00000102 r8 : 94cb3000
r7 : 94cb3000 r6 : 94d0f000 r5 : 94d0e8e8 r4 : 94d0e000
r3 : 7f0eb21c r2 : 00000000 r1 : 94cb3000 r0 : 00000000
Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user
Control: 10c5387d Table: 31d7c04a DAC: 00000015
Process dbus-daemon (pid: 250, stack limit = 0xa08c8238)
Stack: (0xa08c9c98 to 0xa08ca000)
...
Backtrace:
[<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio])
[<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4)
[<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0)
[<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148)
[<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380)
[<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc)
[<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8)
[<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8)
[<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78)
[<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100)
[<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8)
[<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c)
[<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0)
[<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20)
Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008)
Kernel panic - not syncing: Fatal exception in interrupt
There is a race between retire_capture_urb() and stop_endpoints().
The latter is called at stopping the stream and it sets some endpoint
fields to NULL. But its call is asynchronous, thus the pending
complete callback might get called after these NULL clears, and it
leads the NULL dereference like the above.
The fix is to move the NULL clearance after the synchronization,
i.e. wait_clear_urbs(). This is called at prepare and hw_free
callbacks, so it's assured to be called before the restart of the
stream or the release of the stream.
Also, while we're at it, put the EP_FLAG_RUNNING flag check at the
beginning of snd_complete_urb() to skip the pending complete after the
stream is stopped.
Fixes: b2eb950de2 ("ALSA: usb-audio: stop both data and sync...")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Reported-by: Mark Craske <Mark_Craske@mentor.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function returns -EINVAL even if it builds the stream properly.
The bogus error code sneaked in during the code refactoring, but it
wasn't noticed until now since the returned error code itself is
ignored in anyway. Kill it here, but there is no behavior change by
this patch, obviously.
Fixes: e5779998bf ('ALSA: usb-audio: refactor code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Userspace apps have to claim USB interfaces before using endpoints in
them (drivers/usb/core/devio.c:checkintf()). It's a lock mechanism so
that two "drivers" don't steal data from each other. Kernel drivers don't
have to claim interfaces to work - but they should, to lock out userspace.
While there, fix line6_properties struct to match checkpatch.pl.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver implements the deferred device disconnection for
the device in use. In this mode, the disconnection callback returns
immediately while the actual ALSA card object removal happens later
when all files get closed. As Shuah reported, this code flow,
however, leads to a use-after-free, detected by KASAN:
BUG: KASAN: use-after-free in snd_usb_audio_free+0x134/0x160 [snd_usb_audio] at addr ffff8801c863ce10
Write of size 8 by task pulseaudio/2244
Call Trace:
[<ffffffff81b31473>] dump_stack+0x67/0x94
[<ffffffff81564ef1>] kasan_object_err+0x21/0x70
[<ffffffff8156518a>] kasan_report_error+0x1fa/0x4e0
[<ffffffff81564ad7>] ? kasan_slab_free+0x87/0xb0
[<ffffffff81565733>] __asan_report_store8_noabort+0x43/0x50
[<ffffffffa0fc0f54>] ? snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0f54>] snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0fb1>] snd_usb_audio_dev_free+0x31/0x40 [snd_usb_audio]
[<ffffffff8243c78a>] __snd_device_free+0x12a/0x210
[<ffffffff8243d1f5>] snd_device_free_all+0x85/0xd0
[<ffffffff8242cae4>] release_card_device+0x34/0x130
[<ffffffff81ef1846>] device_release+0x76/0x1e0
[<ffffffff81b37ad7>] kobject_release+0x107/0x370
.....
Object at ffff8801c863cc80, in cache kmalloc-2048 size: 2048
Allocated:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff8156450d>] kasan_kmalloc+0xad/0xe0
[<ffffffff81560d1a>] kmem_cache_alloc_trace+0xfa/0x240
[<ffffffff8214ea47>] usb_alloc_dev+0x57/0xc90
[<ffffffff8216349d>] hub_event+0xf1d/0x35f0
....
Freed:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff81564ac1>] kasan_slab_free+0x71/0xb0
[<ffffffff81560929>] kfree+0xd9/0x280
[<ffffffff8214de6e>] usb_release_dev+0xde/0x110
[<ffffffff81ef1846>] device_release+0x76/0x1e0
....
It's the code trying to clear drvdata of the assigned usb_device where
the usb_device itself was already released in usb_release_dev() after
the disconnect callback.
This patch fixes it by checking whether the code path is via the
disconnect callback, i.e. chip->shutdown flag is set.
Fixes: 79289e2419 ('ALSA: usb-audio: Refer to chip->usb_id for quirks...')
Reported-and-tested-by: Shuah Khan <shuahkh@osg.samsung.com>
Cc: <stable@vger.kernel.org> # v4.6+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error checking here is messed up so we could end up dereferencing
-EFAULT.
Fixes: a16039cbf1 ('ALSA: line6: Add hwdep interface to access the POD control messages')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit c039aaa77a was incomplete,
missing part of the setup for Live. This makes also audio input work,
in addition to audio output.
Fixes: c039aaa77a
Reported-by: Eddi De Pieri <eddi@depieri.net>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, usb-line6 module exports an array of MIDI manufacturer ID and
usb-pod module uses it. However, the declaration is not the definition in
common header. The difference is explicit length of array. Although
compiler calculates it and everything goes well, it's better to use the
same representation between definition and declaration.
This commit fills the length of array for usb-line6 module. As a small
good sub-effect, this commit suppress below warnings from static analysis
by sparse v0.5.0.
sound/usb/line6/driver.c:274:43: error: cannot size expression
sound/usb/line6/driver.c:275:16: error: cannot size expression
sound/usb/line6/driver.c:276:16: error: cannot size expression
sound/usb/line6/driver.c:277:16: error: cannot size expression
Fixes: 705ececd1c ("Staging: add line6 usb driver")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DragonFly quirk added in 42e3121d90 ("ALSA: usb-audio: Add a more
accurate volume quirk for AudioQuest DragonFly") applies a custom dB map
on the volume control when its range is reported as 0..50 (0 .. 0.2dB).
However, there exists at least one other variant (hw v1.0c, as opposed
to the tested v1.2) which reports a different non-sensical volume range
(0..53) and the custom map is therefore not applied for that device.
This results in all of the volume change appearing close to 100% on
mixer UIs that utilize the dB TLV information.
Add a fallback case where no dB TLV is reported at all if the control
range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also
restrict the quirk to only apply to the volume control as there is also
a mute control which would match the check otherwise.
Fixes: 42e3121d90 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly")
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: David W <regulars@d-dub.org.uk>
Tested-by: David W <regulars@d-dub.org.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.
NULL check before some freeing functions is not needed.
Based on checkpatch warning
"kfree(NULL) is safe this check is probably not required"
and kfreeaddr.cocci by Julia Lawall.
Generated by: scripts/coccinelle/free/ifnullfree.cocci
CC: Andrej Krutak <dev@andree.sk>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We must do it this way, because e.g. POD X3 won't play any sound unless
the host listens on the bulk EP, so we cannot export it only via libusb.
The driver currently doesn't use the bulk EP messages in other way,
in future it could e.g. sense/modify volume(s).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only initialize PCM for POD HD devices that support it.
No POD HD seems to support MIDI, thus drop the initialization.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes audio in/out and basic initialization via control EP (emulates
what original driver does). The initialization is done similarly to original
POD, firmware and serial IDs are read and exported via sysfs.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all PODs use MIDI via USB data interface, thus allow avoiding
that code and instead using direct processing.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
POD X3 can initialize similarly to older PODs, but it doesn't have the MIDI
interface. Instead, configuration is done via proprietary bulk EP messages.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. POD X3 seems to require playback data to be sent to it to generate
capture data. Otherwise the device stalls and doesn't send any more capture
data until it's reset.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Splits max_packet_size to max_packet_size_in/out (e.g. for
different channel counts).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This has two parts:
* intervals_per_second setup
(high speed needs 8000, instead of 1000)
* iso_buffers setup (count of iso buffers depends on
USB speed, 2 is not enough for high speed)
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This basically changes LINE6_ISO_BUFFERS constant to a configurable
iso_buffers property.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on B850V3 CP2114 as it is unsupported and
causes noisy "current rate is different from the runtime rate" messages
when playback starts.
Signed-off-by: Ken Lin <ken.lin@advantech.com.tw>
Signed-off-by: Akshay Bhat <akshay.bhat@timesys.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc already print similar error once failing to alloc
enough memory, so let's remove this dump here.
Signed-off-by: Shawn Lin <shawn.lin@rock-chips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Users of devices affected by the Tenor feedback data error report
buffer underruns, even with the +/- 0x1.0000 quirk applied.
Compensating the error with 0xf000 instead seems to reliably fix
that issue.
See
https://sourceforge.net/p/alsa/mailman/message/35230259/
Reported-and-tested-by: Norman Nolte <norman.nolte@gmx.net>
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to
more that are based on the Tenor 8802TL chipset. Devices built by T+A
are affected too, and they apparently all use the same USB PID:PID.
Extend the quirky handling for that device as well, and rename the
quirks flag.
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
That's a quirk, after all, so move it where to all the other quirks
live.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 02fc76f6a changed base of the sysfs attributes from device to card.
The "show" callbacks dereferenced wrong objects because of this.
Fixes: 02fc76f6a7 ('ALSA: line6: Create sysfs via snd_card_add_dev_attr()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's an error, pcm is released in line6_pcm_acquire already.
Fixes: 247d95ee6d ('ALSA: line6: Handle error from line6_pcm_acquire()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc will print enough information in case of failure.
Signed-off-by: Wolfram Sang <wsa-dev@sang-engineering.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ELP HD USB Camera (05a3:9420) needs this quirk for suppressing
the unsupported sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VF0610 does not support reading the sample rate which leads to many
lines of "cannot get freq at ep 0x82". This patch adds the USB ID
(0x041E:4080) to snd_usb_get_sample_rate_quirk() list.
Signed-off-by: Piotr Karasinski <peter.karasinski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To allow for structure randomisation, replace the in order struct
initialisation style with explicit field style.
The Coccinelle semantic patch used to make this change is as follows:
@decl@
identifier i1,fld;
type T;
field list[n] fs;
@@
struct i1 {
fs
T fld;
...};
@@
identifier decl.i1,i2,decl.fld;
expression e;
position bad.p, bad.fix;
@@
struct i1 i2@p = { ...,
+ .fld = e
- e@fix
,...};
Signed-off-by: Amitoj Kaur Chawla <amitoj1606@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many USB audio devices with buggy firmware that don't react
with the sample rate reading properly. This often results in the
flood of error messages and slowing down the operation.
The sample rate read back is basically only for confirming the sample
rate setup, and it's not critically important. As a compromise, in
this patch, we stop the sample rate read back once when the device
gives errors more than tolerance (twice, as of now). This should
improve most of error cases while we still can catch the firmware
bugginess.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
we've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy.
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Merge tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"We've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy"
* tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix inconsistent monitor_present state until repoll
ALSA: hda - Fix regression of monitor_present flag in eld proc file
ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
ALSA: sscape: Use correct format identifier for size_t
ALSA: usb-audio: Add a quirk for Plantronics BT300
ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
ALSA: hda - Bind with i915 only when Intel graphics is present
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.
[Some background: the patch essentially skips the controls of a couple
of FU volumes. Although the firmware exposes the dB and the value
information via the usb descriptor, changing the values (we set the
min volume as default) screws up the device. Although this has been
fixed in the newer firmware, the devices are shipped with the old
firmware, thus we need the workaround in the driver side. -- tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
miniDSP USBStreamer UAC2 devices send clock validity changes with the
control field set to zero. The current interrupt handler ignores all
packets if the control field does not match the mixer element's, but
it really should only do that in case that field is needed to
distinguish multiple elements with the same ID.
This patch implements a logic that lets notifications packets pass
if the element ID is unique for a given device.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 specifies clock sources that optionally have validity controls.
This patch exposes them as mixer controls, so they can be read (and
at least in theory even be written) by userspace applications in order
to make clock selection policy decisions.
This implementation does nothing if the device is not UAC2 compliant,
or if the clock source does not define said validity control bits.
Tested with a miniDSP USBStreamer (0x2752/0x0016).
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:
"Some bug fixes on au0828 and snd-usb-audio:
- the au0828+snd-usb-audio MC patch broke several things and produced
some race conditions. Better to revert the patches, and re-work on
them for a next version
- fix a regression at tuner disable links logic
- properly handle dev_state as a bitmask"
* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] Revert "[media] media: au0828 change to use Managed Media Controller API"
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
[media] au0828: Fix dev_state handling
[media] au0828: fix au0828_v4l2_close() dev_state race condition
[media] media: au0828 fix to clear enable/disable/change source handlers
[media] v4l2-mc: cleanup a warning
[media] au0828: disable tuner links and cache tuner/decoder
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
usb 2-1.3: 2:1: cannot get freq at ep 0x85
usb 2-1.3: 1:1: cannot get freq at ep 0x2
....
Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes:
- A fix in ALSA timer core to avoid possible BUG() trigger
- A fix in ALSA timer core 32bit compat layer
- A few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- Fixes of USB-audio double-free at some error paths
- A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- a fix in ALSA timer core to avoid possible BUG() trigger
- a fix in ALSA timer core 32bit compat layer
- a few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- fixes of USB-audio double-free at some error paths
- a fix for memory leak in DICE driver at hotunplug"
* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: timer: Use mod_timer() for rearming the system timer
ALSA: hda - fix front mic problem for a HP desktop
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
ALSA: dice: fix memory leak when unplugging
ALSA: hda - Apply fix for white noise on Asus N550JV, too
ALSA: hda - Fix white noise on Asus N750JV headphone
ALSA: hda - Asus N750JV external subwoofer fixup
ALSA: timer: fix gparams ioctl compatibility for different architectures
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
https://bugzilla.kernel.org/show_bug.cgi?id=115561
It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.
So, better to revert it and fix the core before reapplying this
change.
This reverts commit aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed.
So this request came up: a collection of small fixes to paper over
the inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too.
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Merge tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed. So
this request came up: a collection of small fixes to paper over the
inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too"
* tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix missing ELD update at unplugging
ALSA: usb-audio: add Microsoft HD-5001 to quirks
ALSA: hda - Workaround for unbalanced i915 power refcount by concurrent probe
ALSA: hda - Fix spurious kernel WARNING on Baytrail HDMI
ALSA: hda - Fix forgotten HDMI monitor_present update
ALSA: hda - Really restrict i915 notifier to HSW+
The Microsoft HD-5001 webcam microphone does not support sample rate
reading as the HD-5000 one.
This results in dmesg errors and sound hanging with pulseaudio.
Signed-off-by: Victor Clément <victor.clement@openmailbox.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514
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Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"
* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
ALSA: mixart: silence an uninitialized variable warning
ALSA: usb-audio: Add sanity checks for endpoint accesses
ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
ALSA: hda - Limit i915 HDMI binding only for HSW and later
ALSA: hda - Fix unconditional GPIO toggle via automute
ALSA: mixart: silence unitialized variable warnings
ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
ASoC: rsnd: add simplified module explanation
ASoC: hdac_hdmi: Add broxton device ID
ASoC: Intel: Bxtn: Add Broxton PCI ID
ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
ASoC: Intel: add dmabuffer to common sst_dsp
ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
ASoC: Intel: Skylake: Fix whitepsace issues
ASoC: Intel: Skylake: Move module id defines
...
Add some sanity check codes before actually accessing the endpoint via
get_endpoint() in order to avoid the invalid access through a
malformed USB descriptor. Mostly just checking bNumEndpoints, but in
one place (snd_microii_spdif_default_get()), the validity of iface and
altsetting index is checked as well.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create_fixed_stream_quirk() may cause a NULL-pointer dereference by
accessing the non-existing endpoint when a USB device with a malformed
USB descriptor is used.
This patch avoids it simply by adding a sanity check of bNumEndpoints
before the accesses.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* commit '840f5b0572ea': (381 commits)
media: au0828 disable tuner to demod link in au0828_media_device_register()
[media] touptek: cast char types on %x printk
[media] touptek: don't DMA at the stack
[media] mceusb: use %*ph for small buffer dumps
[media] v4l: exynos4-is: Drop unneeded check when setting up fimc-lite links
[media] v4l: vsp1: Check if an entity is a subdev with the right function
[media] hide unused functions for !MEDIA_CONTROLLER
[media] em28xx: fix Terratec Grabby AC97 codec detection
[media] media: add prefixes to interface types
[media] media: rc: nuvoton: switch attribute wakeup_data to text
[media] v4l2-ioctl: fix YUV422P pixel format description
[media] media: fix null pointer dereference in v4l_vb2q_enable_media_source()
[media] v4l2-mc.h: fix yet more compiler errors
[media] staging/media: add missing TODO files
[media] media.h: always start with 1 for the audio entities
[media] sound/usb: Use meaninful names for goto labels
[media] v4l2-mc.h: fix compiler warnings
[media] media: au0828 audio mixer isn't connected to decoder
[media] sound/usb: Use Media Controller API to share media resources
[media] dw2102: add support for TeVii S662
...
Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. Media Controller
specific initialization is done after sound card is registered. ALSA
creates Media interface and entity function graph nodes for Control,
Mixer, PCM Playback, and PCM Capture devices.
snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is
granted, it will release it from snd_usb_hw_free(). If resource is
busy, -EBUSY is returned.
Media specific cleanup is done in usb_audio_disconnect().
Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
Plantronics DA45 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x4" and "cannot get freq at
ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'umidi' object will be free'd on the error path by snd_usbmidi_free()
when tearing down the rawmidi interface. So we shouldn't try to free it
in snd_usbmidi_create() after having registered the rawmidi interface.
Found by KASAN.
Signed-off-by: Andrey Konovalov <andreyknvl@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam HD-6000 (045e:076f) requires the similar quirk for
avoiding the stall due to the invalid sample rate reads.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=111491
Signed-off-by: Lev Lybin <lev.lybin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got
through. This patch fixes the vendor ID and aligns the comment.
Fixes: a4eae3a506 ('ALSA: usb: Add native DSD support for Oppo HA-1')
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new option "quirk_alias" to snd-usb-audio driver for
allowing user to pass the quirk alias list. A quirk alias consists of
a string form like 0123abcd:5678beef, which makes to apply a quirk to
a device with USB ID 0123:abcd treated as if it were 5678:beef.
This feature is useful to test an existing quirk, typically for a
newer model of the same vendor, without patching / rebuilding the
kernel driver.
The current implementation is fairly simplistic: since there is no API
for matching a usb_device_id to the given ID pair, it has an open code
to loop over the id table and matches only with vendor:product pair.
So far, this is OK, as all existing entries are with vendor:product
pairs, indeed. Once when we have another matching entry, however,
we'd need to update get_alias_quirk() as well.
Note that this option is provided only for testing / development. If
you want to have a proper support, contact to upstream for adding the
matching quirk in the driver code statically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the later change to allow a better
quirk ID management. In the current USB-audio code, there are a few
places looking at usb_device idVendor and idProduct fields directly
even though we have already a static member in snd_usb_audio.usb_id.
This patch modifies such codes to refer to the latter field.
For achieving this, two slightly intensive changes have been done:
- The snd_usb_audio object is set/reset via dev_getdrv() for the given
USB device; it's needed for minimizing the changes for some existing
quirks that take only usb_device object.
- __snd_usbmidi_create() is introduced to receive the pre-given usb_id
argument. The exported snd_usbmidi_create() is unchanged by calling
this new function internally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TEAC UD-501/UD-503/NT-503 fail to switch properly between different
rate/format. Similar to 'Playback Design', this patch corrects the
invalid clock source error for TEAC products and avoids complete
freeze of the usb interface of 503 series.
Signed-off-by: Guillaume Fougnies <guillaume@eulerian.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree in
anyway sooner or later.]
Core
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc
- A few code cleanups of fm801 driver
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Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[ NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree
in anyway sooner or later. ]
Core:
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC:
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio:
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire:
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio:
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc:
- A few code cleanups of fm801 driver"
* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
ALSA: timer: Code cleanup
ALSA: timer: Harden slave timer list handling
ALSA: hda - Add fixup for Dell Latitidue E6540
ALSA: timer: Fix race among timer ioctls
ALSA: hda - add codec support for Kabylake display audio codec
ALSA: timer: Fix double unlink of active_list
ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
ALSA: hda - fix the headset mic detection problem for a Dell laptop
ALSA: hda - Fix white noise on Dell Latitude E5550
ALSA: hda_intel: add card number to irq description
ALSA: seq: Fix race at timer setup and close
ALSA: seq: Fix missing NULL check at remove_events ioctl
ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
ASoC: AMD: Add missing include file
ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
ALSA: usb: Add native DSD support for Oppo HA-1
ASoC: Make aux_dev more like a generic component
ASoC: bcm2835: cleanup includes by ordering them alphabetically
ASoC: AMD: Manage ACP 2.x SRAM banks power
...
Pull trivial tree updates from Jiri Kosina.
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial:
floppy: make local variable non-static
exynos: fixes an incorrect header guard
dt-bindings: fixes some incorrect header guards
cpufreq-dt: correct dead link in documentation
cpufreq: ARM big LITTLE: correct dead link in documentation
treewide: Fix typos in printk
Documentation: filesystem: Fix typo in fs/eventfd.c
fs/super.c: use && instead of & for warn_on condition
Documentation: fix sysfs-ptp
lib: scatterlist: fix Kconfig description
The commit [da6d276957: ALSA: usb-audio: Add resume support for
Native Instruments controls] brought a regression where the Native
Instrument audio devices don't get the correct value at update due to
the missing shift at writing. This patch addresses it.
Fixes: da6d276957 ('ALSA: usb-audio: Add resume support for Native Instruments controls')
Reported-and-tested-by: Owen Williams <owilliams@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM may still have a leftover instance after disconnection and
it delays its release. The problem is that the PCM close code path of
USB-audio driver has a call of snd_usb_autosuspend(). This involves
with the call of usb_autopm_put_interface() and it may lead to a
kernel Oops due to the NULL object like:
BUG: unable to handle kernel NULL pointer dereference at 0000000000000190
IP: [<ffffffff815ae7ef>] usb_autopm_put_interface+0xf/0x30 PGD 0
Call Trace:
[<ffffffff8173bd94>] snd_usb_autosuspend+0x14/0x20
[<ffffffff817461bc>] snd_usb_pcm_close.isra.14+0x5c/0x90
[<ffffffff8174621f>] snd_usb_playback_close+0xf/0x20
[<ffffffff816ef58a>] snd_pcm_release_substream.part.36+0x3a/0x90
[<ffffffff816ef6b3>] snd_pcm_release+0xa3/0xb0
[<ffffffff816debb0>] snd_disconnect_release+0xd0/0xe0
[<ffffffff8114d417>] __fput+0x97/0x1d0
[<ffffffff8114d589>] ____fput+0x9/0x10
[<ffffffff8109e452>] task_work_run+0x72/0x90
[<ffffffff81088510>] do_exit+0x280/0xa80
[<ffffffff8108996a>] do_group_exit+0x3a/0xa0
[<ffffffff8109261f>] get_signal+0x1df/0x540
[<ffffffff81040903>] do_signal+0x23/0x620
[<ffffffff8114c128>] ? do_readv_writev+0x128/0x200
[<ffffffff810012e1>] prepare_exit_to_usermode+0x91/0xd0
[<ffffffff810013ba>] syscall_return_slowpath+0x9a/0x120
[<ffffffff817587cd>] ? __sys_recvmsg+0x5d/0x70
[<ffffffff810d2765>] ? ktime_get_ts64+0x45/0xe0
[<ffffffff8115dea0>] ? SyS_poll+0x60/0xf0
[<ffffffff818d2327>] int_ret_from_sys_call+0x25/0x8f
We have already a check of disconnection in snd_usb_autoresume(), but
the check is missing its counterpart. The fix is just to put the same
check in snd_usb_autosuspend(), too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset
but they use their own vendor ID.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.
Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.
However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.
Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).
Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.
v2: incorporated Takashi Iwai's suggestion for the quirk application
method
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb_protocol_ops structures are never modified, so declare them as
const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fix multiple spelling typos found in
various part of kernel.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The snd_rawmidi_global_ops structures are never modified, so declare them
as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 939f325f4a ("usb: add usb_endpoint_maxp() macro") and commit
29cc88979a ("USB: use usb_endpoint_maxp() instead of le16_to_cpu()")
introduced a new helper macro. This trivial patch convert remaining
users found in ua101 driver.
Signed-off-by: Cheah Kok Cheong <thrust73@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the many faults of the QinHeng CH345 USB MIDI interface chip is
that it does not handle received SysEx messages correctly -- every second
event packet has a wrong code index number, which is the one from the last
seen message, instead of 4. For example, the two messages "FE F0 01 02 03
04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event
packets:
correct: CH345:
0F FE 00 00 0F FE 00 00
04 F0 01 02 04 F0 01 02
04 03 04 05 0F 03 04 05
04 06 07 08 04 06 07 08
04 09 0A 0B 0F 09 0A 0B
04 0C 0D 0E 04 0C 0D 0E
05 F7 00 00 05 F7 00 00
A class-compliant driver must interpret an event packet with CIN 15 as
having a single data byte, so the other two bytes would be ignored. The
message received by the host would then be missing two bytes out of six;
in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7".
These corrupted SysEx event packages contain only data bytes, while the
CH345 uses event packets with a correct CIN value only for messages with
a status byte, so it is possible to distinguish between these two cases by
checking for the presence of this status byte.
(Other bugs in the CH345's input handling, such as the corruption resulting
from running status, cannot be worked around.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CH345 USB MIDI chip has two output ports. However, they are
multiplexed through one pin, and the number of ports cannot be reduced
even for hardware that implements only one connector, so for those
devices, data sent to either port ends up on the same hardware output.
This becomes a problem when both ports are used at the same time, as
longer MIDI commands (such as SysEx messages) are likely to be
interrupted by messages from the other port, and thus to get lost.
It would not be possible for the driver to detect how many ports the
device actually has, except that in practice, _all_ devices built with
the CH345 have only one port. So we can just ignore the device's
descriptors, and hardcode one output port.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.
This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.
It would benefit from some regresison testing with other devices if
possible.
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to verify that "value" is either zero or one, so we test if it
is greater than one. Unfortunately, this is a signed int so it could
also be negative. I think this is harmless but it introduces a static
checker warning. Let's make "value" unsigned.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of cval->cached should be zero-based (including master channel).
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In theory, the device may get suspended even at runtime PM suspend.
Currently we don't save the mixer state for autopm, and it may bring
inconsistency.
This patch removes the special handling for autosuspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent fix of runtime PM for USB-audio driver, we got a
lockdep warning like:
=============================================
[ INFO: possible recursive locking detected ]
4.2.0-rc8+ #61 Not tainted
---------------------------------------------
pulseaudio/980 is trying to acquire lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
but task is already holding lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
This comes from snd_usb_autoresume() invoking down_read() and it's
used in a nested way. Although it's basically safe, per se (as these
are read locks), it's better to reduce such spurious warnings.
The read lock is needed to guarantee the execution of "shutdown"
(cleanup at disconnection) task after all concurrent tasks are
finished. This can be implemented in another better way.
Also, the current check of chip->in_pm isn't good enough for
protecting the racy execution of multiple auto-resumes.
This patch rewrites the logic of snd_usb_autoresume() & co; namely,
- The recursive call of autopm is avoided by the new refcount,
chip->active. The chip->in_pm flag is removed accordingly.
- Instead of rwsem, another refcount, chip->usage_count, is introduced
for tracking the period to delay the shutdown procedure. At
the last clear of this refcount, wake_up() to the shutdown waiter is
called.
- The shutdown flag is replaced with shutdown atomic count; this is
for reducing the lock.
- Two new helpers are introduced to simplify the management of these
refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
the shutdown state, and does autoresume. snd_usb_unlock_shutdown()
does the opposite. Most of mixer and other codes just need this,
and simply returns an error if it receives an error from lock.
Fixes: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Gustard DAC-X20U.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input terminal parser recurses into the referenced clock entity to verify
it is existant and thus the terminal descriptor is valid. The actual property
values of the term instance which is initially parsed must not be overriden by
the recursion. For this to work the term properties have to be assigned after
recursing into the referenced clock entity descriptors.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix for deadlock in PM in commit [1ee23fe07e: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio Class version 2.0 supports three different parameter block sizes for
CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes
(5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter
Block). Use the correct size according to the specific control as it was
already done for UACv1. The allocated block size for control requests is
increased to support the 4 byte worst case.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2.
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a monitor stream is active, the next PCM stream access results in
EBUSY error because of the check in line6_stream_start(). Fix this by
just skipping the submission of pending URBs when the stream is
already running instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431
Cc: <stable@vger.kernel.org> # v4.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Steinberg MI2 and MI4 interfaces are compatible with the USB class
audio spec, but the MIDI part of the devices is reported as a vendor
specific interface.
This patch adds entries to quirks-table.h to recognize the MIDI
endpoints. Audio functionality was already working and is unaffected by
this change.
Signed-off-by: Dominic Sacré <dominic.sacre@gmx.de>
Signed-off-by: Albert Huitsing <albert@huitsing.nl>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver worked around an error in the MAYA44 USB(+)'s mixer unit
descriptor by aborting before parsing the missing field. However,
aborting parsing too early prevented parsing of the other units
connected to this unit, so the capture mixer controls would be missing.
Fix this by moving the check for this descriptor error after the parsing
of the unit's input pins.
Reported-by: nightmixes <nightmixes@gmail.com>
Tested-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer control names for the ESI Maya44 USB+ (which appears to be
identical width the AudioTrak Maya44 USB).
Reported-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk allows us to avoid the noisy:
current rate 0 is different from the runtime rate
message every time playback starts. While USB DAC in the RR2150
supports reading the sample rate, it never returns a sample rate
other than zero in my observation with common sample rates.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this from the logs:
usb 7-1: New USB device found, idVendor=046d, idProduct=08ca
...
usb 7-1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong.
usb 7-1: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by sparse:
sound/usb/bcd2000/bcd2000.c:73:1: warning: symbol 'devices_used' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam Studio (045e:0772) needs a similar quirk for
suppressing the wrong sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Roland SC-D70 reports its device class as vendor specific class and
the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output.
In the quirks table the sampling rate was hard-coded to 44100 Hz
and therefore not worked when the sound module was in 48000 Hz mode.
In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE
but as the sound module reports incorrect bSubframeSize in its
descriptors, additional change is made in format.c to detect it and
to override it (which uses the existing code for Edirol SD-90).
Tested both when the sound module was in 44100 Hz mode and 48000 Hz
mode and both audio input and output. MIDI related part of the driver
is not touched.
Signed-off-by: Takamichi Horikawa <takamichiho@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate.
Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed.
Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961.
Signed-off-by: Adam Honse <calcprogrammer1@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3237"
Signed-off-by: Dmitry M. Fedin <dmitry.fedin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this quirk allows us to avoid the noisy
"cannot get freq at ep 0x1" message in dmesg output every time
playback starts.
This ought to affect other Benchmark DAC1 variations using the same
"Microchip Technology, Inc." chip as well, but I have only tested
with the "Pre" variant.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device complies to the UAC1 standard but hides that fact with
proprietary descriptors. The autodetect quirk for Roland devices
catches the audio interface but misses the MIDI part, so a specific
quirk is needed.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-by: Rafa Lafuente <rafalafuente@gmail.com>
Tested-by: Raphaël Doursenaud <raphael@doursenaud.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usages of clamp() macro in sound/usb/line6/playback.c are just
wrong, the low and high values are swapped.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix problem where playback of Denon DA-300USB DAC sometimes does not
start and leads to error messages like "clock source 41 is not valid,
cannot use".
Solution: Treat this device the same as other Denon/Marantz devices in
sound/usb/quirks.c.
Tested with both PCM and DSD formats.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261
Signed-off-by: Frank C Guenther <bugzilla.frnkcg@spamgourmet.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate.
This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound.
[minor tidy up by tiwai]
Signed-off-by: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The address cannot be negative so make it unsigned. Also, an unsigned
int is always sufficient for the length, so no need to overdo it with a
size_t. Finally, add in range checks to see if the values passed in
actually fit where they are used.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmware version is a single byte so have the variable type agree.
Since the address to this member is passed to the read function, using
an int is not even portable.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put an upper bound on how long we will wait for the device to respond to
a read/write request (i.e., 100 milliseconds) and return an error if
this is reached.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device indicates the result of a read/write operation by making the
status available on a subsequent request from the driver. This is not
ready immediately, though, so the driver is currently slamming the
device with hundreds of pointless requests before getting the expected
response. Add a two millisecond delay before each attempt. This is
approximately the behavior observed with version 4.2.7.1 of the Windows
driver.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The first URBs are submitted during the prepare stage. When .trigger is
called, the ALSA core saves a trigger tstamp that doesn't correspond to
the actual time when the samples are submitted. The trigger_tstamp is
now updated when the first data are submitted to avoid any time offsets.
A usb-specific trigger_tstamp_pending_update flag is used for now,
at some point the flag would need to move to the ALSA core, USB
is not the only interface where silent block transfers are programmed
as part of the prepare stage, with actual data enabled when .trigger
is called.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a unique name for each driver instead of using "line6usb" for
all of them. This will allow for different configurations based on the
driver type.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is unlikely this function would ever be used in a context without a
pointer to a `struct usb_line6_toneport', so grab the device type from
it rather than having the caller do it.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a predicate for testing if the device supports source selection to
make the conditional logic around this a bit cleaner.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_start_timer passes an unsigned int as argument to be used in mod_timer
which is then used by mod_timer as unsigned long, this just fixes up the
argument type. This change helps make static code checkers happy.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is only an API consolidation and should make things more readable
it replaces var * HZ / 1000 by msecs_to_jiffies(var).
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use of this function ended with commits 3e58c868db ("staging: line6:
drop midi_mask_receive") and af89d2897a ("staging: line6: drop
midi_mask_transmit".)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function has not been used since merging the driver into the kernel
(and a good while before that.)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of them are rather relevant with the definitions in driver.h,
and there are only a few lines, so just rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just reformatting the comments and typos fixed, no functional
changes. Particularly,
- avoid the kerneldoc marker "/**",
- reduce multiple comment lines into single lines,
- corrected wrongly referred function names
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-space API definition for usb_stream stuff should be moved
to include/uapi/sound to be exposed publicly.
While we're at it, add the missing ifdef guard for double inclusion,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both playback and capture callbacks are identical, so let's merge
them.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code deals with the stream start / stop solely via
line6_pcm_acquire() and line6_pcm_release(). This was (supposedly)
intended to avoid the races, but it doesn't work as expected. The
concurrent acquire and release calls can be performed without proper
protections, thus this might result in memory corruption.
Furthermore, we can't take a mutex to protect the whole function
because it can be called from the PCM trigger callback that is an
atomic context. Also spinlock isn't appropriate because the function
allocates with kmalloc with GFP_KERNEL. That is, these function just
lead to singular problems.
This is an attempt to reduce the existing races. First off, separate
both the stream buffer management and the stream URB management. The
former is protected via a newly introduced state_mutex while the
latter is protected via each line6_pcm_stream lock.
Secondly, the stream state are now managed in opened and running bit
flags of each line6_pcm_stream. Not only this a bit clearer than
previous combined bit flags, this also gives a better abstraction.
These rewrites allows us to make common hw_params and hw_free
callbacks for both playback and capture directions.
For the monitor and impulse operations, still line6_pcm_acquire() and
line6_pcm_release() are used. They call internally the corresponding
functions for both playback and capture streams with proper lock or
mutex. Unlike the previous versions, these function don't take the
bit masks but the only single type value. Also they are supposed to
be applied only as duplex operations.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing prev_fsize in line6_pcm_acquire() is pretty racy.
This can be called at any time while the stream is being played.
Rather better to clear prev_fbuf and prev_fsize at the proper place
like the stream stop for capture, and just after copying the monitor /
impulse data inside the spinlock.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The impulse and monitor handling in submit_audio_out_urb() isn't
protected thus this can be racy with the capture stream handling.
This patch extends the range to protect via each stream's spinlock
(now the whole submit_audio_*_urb() are covered), and take the capture
stream lock additionally for the impulse and monitor handling part.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check of multi configurations before snd_card_new() as a
short path, and reduce superfluous pointer references.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the private data individually in each driver's
probe at first, let snd_card_new() allocate the data that is called in
line6_probe(). This simplifies the primary probe functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface argument is used just for retrieving the assigned
device, which can be already found in line6->ifcdev. Drop them from
the callbacks. Also, pass the usb id to private_init so that the
driver can deal with it there. This is a preliminary work for the
further cleanup to move the whole allocation into driver.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A minor optimization; while pausing, the driver just copies the zero
that doesn't need any volume changes.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM stream buffer allocation and free are identical for both
playback and capture streams. Provide single helper functions.
These are used only in pcm.c, thus they can be even static.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to unlink and sync URBs are identical for both playback and
capture streams. Consolidate to single helper functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new line6_pcm_stream structure and group individual
fields of snd_line6_pcm struct to playback and capture groups.
This patch itself just does rename and nothing else. More
meaningful cleanups based on these fields shuffling will follow.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the problem still really remains, we should fix it instead of
papering over it like this...
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a decremental loop without particular reasons worsens the
readability a lot. Use incremental loops instead.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The trigger callback is already spinlocked, so we need no more lock
here (even for the linked substreams). Let's drop it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_pcm_acquire() tries to restore the newly obtained resources at
the error path. But some flags aren't recorded and released properly
when the corresponding buffer is already present. These bits have to
be cleared in the error recovery, too.
Also, "flags_final" can be initialized to zero since we pass only the
subset of "channels" bits.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a few places using bits OR wrongly for condition checks.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The midi_transmit_lock is used always inside the send_urb_lock, thus
it doesn't play any role. Let's kill it. Also, rename
"send_urb_lock" as a more simple name "lock" since this is the only
lock for midi.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function isn't used any longer after rewriting from sysfs to leds
class in toneport.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix memory leak at probe error path by rearranging the call order in
line6_destruct() so that the common destructor is always called.
Also this simplifies the error path to a single goto label.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of non-standard sysfs, reimplement the LED controls on
TonePort as LED class devices.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently disconnect callback is used as a driver's destructor, and
this has to be called not only at the disconnection time but also at
the error paths during probe.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that timer_del_sync() in the destructor can be called safely at
any time. Also move the mod_timer() call in toneport_setup(), which
is a bit clearer place.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface and driver objects are always set when callbacks are
called. Drop such superfluous NULL checks in init and disconnect
calls of each driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's utterly unsafe to proceed further the disconnect procedure if the
assigned usbdev is inconsistent with the expected object. Better to
put a WARN_ON() for more cautions and abort immediately.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver version string was removed in an ealier commit for being
useless. These are equally useless.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct spelling includes the space. Fix this in strings and
comments that refer to the manufacturer.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call line6_pcm_disconnect() at disconnect to make sure that all URBs
are cleared. Also reduce the superfluous snd_pcm_stop() calls from
the function (and remove the unused function) since the streams are
guaranteed to be stopped at this point via snd_card_disconnect().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Calling line6_pcm_disconnect() at suspend callback is superfluous and
rather confusing. Let's get rid of it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Such a debug is needed in the core code, not in each lowlevel driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is rather useless for a driver that has been already merged into
the official tree.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fairly big rewrite regarding the card resource management in
line6 drivers:
- The card creation is moved into line6_probe(). This adds the global
destructor to private_free, so that each driver doesn't have to call
it any longer.
- The USB disconnect callback handles the card release, thus each
driver needs to concentrate on only its own resources. No need to
snd_card_*() call in the destructor.
- Fix the potential stall in disconnection by removing
snd_card_free(). It's replaced with snd_card_free_when_closed()
for asynchronous release.
- The only remaining operation for the card in each driver is the call
of snd_card_register(). All the rest are dealt in the common module
by itself.
- These ended up with removal of audio.[ch] as a result of a reduction
of one layer. Each driver just needs to call line6_probe().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM trigger callback is guaranteed to be called already in
spinlock / irq-disabled context.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The line6 drivers don't support the full resume although they set
SNDRV_PCM_INFO_RESUME. These flags have to be dropped to inform
properly to the user-space.
Also, drop the CONFIG_PM in trigger callbacks, too, which are rather
superfluous.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous fix for PCM, attach the card-specific resource into
rawmidi->private_data instead of handling in a snd_device object.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of handling the card-specific resource in snd_device, attach
it into pcm->private_data and release it directly in private_free.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of sysfs and the conditional build with Kconfig, implement the
handling of the impulse response controls via control API, and always
enable the build. Two new controls, "Impulse Response Volume" and
"Impulse Response Period" are added as a replacement for the former
sysfs files.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split to each individual driver for POD, PODHD, TonePort and Variax
with a core LINE6 helper module. The new modules follow the standard
ALSA naming rule with snd prefix: snd-usb-pod, snd-usb-podhd,
snd-usb-toneport and snd-usb-variax, together with the corresponding
CONFIG_SND_USB_* Kconfig items.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Promote line6 driver from staging to sound/usb/line6 directory, and
maintain through sound subsystem tree.
This commit just moves the code and adapts Makefile / Kconfig.
The further renames and misc cleanups will follow.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Akai MPC Element incorrectly reports its bInterfaceClass as 255, but
otherwise implements the USB MIDI spec correctly.
This adds a quirks-table.h entry which allows the device to be
recognized as a standard USB MIDI device.
Signed-off-by: Paul Bonser <misterpib@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 897c329bc ("ALSA: usb: caiaq: check for cdev->n_streams > 1")
introduced a safety check to protect against bogus data provided by
devices. However, the n_streams variable is already divided by
CHANNELS_PER_STREAM, so the correct check is 'n_streams > 0'.
Fix this to un-break support for stereo devices.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Cc: stable@kernel.org [v3.18+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here are a few fixes that have landed after the previous pull
request. All are driver specific fixes including:
- error/int value fixes in OXFW,
- Intel Skylake HD-audio HDMI codec support,
- Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
- a few more DSD support and a quirk for Arcam rPAC in usb-audio,
- a typo fix for Scarlett 6i6,
- fixes for new ASIHPI firmware,
- ASoC Exynos7 cleanups,
- Intel ACPI support, and
- a fix for PCM512 register cache sync.
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Merge tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few fixes that have landed after the previous pull request.
All are driver specific fixes including:
- error/int value fixes in OXFW,
- Intel Skylake HD-audio HDMI codec support,
- Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
- a few more DSD support and a quirk for Arcam rPAC in usb-audio,
- a typo fix for Scarlett 6i6,
- fixes for new ASIHPI firmware,
- ASoC Exynos7 cleanups,
- Intel ACPI support, and
- a fix for PCM512 register cache sync"
* tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
ALSA: hda/realtek - New codec support for ALC298
ALSA: asihpi: update to HPI version 4.14
ALSA: asihpi: increase tuner pad cache size
ALSA: asihpi: relax firmware version check
ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
ALSA: hda - Add quirk for Packard Bell EasyNote MX65
ALSA: usb-audio: add native DSD support for Matrix Audio DACs
ALSA: hda/realtek - New codec support for ALC256
ALSA: hda/realtek - Add new Dell desktop for ALC3234 headset mode
ASoC: Intel: fix possible acpi enumeration panic
ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
ASoC: Intel: fix return value check in sst_acpi_probe()
ALSA: hda - Make add_stereo_mix_input flag tristate
ALSA: hda - Create capture source ctls when stereo mix input is added
ALSA: hda - Fix typos in snd_hda_get_int_hint() kerneldoc comments
ALSA: hda - add codec ID for Skylake display audio codec
ALSA: oxfw: some signedness bugs
ALSA: oxfw: fix detect_loud_models() return value
ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
...
The Arcam rPAC seems to have the same problem - whenever anything
(alsamixer, udevd, 3.9+ kernel from 60af3d037e, ..) attempts to
access mixer / control interface of the card, the firmware "locks up"
the entire device, resulting in
SNDRV_PCM_IOCTL_HW_PARAMS failed (-5): Input/output error
from alsa-lib.
Other operating systems can somehow read the mixer (there seems to be
playback volume/mute), but any manipulation is ignored by the device
(which has hardware volume controls).
Cc: <stable@vger.kernel.org>
Signed-off-by: Jiri Jaburek <jjaburek@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The num_controls field was incorrectly set to 0 causing 6i6 to not be
initialized. Set this to 9.
Reported-and-tested-by: Mark Roberts <sunifiram@gmail.com>
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for two XMOS based DACs from Matrix Audio:
- X-Sabre
- Mini-i Pro
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
* ALSA core
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
* USB-audio
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
* FireWire
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
* HD-audio
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
* ASoC
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
* Others
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle
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Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
ALSA core:
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
USB-audio:
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
FireWire:
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
HD-audio:
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
ASoC:
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
Others:
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle"
* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
ALSA: pcxhr: NULL dereference on probe failure
ALSA: lola: NULL dereference on probe failure
ALSA: hda - Add "eapd" model string for AD1986A codec
ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
ALSA: oxfw: Add hwdep interface
ALSA: oxfw: Add support for capture/playback MIDI messages
ALSA: oxfw: add support for capturing PCM samples
ALSA: oxfw: Add support AMDTP in-stream
ALSA: oxfw: Add support for Behringer/Mackie devices
ALSA: oxfw: Change the way to start stream
ALSA: oxfw: Add proc interface for debugging purpose
ALSA: oxfw: Change the way to make PCM rules/constraints
ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
ALSA: oxfw: Change the way to name card
ALSA: dice: Add support for MIDI capture/playback
ALSA: dice: Add support for capturing PCM samples
ALSA: dice: Support for non SYT-Match sampling clock source mode
ALSA: dice: Add support for duplex streams with synchronization
ALSA: dice: Change the way to start stream
ALSA: jack: Add dummy snd_jack_set_key() definition
...
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Merge tag 'media/v3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media updates from Mauro Carvalho Chehab:
- Two new dvb frontend drivers: mn88472 and mn88473
- A new driver for some PCIe DVBSky cards
- A new remote controller driver: meson-ir
- One LIRC staging driver got rewritten and promoted to mainstream:
igorplugusb
- A new tuner driver (m88rs6000t)
- The old omap2 media driver got removed from staging. This driver
uses an old DMA API and it is likely broken on recent kernels.
Nobody cared enough to fix it
- Media bus format moved to a separate header, as DRM will also use the
definitions there
- mem2mem_testdev were renamed to vim2m, in order to use the same
naming convention taken by the other virtual test driver (vivid)
- Added a new driver for coda SoC (coda-jpeg)
- The cx88 driver got converted to use videobuf2 core
- Make DMABUF export buffer to work with DMA Scatter/Gather and Vmalloc
cores
- Lots of other fixes, improvements and cleanups on the drivers.
* tag 'media/v3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (384 commits)
[media] mn88473: One function call less in mn88473_init() after error
[media] mn88473: Remove uneeded check before release_firmware()
[media] lirc_zilog: Deletion of unnecessary checks before vfree()
[media] MAINTAINERS: Add myself as img-ir maintainer
[media] img-ir: Don't set driver's module owner
[media] img-ir: Depend on METAG or MIPS or COMPILE_TEST
[media] img-ir/hw: Drop [un]register_decoder declarations
[media] img-ir/hw: Fix potential deadlock stopping timer
[media] img-ir/hw: Always read data to clear buffer
[media] redrat3: ensure dma is setup properly
[media] ddbridge: remove unneeded check before dvb_unregister_device()
[media] si2157: One function call less in si2157_init() after error
[media] tuners: remove uneeded checks before release_firmware()
[media] arm: omap2: rx51-peripherals: fix build warning
[media] stv090x: add an extra protetion against buffer overflow
[media] stv090x: Remove an unreachable code
[media] stv090x: Some whitespace cleanups
[media] em28xx: checkpatch cleanup: whitespaces/new lines cleanups
[media] si2168: add support for firmware files in new format
[media] si2168: debug printout for firmware version
...
This makes the midi interface and capture work out of the box with
R16 (and presumably R24 too but untested). Playback stream would also
seem to function fine except for one caveat: no sound is produced,
so it is disabled for now. Mixer descriptors are garbage and will
require further quirks to enable functionality, also disabled here.
Signed-off-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [7a2e9ddc: ALSA: usb-audio: Add native DSD support for
Denon/Marantz DACs] requires the new format definition that has
landed only in for-next branch.
Denon/Marantz USB DACs need a specific vendor command to switch between PCM and
DSD mode. This patch adds a new quirk function to switch between the two modes
using the specific USB vendor command.
This patch applies to the following devices:
- Marantz SA-14S1
- Marantz HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the following devices:
- Marantz SA-14S1
- Marants HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds
DSD_U16_BE and DSD_U32_BE sample formats.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Acked-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Scarlett driver uses almost compatible usb_mixer_elem_info struct, so
we just need to add a couple of simple resume callbacks to handle them
accordingly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous fixes, the mixer accessors are converted to use
usb_mixer_elem_list objects. In addition, the proper shutdown check
are put in get and put callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few FTU mixer controls have the own value handling, so they have to
be rewritten to follow the support for resume callbacks. This ended
up in a fair amount of refactoring. Its own struct is now removed,
instead the values are embedded in kctl private_value totally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The changes at this time are a bit more wider than previous ones.
Firstly, the NI controls didn't cache the values, so I had to
implement the caching. It's stored in bit 24 of private_value.
In addition to that, the initial values have to be read from
registers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time it's about Xonar U1: add the proper resume support for
"Digital Playback Switch" element.
Also, the status is moved into kcontrol private_value from
usb_mixer_interface struct field. One more cut.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar as the previous fix, this adds the proper resume support to
Emu0202 "Front Jack Channels" enum mixer element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rewrite the code to handle LEDs on audigy2nx and co for supporting the
proper resume. A new internal helper function
add_single_ctl_with_resume() is introduced to manage the
usb_mixer_elem_list more easily.
Also while we're at it, move audigy2nx_leds[] in usb_mixer_interface
struct into the private_value of each kctl, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, we blindly assumed that the all usb-audio mixer elements
follow the standard and apply the standard resume method for the
registered elements in the id_elems[] list. However, some quirks
really need the own resume and it's incomplete for now.
This patch enhances the resume handling in two folds:
- split some fields in struct usb_mixer_elem_info into a smaller
header struct (usb_mixer_elem_list) for keeping the minimal
information in the linked-list; the usb_mixer_elem_info embeds this
header struct instead
- add resume and dump callbacks to usb_mixer_elem_list struct to allow
quirks providing the own methods
For the standard mixer elements, these new callbacks are set to the
standard ones as default, thus there is no functional change by this
patch yet.
The dump and resume callbacks are typedef'ed for ease of later patches
using arrays of such function pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce an internal helper macro for avoiding many open codes.
The only slight behavior change is in a couple of get ballcks where
the value is reset at error no matter whether ignore_ctl_error is set
or not. Actually this is even safer than before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_nativeinstruments_control_get() uses a stack as a buffer for
usb_control_msg(), but it's basically not allowed. Replace the call
with a safer helper, snd_usb_ctl_msg(), instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Specified in section 5.2.5.6.1 of the USB Audio Class 2.0 definition.
Solves the following error for C-Media 6632A (Asus Xonar U7):
[ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a USB control message delay quirk for a few specific Marantz/Denon
devices. Without the delay the DACs will not work properly and produces the
following type of messages:
Nov 15 10:09:21 orwell kernel: [ 91.342880] usb 3-13: clock source 41 is not valid, cannot use
Nov 15 10:09:21 orwell kernel: [ 91.343775] usb 3-13: clock source 41 is not valid, cannot use
There are likely other Marantz/Denon devices using the same USB module which exhibit the
same problems. But as this cannot be verified I limited the patch to the devices
I could test.
The following two devices are covered by this path:
- Marantz SA-14S1
- Marantz HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the inline function instead of directly indexing the array.
This allows some architectures with hardware instructions
for bit reversals to eliminate the array.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code contains the Scarlett mixer interface code that was originally
written by Tobias Hoffman and Robin Gareus. Because the device doesn't
properly implement UAC2 this code adds a mixer quirk for the device.
Changes from the original code include removing the metering code along with
dead code and comments. Compiler warnings were fixed. The code to initialize
the sampling rate was causing a crash this was fixed as discussed on the
mailing list. Error, and info messages were convered to dev_err and dev_info
interfaces. The custom scarlett_mixer_elem_info struct was replaced with the
more generic usb_mixer_elem_info to be able to recycle more code from mixer.c.
This patch also makes additional modifications based on upstream comments.
Individual control creation functions are removed and a generic
function is no used. Macros for function calls are removed to improve
readability. Hardcoded control initialization is removed. Save to HW
functionality has been removed. Strings for enums are created dynamically for
the mixer. Strings used for controls are now SNDRV_CTL_ELEM_ID_NAME_MAXLEN
length.
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the functions set_cur_mix_value and get_cur_mix_value accessible by files
that include mixer.h. In addition make usb_mixer_elem_free accessible.
This allows reuse of these functions by mixers that may require quirks.
The following summarizes the renamed functions:
- set_cur_mix_value -> snd_usb_set_cur_mix_value
- get_cur_mix_value -> snd_usb_get_cur_mix_value
- usb_mixer_elem_free -> snd_usb_mixer_elem_free
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a private_data pointer to usb_mixer_elem_info to allow other mixer
implementations to extend the structure as necessary.
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 1762a59d8e.
This quirk is not needed because support for the Scarlett mixers will be added.
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory.
This patch adds the private_free callback to release it properly.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch provides duplex support for the Digidesign Mbox 1 sound
card and has been a work in progress for about a year.
Users have confirmed on my website that previous versions of this patch
have worked on the hardware and I have been testing extensively.
It also enables the mixer control for providing clock source
selector based on the previous patch.
The sample rate has been hardcoded to 48kHz because it works better with
the S/PDIF sync mode when the sample rate is locked. This is the
highest rate that the device supports and no loss of functionality
is observed by restricting the sample rate apart from the inability to selec
a lower rate.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch provides the infrastructure for the Digidesign Mbox 1
to have a mixer control for selecting the clock source.
Valid options are Internal and S/PDIF external sync.
A non-documented command is sent to the device to enable this feature
found by reverse engineering and bus snooping.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Needed due to some important regression fixes at RC core.
* commit 'v3.18-rc4': (587 commits)
Linux 3.18-rc4
ARM: dts: zynq: Enable PL clocks for Parallella
tiny: rename ENABLE_DEV_COREDUMP to ALLOW_DEV_COREDUMP
tiny: reverse logic for DISABLE_DEV_COREDUMP
i2c: core: Dispose OF IRQ mapping at client removal time
i2c: at91: don't account as iowait
i2c: remove FSF address
USB: Update default usb-storage delay_use value in kernel-parameters.txt
sysfs: driver core: Fix glue dir race condition by gdp_mutex
MIPS: Fix build with binutils 2.24.51+
xfs: track bulkstat progress by agino
xfs: bulkstat error handling is broken
xfs: bulkstat main loop logic is a mess
xfs: bulkstat chunk-formatter has issues
xfs: bulkstat chunk formatting cursor is broken
xfs: bulkstat btree walk doesn't terminate
mm: Fix comment before truncate_setsize()
USB: cdc-acm: add quirk for control-line state requests
tty: Fix pty master poll() after slave closes v2
MIPS: R3000: Fix debug output for Virtual page number
...
Conflicts:
drivers/media/rc/rc-main.c
The quirk argument itself was used as iterator, so it cannot be taken
back to the original value, obviously.
Fixes: d4b8fc66f7 ('ALSA: usb-audio: Allow multiple entries for the same iface in composite quirk')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the composite quirk doesn't work when multiple entries are
assigned to the same interface because it marks the interface as
claimed then checks whether the interface has been already claimed for
the secondary entry. But, if you look at the code, you'll notice that
multiple entries are allowed if the entry is the current interface;
i.e. the current behavior is anyway inconsistent, and this is an
unintended shortcoming.
This patch fixes the problem by marking the relevant interfaces as
claimed after applying the all composite entries. This fix will be
needed for the upcoming enhancements for Digidesign Mbox 1 quirks.
Reviewed-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function snd_pcm_stop_xrun() to the standard sequnce
lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the
existing open codes with this helper.
The function checks the PCM running state to prevent setting the wrong
state, too, for more safety.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver detects XRUN at its complete callback, but the
actual code to trigger PCM XRUN is commented out because it caused
deadlock in the past. This patch revives the PCM trigger properly.
It resulted in more than just enabling snd_pcm_stop(), but it had to
deduce the PCM substream with proper NULL checks and holds the stream
lock around the call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the USB-audio disconnect fix and resolves the conflicts
so that we can continue working on development of usb-audio stuff.
Conflicts:
sound/usb/card.c
Some USB-audio devices show weird sysfs warnings at disconnecting the
devices, e.g.
usb 1-3: USB disconnect, device number 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 973 at fs/sysfs/group.c:216 device_del+0x39/0x180()
sysfs group ffffffff8183df40 not found for kobject 'midiC1D0'
Call Trace:
[<ffffffff814a3e38>] ? dump_stack+0x49/0x71
[<ffffffff8103cb72>] ? warn_slowpath_common+0x82/0xb0
[<ffffffff8103cc55>] ? warn_slowpath_fmt+0x45/0x50
[<ffffffff813521e9>] ? device_del+0x39/0x180
[<ffffffff81352339>] ? device_unregister+0x9/0x20
[<ffffffff81352384>] ? device_destroy+0x34/0x40
[<ffffffffa00ba29f>] ? snd_unregister_device+0x7f/0xd0 [snd]
[<ffffffffa025124e>] ? snd_rawmidi_dev_disconnect+0xce/0x100 [snd_rawmidi]
[<ffffffffa00c0192>] ? snd_device_disconnect+0x62/0x90 [snd]
[<ffffffffa00c025c>] ? snd_device_disconnect_all+0x3c/0x60 [snd]
[<ffffffffa00bb574>] ? snd_card_disconnect+0x124/0x1a0 [snd]
[<ffffffffa02e54e8>] ? usb_audio_disconnect+0x88/0x1c0 [snd_usb_audio]
[<ffffffffa015260e>] ? usb_unbind_interface+0x5e/0x1b0 [usbcore]
[<ffffffff813553e9>] ? __device_release_driver+0x79/0xf0
[<ffffffff81355485>] ? device_release_driver+0x25/0x40
[<ffffffff81354e11>] ? bus_remove_device+0xf1/0x130
[<ffffffff813522b9>] ? device_del+0x109/0x180
[<ffffffffa01501d5>] ? usb_disable_device+0x95/0x1f0 [usbcore]
[<ffffffffa014634f>] ? usb_disconnect+0x8f/0x190 [usbcore]
[<ffffffffa0149179>] ? hub_thread+0x539/0x13a0 [usbcore]
[<ffffffff810669f5>] ? sched_clock_local+0x15/0x80
[<ffffffff81066c98>] ? sched_clock_cpu+0xb8/0xd0
[<ffffffff81070730>] ? bit_waitqueue+0xb0/0xb0
[<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore]
[<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore]
[<ffffffff8105973e>] ? kthread+0xce/0xf0
[<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0
[<ffffffff814a8b7c>] ? ret_from_fork+0x7c/0xb0
[<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0
---[ end trace 40b1928d1136b91e ]---
This comes from the fact that usb-audio driver may receive the
disconnect callback multiple times, per each usb interface. When a
device has both audio and midi interfaces, it gets called twice, and
currently the driver tries to release resources at the last call.
At this point, the first parent interface has been already deleted,
thus deleting a child of the first parent hits such a warning.
For fixing this problem, we need to call snd_card_disconnect() and
cancel pending operations at the very first disconnect while the
release of the whole objects waits until the last disconnect call.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80931
Reported-and-tested-by: Tomas Gayoso <tgayoso@gmail.com>
Reported-and-tested-by: Chris J Arges <chris.j.arges@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some functions in mixer.c and endpoint.c receive list_head instead of
the object itself. This is not obvious and rather error-prone. Let's
pass the proper object directly instead.
The functions in midi.c still receive list_head and this can't be
changed since the object definition isn't exposed to the outside of
midi.c, so left as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio probe and disconnect functions have been split just for
adapting the (new!) API at 2.5 kernel time. We left them until now,
partly because we wanted to build with the pretty old kernels in the
external alsa-driver tree. But the support of such old kernels has
been longly stopped, so it's good time to clean up this mess.
One good point by this cleanup is that now the probe function returns
a proper error code instead of only -EIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The au0828 quirks table is currently not in sync with the au0828
media driver.
Syncronize it and put them on the same order as found at au0828
driver, as all the au0828 devices with analog TV need the
same quirks.
Cc: stable@vger.kernel.org
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
Add a macro to simplify au0828 quirk table. That makes easier
to check it against the USB IDs at drivers/media/usb/au0828/au0828-cards.c.
Cc: stable@vger.kernel.org
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
Here are a chunk of small fixes since rc1: two PCM core fixes, one is
a long-standing annoyance about lockdep and another is an ARM64 mmap
fix. The rest are a HD-audio HDMI hotplug notification fix, a fix for
missing NULL termination in Realtek codec quirks and a few new
device/codec-specific quirks as usual.
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Merge tag 'sound-3.18-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a chunk of small fixes since rc1: two PCM core fixes, one is
a long-standing annoyance about lockdep and another is an ARM64 mmap
fix.
The rest are a HD-audio HDMI hotplug notification fix, a fix for
missing NULL termination in Realtek codec quirks and a few new
device/codec-specific quirks as usual"
* tag 'sound-3.18-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Add missing terminating entry to SND_HDA_PIN_QUIRK macro
ALSA: pcm: Fix false lockdep warnings
ALSA: hda - Fix inverted LED gpio setup for Lenovo Ideapad
ALSA: hda - hdmi: Fix missing ELD change event on plug/unplug
ALSA: usb-audio: Add support for Steinberg UR22 USB interface
ALSA: ALC283 codec - Avoid pop noise on headphones during suspend/resume
ALSA: pcm: use the same dma mmap codepath both for arm and arm64
this is a series of patches to just convert the plain info callback
for enum ctl elements to snd_ctl_elem_info(). Also, it includes the
extension of snd_ctl_elem_info(), for catching the unexpected string
cut-off and handling the zero items.
Don't assign 'len' in cases where we don't make use of the returned value.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time it's a relatively calm update batch, but the amount isn't
too small in the end. Here we go over some highlights:
- ALSA core
- One major change is the support of nonatomic PCM operations.
This allows the trigger and other callbacks to call schedule(),
which would be useful for mailbox type communications. Already
some drivers (Digigram ones) have been converted to use together
with threaded irqs as an example.
- Improvement / fixes of DSD PCM format support
- HD-audio
- Large volume of rewrites are found in Realtek codec driver for
converting Dell and HP quirks to generic forms.
- Inverted dmic code cleanup from David.
- Realtek COEF access has been optimized.
- Now HD-audio jack infrastructure allows multiple callbacks, which
fixes / simplifies the jack-dependent power controls on STAC/IDT
and VIA codecs.
- Many additional device-specific fixups as usual
- A few deadcode cleanups, CA0132 code cleanup, etc.
- ASoC
- More componentization work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes
and enhancements for the associated CODEC drivers, this is going
to need a lot quirks over time due to the lack of any firmware
description of the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale
drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32,
Everest Semiconductor ES8328 and Freescale cards using the ASRC
in newer i.MX processors.
- A few simple-card fixes, mostly cleanups but also a fix for
- interaction between GPIO 0 and simple-card.
- Misc
- Virtuoso / Oxygen updates by Clemens
- USB-audio: Yamaha MOTIF XF MIDI port name fixes
- Conversion of kernel messages to standard dev_*() in ctxfi
driver.
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Merge tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This time it's a relatively calm update batch, but the amount isn't
too small in the end. Here we go over some highlights:
ALSA core:
- One major change is the support of nonatomic PCM operations. This
allows the trigger and other callbacks to call schedule(), which
would be useful for mailbox type communications. Already some
drivers (Digigram ones) have been converted to use together with
threaded irqs as an example.
- Improvement / fixes of DSD PCM format support
HD-audio:
- Large volume of rewrites are found in Realtek codec driver for
converting Dell and HP quirks to generic forms.
- Inverted dmic code cleanup from David.
- Realtek COEF access has been optimized.
- Now HD-audio jack infrastructure allows multiple callbacks, which
fixes / simplifies the jack-dependent power controls on STAC/IDT
and VIA codecs.
- Many additional device-specific fixups as usual
- A few deadcode cleanups, CA0132 code cleanup, etc.
ASoC:
- More componentization work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to
need a lot quirks over time due to the lack of any firmware
description of the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32,
Everest Semiconductor ES8328 and Freescale cards using the ASRC in
newer i.MX processors.
- A few simple-card fixes, mostly cleanups but also a fix for
interaction between GPIO 0 and simple-card.
Misc:
- Virtuoso / Oxygen updates by Clemens
- USB-audio: Yamaha MOTIF XF MIDI port name fixes
- Conversion of kernel messages to standard dev_*() in ctxfi driver"
* tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (251 commits)
ASoC: mc13783: Ensure we only try to dereference valid of_nodes
ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl
ALSA: hda - Add dock port support to Thinkpad L440 (71aa:501e)
ALSA: Allow pass NULL dev for snd_pci_quirk_lookup()
ASoC: imx-es8328: Fix of_node_put() call with uninitialized object
ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
ASoC: simple-card: Initialize headphone and mic GPIO numbers
ASoC: imx-es8328: Fix missing return code in imx_es8328_probe()
ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
ALSA: usb: caiaq: check for cdev->n_streams > 1
ASoC: 88pm860x-codec: Fix possibly missing string termination
ASoC: core: fix use after free in snd_soc_remove_platform()
ASoC: soc-dapm: fix use after free
ALSA: hda - Make the inv dmic handling for Realtek use generic parser
ALSA: hda - Add Inverted Internal mic for Samsung Ativ book 9 (NP900X3G)
ALSA: hda - Add inverted internal mic for Asus Aspire 4830T
ASoC: Intel: byt-rt5640: fix coccinelle warnings
ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string
ASoC: da732x: Remove unnecessary KERN_ERR in pr_err()
ASoC: simple-card: Fix detect gpio documentation.
...
Coverity spotted a possible DIV0 condition when cdev->n_streams is 0.
Fix this by making sure the value is > 1 in snd_usb_caiaq_audio_init().
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
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Merge tag 'asoc-v3.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.18
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
includes miscellaneous cleanup of other PHY drivers.
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Merge tag 'phy-for_3.18' of git://git.kernel.org/pub/scm/linux/kernel/git/kishon/linux-phy into usb-next
Kishon writes:
Adds 3 new PHY drivers stih407, stih41x and rcar gen2 PHY. It also
includes miscellaneous cleanup of other PHY drivers.
Conflicts:
MAINTAINERS
USB hub has started to use a workqueue instead of kthread. Let's update
the documentation and comments here and there.
This patch mostly just replaces "khubd" with "hub_wq". There are only few
exceptions where the whole sentence was updated. These more complicated
changes can be found in the following files:
Documentation/usb/hotplug.txt
drivers/net/usb/usbnet.c
drivers/usb/core/hcd.c
drivers/usb/host/ohci-hcd.c
drivers/usb/host/xhci.c
Signed-off-by: Petr Mladek <pmladek@suse.cz>
Acked-by: Alan Stern <stern@rowland.harvard.edu>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
KoreController and KoreController2 need an EP1_CMD_DIMM_LEDS command to set
their LEDs, not EP1_CMD_WRITE_IO.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-and-tested-by: Brad Wilson <brad.wilson.00@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirks for XMOS based DACs for native DSD playback support using the new
DSD_U32_LE sample format.
This version adds native DSD support for:
- iFi Audio micro iDSD/nano iDSD (they use the same prod. id)
- DIYINHK USB to I2S/DSD converter
Changes from v2:
- fix and simplify switch statement
Changes from v1:
- use specific product id and alt setting per XMOS based device
[fixed a misc coding style issue by tiwai]
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BOSS ME-25 turns out not to have any useful descriptors in its MIDI
interface, so its needs a quirk entry after all.
Reported-and-tested-by: Kees van Veen <kees.vanveen@gmail.com>
Fixes: 8e5ced83dd ("ALSA: usb-audio: remove superfluous Roland quirks")
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/card.c registers USB suspend and resume but did not previously
kill the input URBs. This means that USB MIDI devices left open across
suspend/resume had non-functional input (output still usually worked,
but it looks like that is another issue). Before this change, we would
get ESHUTDOWN for each of the input URBs at suspend time, killing input.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original patch fixed the original problem, but the sound was far too low
for most users. This patch references a compare matrix to allow the
volume levels to act normally. I personally tested this patch myself,
and volume levels returned to normal. Please see this discussion for
more details: https://bugzilla.kernel.org/show_bug.cgi?id=65251
Signed-off-by: Paul S McSpadden <fisch602@gmail.com>
Cc: <stable@vger.kernel.org> [v3.14+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs(). That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.
Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep. The problem is the
succeeding kfree() in snd_pcm_endpoint_free().
This patch moves out the EP deallocation into the later point, the
destructor callback. At this stage, all PCMs must have been already
closed, so it's safe to free the objects.
Reported-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.
Add a workaround to detect and fix the corruption.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls. For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.
Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance. In such a case, it's superfluous to save the mixer
values multiple times. This patch fixes it by checking the counter.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This (widely used) construction:
if(printk_ratelimit())
dev_dbg()
Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.
[ 533.803964] retire_playback_urb: 852 callbacks suppressed
[ 538.807930] retire_playback_urb: 852 callbacks suppressed
[ 543.811897] retire_playback_urb: 852 callbacks suppressed
[ 548.815745] retire_playback_urb: 852 callbacks suppressed
[ 553.819826] retire_playback_urb: 852 callbacks suppressed
So use dev_dbg_ratelimited() instead of this construction.
Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/1305133
Malfunctioning or slow devices can cause a flood of dmesg SPAM.
I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour
of prior art in sound/usb/pcm.c.
WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit
+ if (printk_ratelimit() &&
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Eldad Zack <eldad@fogrefinery.com>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds initial support for the Behringer BCD2000 USB DJ controller.
At the moment, only the MIDI part of the device is working, i.e. knobs,
buttons and LEDs.
I also plan to add support for the audio part, but I assume that this will
require more effort than the rather simple MIDI interface. Progress can be
tracked at https://github.com/anyc/snd-usb-bcd2000.
Signed-off-by: Mario Kicherer <dev@kicherer.org>
Reviewed-by: Daniel Mack <daniel@zonque.org>
Reviewed-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
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Merge tag 'asoc-v3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
Logitech C500 (046d:0807) needs the same workaround like other
Logitech Webcams.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
As there are too deep indirections (e.g. ep->chip->dev->dev),
a few new local macros, usb_audio_err() & co, are introduced.
Also, the device numbers in some messages are dropped, as they are
shown in the prefix automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid traversing the device object list of the card instance just for
checking the PCM streams. The driver's private object already
contains the array of substream pointers, so it can be simply looked
through. The card internal may be restructured in future, thus better
not to rely on it.
Also, this fixes the possible deadlocks in PCM mutex. Instead of
taking multiple PCM mutexes, just take the common mutex in all
places. Along with it, rename prepare_mutex as pcm_mutex.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the driver tries to access Function Unit 10, the KEF X300A
speakers' firmware apparently locks up, making even PCM streaming
impossible. Work around this by ignoring this FU.
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of SNDRV_DEV_LOWLEVEL, use SNDRV_DEV_CODEC type for mixer
objects so that they are managed in a proper release order.
No functional change at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement reset_resume callback so that the mixer values are properly
restored. Still no boot quirks are called, so it might not work well
on some devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 44dcbbb1cd introduced the usage of bitreverse helpers but
forgot to add the dependency. This patch adds the selection for
CONFIG_BITREVERSE.
Fixes: 44dcbbb1cd ('ALSA: snd-usb: add support for bit-reversed byte formats')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No code change, just a cosmetic cleanup to keep entries ordered by the
device ID within a block of unique vendor IDs.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Creative Live! Cam Vista IM (VF0420) reports rate of 16kHz while working
at 8kHz. The patch adds its USB ID to the existing quirk.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Vaughan device support the 352800 rate and not
the 352000
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics Gamecom 780 headset has a firmware problem, and when the
FU 0x09 volume is changed, it results in either too loud or silence
except for a very narrow range. This patch provides a workaround,
ignoring the node, initialize the volume in a sane value and keep
untouched.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5. This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.
Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for front jack channel selector which is present on EMU0204.
It allows to get 4 channels out of this soundcard.
Tested-by: Yury Bushmelev <jay@jay-tech.ru>
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to USB Audio spec v2 bits 25 and 26 of bmChannelConfig are
"Back Left of Center - BLC" and "Back Right of Center - BRC",
respectively.
They are currently assigned to ALSA channels BLC/BRC. However, the ALSA
BLC/BRC are actually the rather nonsensical "bottom left center" and
"bottom right center", so the channels will be assigned wrongly. The
comments in the USB code are also similarly wrong, so this is not
readily apparent without looking at the actual specification.
Fix the channel mapping by mapping bits 25 and 26 to RLC (Rear Left
Center) and RRC (Rear Right Center), respectively, instead.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case the channel count of the input terminal is not the same as
the channel count of the streaming descriptor, the channel config of
the input terminal can not be trusted. Instead fall back to a default
(guessed) channel map.
This was found on a Logitech USB Headset.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The channel config from the streaming descriptor is probably a
better indicator of the channel map than the input terminal.
Use the input terminal's channel map as fallback only.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If wChannelconfig is given for some formats but not others, userspace
might not be able to set the channel map.
This is RFC because I'm not sure what the best behaviour is - to guess
the channel map from the given number of channels (it's quite likely
that one channel is MONO and two channels is FL FR), or just to supply
UNKNOWN for all channels.
But the complete lack of channel map for a format leads userspace to
believe that the format is not available at all. Or am I
misunderstanding how this should be used?
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The probe code of snd-usb-6fire driver overrides the devices[] pointer
wrongly without checking whether it's already occupied or not. This
would screw up the device disconnection later.
Spotted by coverity CID 141423.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
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Merge tag 'asoc-v3.13' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.13
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
The pcm_usb_stream plugin requires the mremap explicitly for the read
buffer, as it expands itself once after reading the required size.
But the commit [314e51b9: mm: kill vma flag VM_RESERVED and
mm->reserved_vm counter] converted blindly to a combination of
VM_DONTEXPAND | VM_DONTDUMP like other normal drivers, and this
resulted in the failure of mremap().
For fixing this regression, we need to remove VM_DONTEXPAND for the
read-buffer mmap.
Reported-and-tested-by: James Miller <jamesstewartmiller@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for dev speed USB_SPEED_WIRELESS in
snd_usb_parse_datainterval which allows the usb sound core to create
ISO urbs with the correct number and size of buffers.
Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As Clemens Ladisch kindly explained:
"Please note that there are two methods to identify alternate settings:
the number, which is the value in bAlternateSetting, and the index,
which is the index in the descriptor array. There might be some wording
in the USB spec that these two values must be the same, but in reality,
[insert standard rant about firmware writers], bAlternateSetting
must be treated as a random ID value."
This patch changes the name to express the correct usage semantics.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of snd_usb_endpoint_deactivate() is not used,
make the function have no return value.
Update the documentation to reflect what the function is actually
doing.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If an endpoint in use, its associated URBs should not be
deactivated.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only call site for deactivate_endpoints() at snd_usb_hw_free().
The return value is not checked there, as it is irrelevant if it
fails on hw_free.
This patch moves the deactivation of the endpoints directly into
snd_usb_hw_free().
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the format is not actually used in sync_ep_set_params(),
there is no need to pass it down.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The frame check in i_usX2Y_urb_complete() and
i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as
described in this LAU thread:
http://linuxaudio.org/mailarchive/lau/2013/5/20/200177
This patch removes the check code entirely.
Cc: fzu@wemgehoertderstaat.de
Reported-by: Dr Nicholas J Bailey <nicholas.bailey@glasgow.ac.uk>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds LED support for the Native Instruments Maschine
Controller. It adds ALSA controls for dimming the LEDs of all
buttons and the backlight of the two displays.
Signed-off-by: Hannes Gräuler <hgraeule@uos.de>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the way URBs are allocated and their sizes are
determined for PCM playback in the snd-usb-audio driver. Currently
the driver allocates too few URBs for endpoints that don't use
implicit sync, making underruns more likely to occur. This may be a
holdover from before I/O delays could be measured accurately; in any
case, it is no longer necessary.
The patch allocates as many URBs as possible, subject to four
limitations:
The total number of URBs for the endpoint is not allowed to
exceed MAX_URBS (which the patch increases from 8 to 12).
The total number of packets per URB is not allowed to exceed
MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is
decreased from 20 to 6.
The total duration of queued data is not allowed to exceed
MAX_QUEUE, which is decreased from 24 ms to 18 ms.
The total number of ALSA frames in the output queue is not
allowed to exceed the ALSA buffer size.
The last requirement is the hardest to implement. Currently the
number of URBs needed to fill a buffer cannot be determined in
advance, because a buffer contains a fixed number of frames whereas
the number of frames in an URB varies to match shifts in the device's
clock rate. To solve this problem, the patch changes the logic for
deciding how many packets an URB should contain. Rather than using as
many as possible without exceeding an ALSA period boundary, now the
driver uses only as many packets as needed to transfer a predetermined
number of frames. As a result, unless the device's clock has an
exceedingly variable rate, the number of URBs making up each period
(and hence each buffer) will remain constant.
The overall effect of the patch is that playback works better in
low-latency settings. The user can still specify values for
frames/period and periods/buffer that exceed the capabilities of the
hardware, of course. But for values that are within those
capabilities, the performance will be improved. For example, testing
shows that a high-speed device can handle 32 frames/period and 3
periods/buffer at 48 KHz, whereas the current driver starts to get
glitchy at 64 frames/period and 2 periods/buffer.
A side effect of these changes is that the "nrpacks" module parameter
is no longer used. The patch removes it.
Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
CC: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Daniel Mack <zonque@gmail.com>
Tested-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert 0 to false and 1 to true when assigning values to bool
variables. Inspired by commit 3db1cd5c05.
The simplified semantic patch that find this problem is as
follows (http://coccinelle.lip6.fr/):
@@
bool b;
@@
(
-b = 0
+b = false
|
-b = 1
+b = true
)
Signed-off-by: Peter Senna Tschudin <peter.senna@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
Add the volume control quirk for avoiding the kernel warning
for the Logitech HD Webcam C525
as in the similar commit 36691e1be6
for the Logitech HD Webcam C310.
Reported-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Tested-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Cc: <stable@vger.kernel.org> # 3.10.5+
Signed-off-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit aafe77cc45 (ALSA: usb-audio: add support for many Roland/Yamaha
devices) had several logic errors that prevented create_auto_midi_quirk
from enumerating any MIDI ports.
Reported-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Patch makes pcm buffers DMA-able by allocating each one separately.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to pass constants via stack. The width may be explicitly
specified in the format.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver used to assume that the streaming endpoint's wMaxPacketSize
value would be an indication of how much data the endpoint expects or
sends, and compute the number of packets per URB using this value.
However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes,
while only about 88 or 44 bytes are be actually used. This discrepancy
would result in URBs with far too few packets, which would not work
correctly on the EHCI driver.
To get correct URBs, use wMaxPacketSize only as an upper limit on the
packet size.
Reported-by: James Stone <jamesmstone@gmail.com>
Tested-by: James Stone <jamesmstone@gmail.com>
Cc: <stable@vger.kernel.org> # 2.6.35+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to
be DMA-able, which stack is not. Furthermore, transfer_buffer should not be
allocated as part of larger device structure because DMA coherency issues and
patch fixes this issue too.
Cc: stable@vger.kernel.org
Signed-off-by: Jussi Kivilinna <jussi.kivilinna@iki.fi>
Tested-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Prevent NULL dereference in snd_usb_add_endpoints(), when
alts is passed as NULL. In this case, WARN (since this is
a non-fatal bug) and return NULL ep. Call sites treat a NULL
return value as an error.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the quirks all apply to implicit feedback (the source endpoint
is always a data endpoint), there's no need to set and check
a flag for it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An implicit feedback endpoint can only be a capture source. The
consumer (sink) of the implicit feedback endpoint is therefore limited
to playback EPs.
Check if the target endpoint is a playback first and remove redundant
checks.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since implicit_fb is not changed, !implicit_fb will always
be true - it is set only after these checks.
Similarly, there's also no need to set it at the top of the function.
Change the type of implicit_fb to bool (more appropriate).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the sync endpoint currently takes up about half of set_format().
Move it to a dedicated function.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test here is always true because S[i].urb is an array not a pointer.
Also it's bogus because the intent was to test:
if (S->urb[i]) {
instead of:
if (S[i].urb) {
Anyway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can
just remove this.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of hiface, as expected by snd_pcm_update_hw_ptr0().
Caught by sparse.
Cc: Antonio Ospite <ospite@studenti.unina.it>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of 6fire, as expected by snd_pcm_update_hw_ptr0().
Caught by sparse.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_stop() must be called in the PCM substream lock context.
Cc: <stable@vger.kernel.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 8f898e92ae removed the redundant
reads of bInterfaceProtocol from the descriptors, but introduced a
regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT,
since fp->protocol is not set in setup process.
As a consequence, audio streams would not get initialized, as the
following logs show:
[ 48.923043] setting usb interface 3:1
[ 48.923056] Creating new capture data endpoint #81
[ 48.923484] 4:3:1: cannot set freq 48000 to ep 0x81
This patch sets fp->protocol in create_fixed_stream_quirk() and
resolves the regression.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is adding extensive support (beside standard usb audio class)
for Audio Advantage Micro II usb sound card.
Features included:
- Access to AES bits (so now sending the IEC61937 compliant stream is
possible).
- Mixer SPDIF control added to turn on/off the optical transmitter.
Signed-off-by: Przemek Rudy <prudy1@o2.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For adding support for many Roland and Yamaha devices:
* 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate:
ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
ALSA: usb-audio: claim autodetected PCM interfaces all at once
ALSA: usb-audio: remove superfluous Roland quirks
ALSA: usb-audio: add MIDI port names for some Roland devices
ALSA: usb-audio: add support for many Roland/Yamaha devices
ALSA: usb-audio: detect implicit feedback on Roland devices
ALSA: usb-audio: store protocol version in struct audioformat
The Roland Quad/Octo-Capture devices use some unknown vendor-specific
mechanism to switch sample rates (and to manage other controls). To
prevent the driver from attempting to use any other than the default
44.1 kHz sample rate, use quirks to hide the other alternate settings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
snd_card_register() registers all devices newly added since the last
call. However, the playback/capture streams are handled as one ALSA
device, so the second /dev device will not be registered if the PCM
streams are added in two steps.
QUIRK_AUTODETECT caused the probe callback to be called once for each
interface, which triggered this problem. Work around this by handling
this like the composite quirk, i.e., autodetecting all other interfaces
that might be used for PCM or MIDI.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Remove all quirks that are no longer needed now that the generic Roland
quirks can handle the vendor-specific descriptors correctly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add quirks to detect the various vendor-specific descriptors used by
Roland and Yamaha in most of their recent USB audio and MIDI devices.
Together with the previous patch, this should add audio/MIDI support for
the following USB devices:
- Edirol motion dive .tokyo performance package
- Roland MC-808 Synthesizer
- Roland BK-7m Synthesizer
- Roland VIMA JM-5/8 Synthesizer
- Roland SP-555 Sequencer
- Roland V-Synth GT Synthesizer
- Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ
- Edirol V-Mixer M-200i/300/380/400/480/R-1000
- BOSS GT-10B Effects Processor
- Roland Fantom G6/G7/G8 Keyboard
- Cakewalk Sonar V-Studio 20/100/700 Audio Interface
- Roland GW-8 Keyboard
- Roland AX-Synth Keyboard
- Roland JUNO-Di/STAGE/Gi Keyboard
- Roland VB-99 Effects Processor
- Cakewalk UM-2G MIDI Interface
- Roland A-500S Keyboard
- Roland SD-50 Synthesizer
- Roland OCTAPAD SPD-30 Controller
- Roland Lucina AX-09 Synthesizer
- BOSS BR-800 Digital Recorder
- Roland DUO/TRI-CAPTURE (EX) Audio Interface
- BOSS RC-300 Loop Station
- Roland JUPITER-50/80 Keyboard
- Roland R-26 Recorder
- Roland SPD-SX Controller
- BOSS JS-10 Audio Player
- Roland TD-11/15/30 Drum Module
- Roland A-49/88 Keyboard
- Roland INTEGRA-7 Synthesizer
- Roland R-88 Recorder
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
All the Roland/Edirol/BOSS USB audio devices that need implicit feedback
show this unambiguously in their descriptors, so it might be a good idea
to let the driver detect this.
This should make playback work correctly (at least with Jack) with the
following devices:
- BOSS GT-100
- BOSS JS-8 Jam Station
- Edirol M-16DX
- Roland GAIA SH-01
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Instead of reading bInterfaceProtocol from the descriptor whenever it's
needed, store this value in the audioformat structure. Besides
simplifying some code, this will allow us to correctly handle vendor-
specific devices where the descriptors are marked with other values.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
In sound/usb/card.c and sound/usb/misc/ua101.c there are no spaces
between the vendor and the device names, use this style in the other
drivers too.
This also helps keeping consistency when new drivers copies from the
ones already in the mainline tree.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For USB devices it's not necessary to allocate physically contiguous
buffers.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For USB devices it's not necessary to allocate physically contiguous
buffers.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_card_used variable is only read but never written, remove it.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the previous fix for LogitechHD Webcam c270 in commit
11e7064f35, c310 model also requires the
same workaround for avoiding the kernel warning.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59741
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the Android firmware enables the audio interfaces in accessory
mode, it always declares in the control interface's baInterfaceNr array
that interfaces 0 and 1 belong to the audio function. However, the
accessory interface itself, if also enabled, already is at index 0 and
shifts the actual audio interface numbers to 1 and 2, which prevents the
PCM streaming interface from being seen by the host driver.
To get the PCM interface interface to work, detect when the descriptors
point to the (for this driver useless) accessory interface, and redirect
to the correct one.
Reported-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Tested-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 927c9423dd (ALSA: usb-audio: add
Edirol UM-3G support) used a wrong quirk type, which would make the
driver refuse to attach with the error message "MIDIStreaming interface
descriptor not found".
Cc: <stable@vger.kernel.org> # 3.3 and later
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check only the uppermost 16 bits instead of the whole 32 bits of
the version information. Do this because all firmware version tested
with this version information worked correctly and the strict check
causes problems for several users.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
freqshift is only set for the data endpoint and syncmaxsize is only set
for the sync endpoint. This results in a syncmaxsize of zero used in the
proc output feedback format calculation, which gives a feedback format
incorrectly shown as 8.16 for UAC2 devices.
As neither the data nor the sync endpoint gives all the relevant
content, output the two combined.
Also remove the sync_endpoint "packet size" which is always zero
and the sync_endpoint "momentary freq" which is constant.
Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async.
Reported-by: B. Zhang <bb.zhang@free.fr>
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code does this:
be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1])
Which is effectively (neglecting the index):
be16_to_cpu(be16_to_cpu(*((u16 *) buf)))
This means the int16 in the buffer is not converted at all.
Daniel Mack confirmed that the driver works on little endian
CPUs, leading to the conclusion that the device-side structure
is actually little endian.
This changes the code to use le16_to_cpu().
Caught by sparse.
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes in the USB API ("implement new semantics for
URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the
default, and changed this flag to mean that URBs can be delayed.
This is not the behaviour wanted by any of the audio drivers because
it leads to discontinuous playback with very small period sizes.
Therefore, our URBs need to be submitted without this flag.
Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
....
It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.
Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.
Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.
There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.
When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.
Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.
More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.
Signed-off-by: Daniel Schürmann <daschuer@mixxx.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.
That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.
The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.
ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.
This patch adds support for this by adding a boolean flag to the
audio format struct.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.
The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.
To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.
The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices. However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions. With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.
Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.
Commit f5f165418c (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.
To work around all this, just disable autopm for all USB MIDI devices.
Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.
Userspace expected: L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1
Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.
Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()
Caught by sparse:
sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).
In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.
If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.
This patch does not introduce any logic flow change.
It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor style fix, following a general code style in the kernel.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().
No functional change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.
Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.
The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.
Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
Creation of individual mixer controls may fail, but that shouldn't cause
the entire mixer creation to fail. Even worse, if the mixer creation
fails, that will error out the entire device probing.
All the functions called by parse_audio_unit() should return -EINVAL if
they find descriptors that are unsupported or believed to be malformed,
so we can safely handle this error code as a non-fatal condition in
snd_usb_mixer_controls().
That fixes a long standing bug which is commonly worked around by
adding quirks which make the driver ignore entire interfaces. Some of
them might now be unnecessary.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In check_input_term() and parse_audio_feature_unit(), propagate the
error value that has been returned by a failing function instead of
-EINVAL. That helps cleaning up the error pathes in the mixer.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in
the same way when parsing the unit. Otherwise parse_audio_unit() fails when it
sees an extension unit on a UAC2 device.
UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Playback Design" products need a 50ms delay after setting the USB
interface.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This field may use up to 32 bits, so it should be handled as unsigned
int.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.
Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.
The datainterval is also ignored but there are not currently any quirks
which choose to override this.
Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware also has a PCM capture device which is not implemented in
this patch.
It may be possible to generalise this to Saffire 6 USB support and some
of the other Focusrite interfaces, but as I don't have access to these
devices we should wait until capture support is working first.
Capture support is not implemented because the code assumes the endpoint
to have its own interface (instead, it shares the interface with playback)
and some thought will be needed to lift this limitation.
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the
interface associations come out wrong, which results in the driver
erroring out with the message "Audio class v2 interfaces need an
interface association".
Work around this by searching for the interface association descriptor
also in some other place where it might have ended up.
Reported-and-tested-by: Dave Helstroom <helstroom@google.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix three smatch warnings recently introduced:
sound/usb/caiaq/device.c:166 usb_ep1_command_reply_dispatch() warn:
variable dereferenced before check 'cdev' (see line 163)
sound/usb/caiaq/device.c:517 snd_disconnect() warn: variable
dereferenced before check 'card' (see line 514)
sound/usb/caiaq/input.c:510 snd_usb_caiaq_ep4_reply_dispatch() warn:
variable dereferenced before check 'cdev' (see line 506)
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Get rid of the proprietary functions log() and debug() and use the
generic dev_*() approach. A macro is needed to cast a cdev to a struct
device *.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is needed in order to make the device namespace cleaner, and will
help when moving this driver over to dev_*() logging.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
bootresponse in snd_usb_mbox2_boot_quirk is only 12 (decimal) u8's
long, but i9s passed to snd_usb_ctl_msg as it would be 0x12 (hexa)
long. Fix that by having proper size of the array, i.e. 0x12.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds quirks and mixer support for the M-Audio Fast Track C600 USB
audio interface. This device is very similar to the C400 - the C600
simply has some more inputs and outputs, so the existing C400 support
is extended to support this device as well.
Signed-off-by: Matt Gruskin <matthew.gruskin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the
wrong interface number, which prevented the driver from attaching to the
device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.37+ <stable@vger.kernel.org>
It looks like MODULE_SUPPORTED_DEVICES() is not implemented yet, but
still, having the entries in the list consistently separated by commas
and with balanced parenthesis won't hurt.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 23caaf19b1 (ALSA: usb-mixer: Add support for Audio Class v2.0)
forgot to adjust the length check for UAC 2.0 feature unit descriptors.
This would make the code abort on encountering a feature unit without
per-channel controls, and thus prevented the driver to work with any
device having such a unit, such as the RME Babyface or Fireface UCX.
Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: Matthew Robbetts <wingfeathera@gmail.com>
Tested-by: Michael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add names of the clock sources for the M-Audio Fast Track
C400.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.
Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.
However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?
BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP. But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.
This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.
Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
"bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
invalid midi endpoint.
It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:
no or invalid class specific endpoint descriptor
Now it still prints these messages on hotplug:
snd-usb-audio: probe of ...:1.0 failed with error -5
snd-usb-audio: probe of ...:1.2 failed with error -5
snd-usb-audio: probe of ...:1.3 failed with error -5
But the device works correctly, including the HID support.
The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.
Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable delay report on capture path. The delay is reset when an
URB is retired and increment at each call to .pointer based
on frame counter changes. The precision of the delay
information is limited to 1ms as in the playback case.
This reverts commit 3f94fad095.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This update contains overall only driver-specific fixes.
Slightly large LOC are seen in usb-audio driver for a couple of new
device quirks and cs42l71 ASoC driver for enhanced features.
The others are a few small (regression) fixes HD-audio, and yet other
small / trival ASoC fixes.
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Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This update contains overall only driver-specific fixes. Slightly
large LOC are seen in usb-audio driver for a couple of new device
quirks and cs42l71 ASoC driver for enhanced features. The others are
a few small (regression) fixes HD-audio, and yet other small / trival
ASoC fixes."
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
ALSA: HDA: Fix sound resume hang
ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
ASoC: atmel-ssc: change disable to disable in dts node
ASoC: Prevent pop_wait overwrite
ALSA: usb-audio: ignore-quirk for HP Wireless Audio
ALSA: hda - Always turn on pins for HDMI/DP
ALSA: hda - Fix pin configuration of HP Pavilion dv7
ASoC: core: Fix splitting of log messages
ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
ASoC: cs42l73: Add DAPM events for power down.
ASoC: cs42l73: Add DMIC's as DAPM inputs.
ASoC: sigmadsp: Fix endianness conversion issue
ASoC: tpa6130a2: Use devm_* APIs
This patch is the result of a lot of trial and error, since there are no specs
available for the device.
Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports. Also, MIDI in and MIDI out both work.
Users will notice that the S/PDIF light also flashes when playback or recording
is active. I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.
Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.
[Modified to make a function static by tiwai]
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As Joe Cooper <swelljoe@gmail.com> reported, "On most HP Envy laptops
the snd-usb-audio module causes the system to become unresponsive and
Gnome Shell 3 to crash.".
See also:
http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.html
Add a quirk to ignore this device (for now) to solve the instability
issue and allow other USB audio devices to be used.
Reported-by: Joe Cooper <swelljoe@gmail.com>
Tested-by: Isaac Smith <hunternet93@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull trivial branch from Jiri Kosina:
"Usual stuff -- comment/printk typo fixes, documentation updates, dead
code elimination."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (39 commits)
HOWTO: fix double words typo
x86 mtrr: fix comment typo in mtrr_bp_init
propagate name change to comments in kernel source
doc: Update the name of profiling based on sysfs
treewide: Fix typos in various drivers
treewide: Fix typos in various Kconfig
wireless: mwifiex: Fix typo in wireless/mwifiex driver
messages: i2o: Fix typo in messages/i2o
scripts/kernel-doc: check that non-void fcts describe their return value
Kernel-doc: Convention: Use a "Return" section to describe return values
radeon: Fix typo and copy/paste error in comments
doc: Remove unnecessary declarations from Documentation/accounting/getdelays.c
various: Fix spelling of "asynchronous" in comments.
Fix misspellings of "whether" in comments.
eisa: Fix spelling of "asynchronous".
various: Fix spelling of "registered" in comments.
doc: fix quite a few typos within Documentation
target: iscsi: fix comment typos in target/iscsi drivers
treewide: fix typo of "suport" in various comments and Kconfig
treewide: fix typo of "suppport" in various comments
...
The only required change is to extend the existing Xonar U1
mixer quirks to the U3, which seems to be controlled the same
way.
Signed-off-by: Denis Washington <denisw@online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch below prevents the 6fire usb driver going into panic state
when stopping playing. On some systems the urb in handler
(usb6fire_pcm_in_urb_handler) is being called while urbs are being
killed off, this causes the driver to set panic state and can result in
the kernel warning 'URB %p submitted while active'.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 947d299686 , "ALSA: snd-usb:
properly initialize the sync endpoint", while correcting the
initialization of the sync endpoint when opening just the data
endpoint, prevents devices that has a sync endpoint, with a channel
number different than that of the data endpoint, from functioning.
Due to a different channel and period bytes count, attempting to
initialize the sync endpoint will fail at the usb host driver.
For example, when using xhci:
cannot submit urb 0, error -90: internal error
With this patch, if a sync endpoint has multiple audioformats, a
matching audioformat is preferred. An audioformat must be found
with at least one channel and support the requested sample rate
and PCM format, otherwise the stream will not be opened.
If the number of channels differ between the selected audioformat
and the requested format, adjust the period bytes count accordingly.
It is safe to perform the calculation on the basis of the channel
count, since the requested PCM audio format and the rate must be
supported by the selected audioformat.
Cc: Jeffrey Barish <jeff_barish@earthlink.net>
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [88a8516a: ALSA: usbaudio: implement USB autosuspend] added
the support of autopm for USB MIDI output, but it didn't take the MIDI
input into account.
This patch adds the following for fixing the autopm:
- Manage the URB start at the first MIDI input stream open, instead of
the time of instance creation
- Move autopm code to the common substream_open()
- Make snd_usbmidi_input_start/_stop() more robust and add the running
state check
Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a similar protection against the disconnection race and the
invalid use of usb instance after disconnection, as well as we've done
for the USB audio PCM.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51201
Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot of headsets/headphones have a "Speaker" mixer control. This confuses
PulseAudio to think it is a speaker instead of a headphone/headset.
Therfore, we rename it to "Headphone".
We determine if something is a headphone similar to how udev determines
form factor (see 78-sound-card.rules).
BugLink: https://bugs.launchpad.net/bugs/1082357
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The playback endpoint uses implicit feedback mode, similar
to the M-Audio FTU. Like with the FTU, we need to associate
the sync pipe ourselves.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ranges for various Fast Track C400 controls, as observed
while using the vendor's mixer control software (res values
are an estimation).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds the unit ID and the control as parameters to the creation of the
effect unit control for the M-Audio Fast Track Ultra. This allows the
code to be shared with other devices that use different unit ID and
control, such as the M-Audio Fast Track C400.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code mishandles the case where the device is a UAC2
and the bDescriptorSubtype is a UAC2 Effect Unit (0x07).
It tries to parse it as a Processing Unit (which is similar to two
other UAC1 units with overlapping subtypes), but since the structure
is different (See: 4.7.2.10, 4.7.2.11 in UAC2 standard), the parsing
is done incorrectly and prevents the device from initializing.
For now, just ignore the unit.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, channel IDs exceeding 31 (0x1f) cannot be used.
The channel ID is derived from the cmask. Extending cmask
to a 64-bit type would only allow it to go up to 63 (0x3f).
Some devices have channel IDs exceeding that as well.
To address that, add an offset to the mixer element which
is then accounted for in the UAC set/get functions.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For implicit feedback endpoints, the number of bytes for each packet
is matched by the corresponding synchronizing endpoint.
The size is calculated by taking the actual size and dividing it by
the stride - currently by the endpoint's stride, but we should use the
synchronization source's stride.
This is evident when the number of channels differ between the
synchronization source and the implicitly fed-back endpoint, as with
M-Audio Fast Track C400 - the synchronization source (capture)
has 4 channels, while the implicit feedback mode endpoint has 6.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support for channel maps of the PCM streams on USB audio
devices. The channel map information is already found in
ChannelConfig descriptor entries, which haven't been referred until
now.
Each chmap entry is added to audioformat list entry and copied to TLV
dynamically instead of creating a whole chmap array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a playback stream is paused, the stream isn't actually stopped,
thus we still need to take care of the in-flight data amount for the
delay calculation. Otherwise the value of subs->last_delay is no
longer reliable and can give a bogus value after resuming from pause.
This will result in "delay: estimated XX, actual YY" error messages.
Also, during pause after all in flight data are processed
(i.e. last_delay = 0), we don't have to calculate the actual delay
from the current frame. Give a short path in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It doesn't make sense to calculate the delay for capture streams in
the current implementation. It's always zero, so we should skip the
computation in snd_usb_pcm_pointer() in the case of capture.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio
driver which causes the code to not initialize the sync endpoint from
configure_endpoint().
Reported-by: Jeffrey Barish <jeff_barish@earthlink.net>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM hw_free and close should wait until all the pending stop
operations have been finished. Basically only PCM trigger callback
should use non-wait calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.
So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop(). (Actually there is only one
place calling this, so it was safe to change.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For further code simplification, drop the conditional call for
usb_kill_urb() with can_wait argument in deactivate_urbs(), and use
only usb_unlink_urb() and wait_clear_urbs() pairs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop(). Also replaced from int to bool.
No functional changes by this commit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The async unlink behavior has been working over years. The option was
provided only as a workaround for 2.4.x kernel. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'rt' was dereferenced before the NULL check.
Moved the code after the check.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Whether" is misspelled in various comments across the tree; this
fixes them. No code changes.
Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend) added
autosuspend code to all files making up the snd-usb-audio driver.
However, midi.c is part of snd-usb-lib and is also used by other
drivers, not all of which support autosuspend. Thus, calls to
usb_autopm_get_interface() could fail, and this unexpected error would
result in the MIDI output being completely unusable.
Make it work by ignoring the error that is expected with drivers that do
not support autosuspend.
Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com>
Reported-by: Devin Venable <venable.devin@gmail.com>
Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk>
Reported-by: Jannis Achstetter <jannis_achstetter@web.de>
Reported-by: Rui Nuno Capela <rncbc@rncbc.org>
Cc: Oliver Neukum <oliver@neukum.org>
Cc: 2.6.39+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Use bitmap_weight to count the total number of bits set in bitmap.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again. There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:
A. snd_usb_audio_disconnect() ->
card.c::register_mutex ->
chip->shutdown_rwsem (write) ->
snd_card_disconnect() ->
pcm.c::register_mutex ->
pcm->open_mutex
B. snd_pcm_open() ->
pcm->open_mutex ->
snd_usb_pcm_open() ->
chip->shutdown_rwsem (read)
Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails. This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():
usbaudio: unit 51: unexpected type 0x09
snd-usb-audio: probe of 1-8:1.0 failed with error -5
Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out. It would be nice to
avoid that.
While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces. This allows audio and MIDI interfaces to be
used without any obvious issues.
Interface 0 is the main one to ignore. It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this. Interface 3 is some sort of control interface that
I don't understand:
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 3
bAlternateSetting 0
bNumEndpoints 0
bInterfaceClass 1 Audio
bInterfaceSubClass 1 Control Device
bInterfaceProtocol 0
iInterface 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 1 (HEADER)
bcdADC 1.00
wTotalLength 9
bInCollection 1
baInterfaceNr( 0) 1
Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.
Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.
The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
mixer.c and others; the device speed is now cached in subs->speed
instead of accessing to chip->dev
The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.
The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks. They'll be covered by the
upcoming change to rwsem.
Also the mixer codes are untouched, too. These will be fixed in
another patch, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and
UAC_EP_CS_ATTR_SAMPLE_RATE attribute.
Signed-off-by: Didier Villevalois <ptitjes@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this quirk the VG-99 will work in standard mode (set under
USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out
but no midi, and is not recognised when set to advanced mode.
After applying this, I can also use the VG-99 in advanced mode: 24
24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far
untested.
In standard mode, the device appears with ID 0x00b3, so the
behaviour isn't affected by this quirk.
Thanks to Clemens Ladisch for simplifying and correcting my initial
attempt!
Signed-off-by: Pete Leigh <pete.leigh@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable ep is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This update contains a few cleanup works, regression/stable fixes
gathered since the last pull request.
- Clean up with generic hd-audio jack handling code by David
Henningsson
- A few regression fixes for standardized HD-audio auto-parser
- Misc clean-up and small fixes
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates #2 from Takashi Iwai:
"This update contains a few cleanup works, regression/stable fixes
gathered since the last pull request.
- Clean up with generic hd-audio jack handling code by David
Henningsson
- A few regression fixes for standardized HD-audio auto-parser
- Misc clean-up and small fixes"
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - do not detect jack on internal speakers for Realtek
ALSA: hda - Fix missing beep on ASUS X43U notebook
ALSA: hda - Remove AZX_DCAPS_POSFIX_COMBO
ALSA: hda - Warn an allocation for an uninitialized array
ALSA: hda/cirrus - Add missing init/free of hda_gen_spec
ALSA: hda - Fix memory leaks at error path in patch_cirrus.c
ALSA: hda - Add missing hda_gen_spec to struct via_spec
ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30)
ALSA: hda - make Cirrus codec use generic unsol event handler
ALSA: hda - make VIA codec use generic unsol event handler
ALSA: hda - Remove dead GPIO code for VIA codec
ALSA: usb-audio: Add TASCAM US122 MKII playback
A long time ago, in v2.4, VM_RESERVED kept swapout process off VMA,
currently it lost original meaning but still has some effects:
| effect | alternative flags
-+------------------------+---------------------------------------------
1| account as reserved_vm | VM_IO
2| skip in core dump | VM_IO, VM_DONTDUMP
3| do not merge or expand | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP
4| do not mlock | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP
This patch removes reserved_vm counter from mm_struct. Seems like nobody
cares about it, it does not exported into userspace directly, it only
reduces total_vm showed in proc.
Thus VM_RESERVED can be replaced with VM_IO or pair VM_DONTEXPAND | VM_DONTDUMP.
remap_pfn_range() and io_remap_pfn_range() set VM_IO|VM_DONTEXPAND|VM_DONTDUMP.
remap_vmalloc_range() set VM_DONTEXPAND | VM_DONTDUMP.
[akpm@linux-foundation.org: drivers/vfio/pci/vfio_pci.c fixup]
Signed-off-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Cc: Alexander Viro <viro@zeniv.linux.org.uk>
Cc: Carsten Otte <cotte@de.ibm.com>
Cc: Chris Metcalf <cmetcalf@tilera.com>
Cc: Cyrill Gorcunov <gorcunov@openvz.org>
Cc: Eric Paris <eparis@redhat.com>
Cc: H. Peter Anvin <hpa@zytor.com>
Cc: Hugh Dickins <hughd@google.com>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: James Morris <james.l.morris@oracle.com>
Cc: Jason Baron <jbaron@redhat.com>
Cc: Kentaro Takeda <takedakn@nttdata.co.jp>
Cc: Matt Helsley <matthltc@us.ibm.com>
Cc: Nick Piggin <npiggin@kernel.dk>
Cc: Oleg Nesterov <oleg@redhat.com>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: Robert Richter <robert.richter@amd.com>
Cc: Suresh Siddha <suresh.b.siddha@intel.com>
Cc: Tetsuo Handa <penguin-kernel@I-love.SAKURA.ne.jp>
Cc: Venkatesh Pallipadi <venki@google.com>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Also fix the calls to next_packet_size() for the pause case. This was
missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size").
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de>
Cc: stable@kernel.org
[ Taking directly because Takashi is on vacation - Linus ]
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
While going through Ubuntu bugs, I discovered this patch being
posted and a confirmation that the patch works as expected.
Finding out how the hw volume really works would be preferrable
to just disabling the broken one, but this would be better than
nothing.
Credit: sndfnsdfin (qawsnews)
BugLink: https://bugs.launchpad.net/bugs/559939
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for USB suspend breakage moved the code to set up EP
from hw_params to prepare, but it means also the EP setup might be
called multiple times unnecessarily because the prepare callback can
be called multiple times without starting the stream (e.g. OSS
emulation).
This patch adds a new flag to struct snd_usb_substream indicating
whether the setup of EP is required, and do it only when necessary,
i.e. right after hw_params or suspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move interface and endpoint configuration from hw_params to prepare
callback. During system suspend/resume when the USB device power isn't
cycled the interface and endpoint configuration need to be set before
audio playback can continue. Resume involves another call to prepare
but not to hw_params, moving it here allows a playing stream to continue
after resume.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the interface to configure an endpoint so that it doesn't require
a hw_params struct. This will allow it to be called from prepare
instead of hw_params, configuring it after system resume.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the peiod_bytes member of snd_usb_substream. It was no longer being
set, but will be needed to resume properly in a future commit.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
delay: estimated 0, actual 352
delay: estimated 353, actual 705
These come from the sanity check in retire_playback_urb(). Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent. And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.
For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.
Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:
- They need a 20ms delay after each class compliant request as the
hardware ACKs the USB packets before the device is actually ready
for the next command. Sending data immediately will result in buffer
overflows in the hardware.
- The devices send bogus feedback data at the start of each stream
which confuse the feedback format auto-detection.
This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.
In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.
Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.
However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.
As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.
Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.
This patch adds them back, restoring the correct delay information
behaviour.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_endpoint_free() frees the structure that contains its argument.
Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.
Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A PCM capture stream on usb-audio causes a scheduling-while-atomic
BUG, as reported in the bugzilla entry below. It's because
snd_usb_endpoint_start() is called at first at trigger START for a
capture stream, and this function contains the left-over EP
deactivation codes. The problem doesn't happen for a playback stream
because the function is called at PCM prepare time, which can sleep.
This patch fixes the BUG by moving the EP deactivation code into the
PCM prepare callback.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
uac_clock_source_is_valid() uses the control selector value to access
the bmControls bitmap of the clock source unit. This is wrong, as
control selector values start from 1, while the bitmap uses all
available bits.
In other words, "Clock Validity Control" is stored in D3..2, not D5..4
of the clock selector unit's bmControls.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The rework of the snd-usb endpoint logic moved the calls to
snd_usb_set_interface() into the snd_usb_endpoint implemenation. This
changed the order in which these calls are issued to the device, and
thereby caused regressions for some webcams.
Fix this by moving the calls back to pcm.c for now to make it work again
and use snd_usb_endpoint_activate() to really tear down all remaining
URBs in the flight, consequently fixing another regression caused by USB
packets on the wire after altsetting 0 has been selected.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net>
Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the new PCM streaming logic, the interface number is assigned to
usb stream instance (subs->interface) after the format and rate setups
are succeeded, but some codes are still passing subs->interface as the
reference to helper functions. This leads to initializing with an
invalid iface number (-1).
This patch replaces the wrong references with the ones from the target
fmt correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the following warning:
CC [M] sound/usb/caiaq/device.o
sound/usb/caiaq/device.c: In function ‘snd_probe’:
sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used
uninitialized in this function [-Wmaybe-uninitialized]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This webcam works mostly ok, exept with skype.
Skype sends lots of ctrl messages to dynamically ajust
record level. If for some reasons it pokes some error
every thing goes broken:
- first pulseaudio blocks sound for all apps
- then video is reseted
- then skype freez
dmesg has lots of messages like:
cannot set freq 16000 to ep 0x86"
Setting ignore_ctl_error=1 fixes this problem.
Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if this cam is pluged in, pulse audio can't initiate capture
device.
dmesg has lots of messages like:
"cannot set freq 16000 to ep 0x86"
Setting ignore_ctl_error=1 fixes this problem.
Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a bogus sanity check for sync pipe in pcm.c. This flaw was
introduced during the streaming logic refactorization.
While at it, improve the error messages that are generated in such cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: <ben@b1c1l1.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow mixer controls to be provided clearly in a table, to avoid
quantity of error checking at each use.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In 3.5 kernel, the endpoint is assigned dynamically for the
substreams, but the PCM assignment still checks the presence of the
endpoint pointer. This ended up in duplicated PCM substream creations
at probing time, resulting in kernel warnings like:
WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6()
Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2
Call Trace:
[<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c
[<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48
[<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b
[<ffffffff811292f0>] proc_register+0x169/0x1a6
[<ffffffff8112962e>] create_proc_entry+0x74/0x8c
[<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd]
[<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm]
[<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio]
[<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio]
[<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb
[<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio]
[<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio]
[<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio]
.....
This patch fixes the regression by checking the fixed endpoint number
for each substream instead of the endpoint pointer.
Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array of sample rates is reallocated every time when opening
the PCM device, but was freed only once when unplugging the device.
Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set some substream struct members to make the proc interface code work
again.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.
Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>
> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized. A short cut for booleans.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.
[fixed missing break by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.
This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ep->fill_max is a 1 bit flag, thus it has to be boolean.
sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Document the new streaming code and some of the functions so that
contributers can catch up easier.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new generic streaming logic for audio over USB.
It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.
A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.
With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.
In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a stereo volume control for analog input channel pair 1/2.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mute control for every analog output channel.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the soft log-conversion and add a dB scale according to
the DAC documentation instead.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused driver version information from the individual files.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 3702b08 added a lock, but did not account for the case of
SNDRV_PCM_POS_XRUN, which would get immediately overwritten.
This could be bundled into one if-else-if statement, but the goto
helps to clarify the 'exceptional' case.
Thanks to Andreas Pape for spotting this.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A malicious USB device could feed in a large nr_rates value. This would
cause the subsequent call to kmemdup() to allocate a smaller buffer than
expected, leading to out-of-bounds access.
This patch validates the nr_rates value and reuses the limit introduced
in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow
in parse_uac2_sample_rate_range()").
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling "propably" to "probably" and "activ" to "active"
in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
ALSA: usb-audio: add Yamaha MOX6/MOX8 support
ALSA: virtuoso: add S/PDIF input support for all Xonars
ALSA: ice1724 - Support for ooAoo SQ210a
ALSA: ice1724 - Allow card info based on model only
ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
ALSA: hdspm - Provide unique driver id based on card serial
ASoC: Dynamically allocate the rtd device for a non-empty release()
ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
ALSA: hda/cirrus - support for iMac12,2 model
ASoC: cx20442: add bias control over a platform provided regulator
ALSA: usb-audio - Avoid flood of frame-active debug messages
ALSA: snd-usb-us122l: Delete calls to preempt_disable
mfd: Put WM8994 into cache only mode when suspending
...
Fix up trivial conflicts in:
- arch/arm/mach-s3c64xx/mach-crag6410.c:
renamed speyside_wm8962 to tobermory, added littlemill right
next to it
- drivers/base/regmap/{regcache.c,regmap.c}:
duplicate diff that had already come in with other changes in
the regmap tree
With some buggy devices, the usb-audio driver may give "frame xxx active"
kernel messages too often. Better to keep it as debug-only using
snd_printdd(), and also add the rate-limit for avoiding floods.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A malicious USB device may feed in carefully crafted min/max/res values,
so that the inner loop in parse_uac2_sample_rate_range() could run for
a long time or even never terminate, e.g., given max = INT_MAX.
Also nr_rates could be a large integer, which causes an integer overflow
in the subsequent call to kmalloc() in parse_audio_format_rates_v2().
Thus, kmalloc() would allocate a smaller buffer than expected, leading
to a memory corruption.
To exploit the two vulnerabilities, an attacker needs physical access
to the machine to plug in a malicious USB device.
This patch makes two changes.
1) The type of "rate" is changed to unsigned int, so that the loop could
stop once "rate" is larger than INT_MAX.
2) Limit nr_rates to 1024.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This resolves the conflict in the arch/arm/mach-s3c64xx/s3c6400.c file,
and it fixes the build error in the arch/x86/kernel/microcode_core.c
file, that the merge did not catch.
The microcode_core.c patch was provided by Stephen Rothwell
<sfr@canb.auug.org.au> who was invaluable in the merge issues involved
with the large sysdev removal process in the driver-core tree.
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
module_param(bool) used to counter-intuitively take an int. In
fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy
trick.
It's time to remove the int/unsigned int option. For this version
it'll simply give a warning, but it'll break next kernel version.
Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon
Roland SH-201 table entry as template. USB MIDI and audio was tested
with Muse and Audacity.
Signed-off-by: John F Leach <jfleach@jfleach.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This converts the drivers in sound/* to use the
module_usb_driver() macro which makes the code smaller and a bit
simpler.
Added bonus is that it removes some unneeded kernel log messages about
drivers loading and/or unloading.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Cc: Paul Gortmaker <paul.gortmaker@windriver.com>
Cc: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Use kmemdup rather than duplicating its implementation
The semantic patch that makes this change is available
in scripts/coccinelle/api/memdup.cocci.
Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent usb-audio driver, the initialization of volume ranges
may be delayed when the device doesn't respond well at the probing time.
But the volume quirks for certain devices are applied only in
mixer_ctl_feature_info() thus only at the very first probe and will be
missing when the volume range is initialized later.
This patch moves the volume quirk code to be always called from the
volume-range extraction (get_min_max()), so that the quirks are properly
applied in the later init time.
Reported-and-tested-by: Alexey Fisher <bug-track@fisher-privat.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux: (230 commits)
Revert "tracing: Include module.h in define_trace.h"
irq: don't put module.h into irq.h for tracking irqgen modules.
bluetooth: macroize two small inlines to avoid module.h
ip_vs.h: fix implicit use of module_get/module_put from module.h
nf_conntrack.h: fix up fallout from implicit moduleparam.h presence
include: replace linux/module.h with "struct module" wherever possible
include: convert various register fcns to macros to avoid include chaining
crypto.h: remove unused crypto_tfm_alg_modname() inline
uwb.h: fix implicit use of asm/page.h for PAGE_SIZE
pm_runtime.h: explicitly requires notifier.h
linux/dmaengine.h: fix implicit use of bitmap.h and asm/page.h
miscdevice.h: fix up implicit use of lists and types
stop_machine.h: fix implicit use of smp.h for smp_processor_id
of: fix implicit use of errno.h in include/linux/of.h
of_platform.h: delete needless include <linux/module.h>
acpi: remove module.h include from platform/aclinux.h
miscdevice.h: delete unnecessary inclusion of module.h
device_cgroup.h: delete needless include <linux/module.h>
net: sch_generic remove redundant use of <linux/module.h>
net: inet_timewait_sock doesnt need <linux/module.h>
...
Fix up trivial conflicts (other header files, and removal of the ab3550 mfd driver) in
- drivers/media/dvb/frontends/dibx000_common.c
- drivers/media/video/{mt9m111.c,ov6650.c}
- drivers/mfd/ab3550-core.c
- include/linux/dmaengine.h
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Revert the check of NO_PRESENCE pincfg default bit
ALSA: hda - Fix a regression for DMA-position check with CA0110
ALSA: hda - Fix silent output regression with ALC861
ALSA: control: remove compilation warning on 32-bit
ALSA: ua101: fix crash when unplugging
If the device is unplugged while running, it is possible for a PCM
device to be closed after the disconnect callback has returned. This
means that kill_stream_urb() and disable_iso_interface() would try to
access already-invalid or freed USB data structures.
The function free_usb_related_resources() was intended to prevent this,
but forgot to clear the affected variables.
Reported-and-tested-by: Olivier Courtay <olivier@courtay.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.33+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots of sound drivers were getting module.h via the implicit presence
of it in <linux/device.h> but we are going to clean that up. So
fix up those users now.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
These files were getting access to these two via the implicit
presence of moduleparam.h everywhere. But that is being fixed, so
get these guys what they need in advance.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
The audio_feature_info[] array should contain all entries for UAC2_FU_*,
but currently a few last entries are missing. Even though, the driver
tries to probe these entries in parse_audio_feature_unit() and may
access the range over the array. This patch fixes the bug by limiting
the loop size properly using ARRAY_SIZE() instead of a hard-coded
magic number.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds partial support for the Maschine controller by Native Instruments.
Supported now are the 1x1 MIDI interface and the 41 buttons, 11 endless
rotary encoders, and 16 pressure-sensitive drum pads. Still to work on are the
dimmable LEDs and the two monochrome screens.
Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a case where a newly-registered input device could be opened before
a necessary variable in the device structure was set. When code tried to use
the variable in the URB reply callback, it would cause an Oops.
This fix sets the aforementioned variable before calling input_register_device.
Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* pm-runtime:
PM / Tracing: build rpm-traces.c only if CONFIG_PM_RUNTIME is set
PM / Runtime: Replace dev_dbg() with trace_rpm_*()
PM / Runtime: Introduce trace points for tracing rpm_* functions
PM / Runtime: Don't run callbacks under lock for power.irq_safe set
USB: Add wakeup info to debugging messages
PM / Runtime: pm_runtime_idle() can be called in atomic context
PM / Runtime: Add macro to test for runtime PM events
PM / Runtime: Add might_sleep() to runtime PM functions
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers. Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.
The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.
Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before clearing the probing flag in the error exit path, check that the
chip pointer is not NULL.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Terratec Aureon 5.1 USB sound card support is broken since kernel
2.6.39.
2.6.39 introduced power management support for USB sound cards that added
a probing flag in struct snd_usb_audio.
During the probe of the card it gives following error message :
usb 7-2: new full speed USB device number 2 using uhci_hcd
cannot find UAC_HEADER
snd-usb-audio: probe of 7-2:1.3 failed with error -5
input: USB Audio as
/devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6
generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio]
on usb-0000:00:1d.1-2/input3
I can not comment about that "cannot find UAC_HEADER" error, but until
2.6.38 the card worked anyway.
With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl
stops in snd_usb_autoresume with -ENODEV.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Roland UM-ONE midi usb interface differs from Roland UM-1.
Signed-off-by: Daniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No code altered at this point, simply preparing for upcoming
refactorizations.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move code from endpoint.c into a new file called stream.c and rename
functions so that their names actually reflect what they're doing.
This way, endpoint.c will be available to functions that hold all the
endpoint logic.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sort its entries in alphabetical order.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.
This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.
Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for Starr Labs USB MIDI devices such as the Z7S, which are
based on an FTDI serial UART chip.
Based on a patch by Daniel Mack.
Signed-off-by: Kristian Amlie <kristian@amlie.name>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch (as1482) adds a macro for testing whether or not a
pm_message value represents an autosuspend or autoresume (i.e., a
runtime PM) event. Encapsulating this notion seems preferable to
open-coding the test all over the place.
Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
When the initial check of dB-range failed due to the read error, try to
check again at the later read, too. When an invalid dB range is found,
remove TLV flags and notify the mixer info change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for testing dB range at the mixer creation time seems
to cause regressions in some devices. In such devices, reading the dB
info at probing time gives an error, thus both dBmin and dBmax are still
zero, and TLV flag isn't set although the later read of dB info succeeds.
This patch adds a workaround for such a case by assuming that the later
read will succeed. In future, a similar test should be performed in a
case where a wrong dB range is seen even in the later read.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The snd_usb_caiaq driver currently assumes that output urbs are serviced
in time and doesn't track when and whether they are given back by the
USB core. That usually works fine, but due to temporary limitations of
the XHCI stack, we faced that urbs were submitted more than once with
this approach.
As it's no good practice to fire and forget urbs anyway, this patch
introduces a proper bit mask to track which requests have been submitted
and given back.
That alone however doesn't make the driver work in case the host
controller is broken and doesn't give back urbs at all, and the output
stream will stop once all pre-allocated output urbs are consumed. But
it does prevent crashes of the controller stack in such cases.
See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Matej Laitl <matej@laitl.cz>
Cc: Sarah Sharp <sarah.a.sharp@linux.intel.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes faulty outbount packets in case the inbound packets
received from the hardware are fragmented and contain bogus input
iso frames. The bug has been there for ages, but for some strange
reasons, it was only triggered by newer machines in 64bit mode.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: William Light <wrl@illest.net>
Reported-by: Pedro Ribeiro <pedrib@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Purely cosmetic, but fixes the following build warning.
CC [M] sound/usb/quirks.o
sound/usb/quirks.c: In function ‘snd_usb_apply_boot_quirk’:
sound/usb/quirks.c:429:6: warning: ‘err’ may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When creating the mixers for an USB audio device, the current code looks
at the host interface stored in mixer->chip->ctrl_if. Change this and
rather keep a local pointer to the interface that was given when
snd_usb_create_mixer() was called.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Reported-by: Lean-Yves LENHOF <jean-yves@lenhof.eu.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Focusrite Scarlett 18i6 USB has them that way, which is probably a
bug. Anyway, the driver should simply ignore this fact.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.
This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.
Signed-off-by: David G Turner <dgturner@iee.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.
Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :
options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08
Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf
Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.
Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the freshly introduced methods to re-use standard mixer
handling and add some controls that are hidden but implemented in a
standard conform way on M-Audio's FastTrack devices.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Original-code-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk type will let the driver assume that there is a standard
mixer on a given interface, or that a specific mixer quirks will handle
the device.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to allow quirks functions to hook up to the standard feature
unit op tables, this patch exports a pointer to the struct that is used
internally.
That way, all the code handling the control can be kept private, and
external code can reference the symbol to re-use it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames add_control_to_empty() to snd_usb_mixer_add_control()
and exports it, so the quirks functions can make use of it.
Also, as "struct mixer_build" is private to mixer.c, rewrite the
function to take an argument of type "struct usb_mixer_interface"
instead.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB X-Fi S51 Pro volume and mute from the volume knob on the unit.
Compiled and tested with 2.6.39-rc7-git12
Signed-off-by: Mathieu Bouffard <mbouffard@strangequarks.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
That way, the class compliant MIDI interface is also handled.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Grant Diffey <gdiffey@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the interface can't report a clock's validity, assume that it's
valid.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Vicente Joel <vicentejoel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was a flaw in the reading of the spec tables - Native Instrument's
"Komplete Audio 6" device has no such extra controls.
This patch also fixes the device name in two comments.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just in case a prototype changes, we'll be warned. This also fixes a
sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just defining it to nothing is dangerous as it can alter the code
execution flow, for example when used in as only function in a
conditional code block.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Terratec Aureon 7.1 USB which uses a
C-Media cm6206 and needs all the quirks already found in the past.
Signed-off-by: Wolfgang Breyha <wbreyha@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some crappy USB-audio devices give broken dB ranges, e.g. both min and max
are 0dB. This confuses the volume control that prefers dB expression such
as alsactl or PulseAudio. In such a case, it's much better not to expose
the broken dB information.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new device by Native Instruments is also compliant to the USB
standard v2.0, but hides this detail at when connected.
It needs the same boot quirks than other models, and also has two
non-class-compliant mixer controls.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't query connections for widgets have no connections
ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
ALSA: HDA: Fix dock mic for Lenovo X220-tablet
ASoC: format_register_str: Don't clip register values
ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
ASoC: zylonite: set .codec_dai_name in initializer
There are many USB MIDI cables out there that have buggy
firmware that reports it can do more than 4 bytes in a
packet when they can only properly handle 4
This patch adds the ID of yet another one of those cables
Signed-off-by: Tarek Soliman <tarek@bashasoliman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digital Thru mixer element added (device can act as converter optical<->coax)
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Firmware loader: magical device bytes check updated (accepts all device
versions now and accepts possibly loaded firmware, if it is knowing to
be working)
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Completion of signedness bug for pcm_runtime.rate: variable will never
get assigned a negative value now.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the Cakewalk UM-1G USB MIDI interface in
"advanced driver" mode. (It already works in standard mode.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some USB devices give trailing spaces in strings returned from
usb_string(). This confuses the automatic card-id creation, resulting
always in "default".
This patch fixes the behavior by removing trailing spaces.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Devices are autosuspended if no pcm nor midi channel is open
Mixer devices may be opened. This way they are active when
in use to play or record sound, but can be suspended while
users have a mixer application running.
[Small clean-ups using static inline by tiwai]
Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ESHUTDOWN must be correctly handled
- the optional interrupt endpoint's URB must be stopped and restarted
Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more affected devices: Logitech Webcam C600 (046d:0808)
Volume range before quirk is 6400, after (also real) is 16.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB audio device is disconnected, snd_usb_audio_disconnect()
kills all audio URBs. At the same time, the application, after being
notified of the disconnection, might close the device, in which case
ALSA calls the .hw_free callback, which should free the URBs too.
Commit de1b8b93a0 "[ALSA] Fix hang-up at disconnection of usb-audio"
prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that
resulted from this race, but this introduced another race because the
URB callbacks could now be executed after snd_usb_hw_free() has
returned, and try to access already freed data.
Fix the first race by introducing a mutex to serialize the disconnect
callback and all PCM callbacks that manage URBs (hw_free and hw_params).
Reported-and-tested-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Cc: <stable@kernel.org>
[CL: also serialize hw_params callback]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use strlcpy() to assure not to overflow the string array sizes by
too long USB device name string.
Reported-by: Rafa <rafa@mwrinfosecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of cases has increased so use switch-case rather than
if-statements.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MK2 generation of Native Instruments' sound cards are in fact
compliant to the USB audio standard of version 2 and other approved USB
standards. However, they come up as vendor-specific device when first
connected but can be told to come up with a new set of descriptors
upon their next enumeration. The interfaces announced by the new
descriptors will be handled by the kernel's class drivers. This is done
by issuing a vendor specific device request and sending the device to
reset.
There are also some vendor-specific USB requests for some mixer elements
that can't be exported in a standard compliant way. The driver now
supports them with quirks handling mechanisms.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
rt->rate is an unsigned char so it's never equal to -1. It's not a huge
problem because the invalid rate is caught inside the call to
usb6fire_pcm_set_rate() which returns -EINVAL. But if we fix the test
then it prints out the correct error message so that's good.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl().
Wake up poll waiters before returning from usb_stream_hwdep_ioctl().
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The US-122L always reads 9 bytes per urb unless they are set to 0xFD.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the 24-bit audio I/Os of the Edirol SD-90 interface.
Reported-any-tested-by: Jim Grusendorf <alsa-user@grusendorf.ca>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for Power/Status LED on Creative USB X-Fi S51.
There is just one LED on the device. The LED can either be On or it
can be set to Blink. There doesn't seem to be a way to switch it off.
The control message to change LED status is similar to that of
audigy2nx except that the index is to be set to 0 and value is 1 for
Blink and 0 for On.
The 'Power LED' control in alsamixer when muted will cause the LED to
Blink continuously. When unmuted the LED will stay On. The Creative
driver under Windows sets the LED to blink whenever audio is muted.
This LED can be treated as the CMSS LED but I figured since there is
just one LED, it should be treated as the Power LED. Is that alright?
I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there
are other external X-Fi devices from Creative like Usb X-Fi Go and
Xmod. The volume knob and LED support patch doesn't apply to them.
Signed-off-by: Mandar Joshi <emailmandar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value
from kmalloc() and may end up dereferencing a null pointer.
The patch below (compile tested only) should take care of that little
problem.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two USB Audio Class specifications (v1 and v2), but neither of
them clearly defines the feedback format for high-speed UAC v1 devices.
Add to this whatever the Creative and M-Audio firmware writers have been
smoking, and it becomes impossible to predict the exact feedback format
used by a particular device.
Therefore, automatically detect the feedback format by looking at the
magnitude of the first received feedback value.
Also, this allows us to get rid of some special cases for E-Mu devices.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
The action of the volume knob is received by lirc when its using the
alsa_usb driver.
Signed-off-by: Mandar Joshi <emailmandar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk entry for the Novation Launchpad USB MIDI controller.
QUIRK_MIDI_FASTLANE gets renamed to *_RAW_BYTES because this quirk type
is now shared by different devices.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Jakob Flierl <jakob.flierl@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirks for more devices (according to driver V.3.0.4-2).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"uinfo->value.enumerated.item" is an unsigned int. If it's negative
when we do the comparison:
if ((int)uinfo->value.enumerated.item >= cval->max)
then we would read past the end of the array on the next line.
I also changed the strcpy() to strlcpy() out of paranoia.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used only when CONFIG_SND_DEBUG=y
sound/usb/mixer.c: In function 'get_min_max':
sound/usb/mixer.c:762: warning: unused variable 'chip'
Reported-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the new Traktor Kontrol S4 by Native
Instruments. It features a new audio data streaming model, MIDI
in and out ports, a huge number of 174 dimmable LEDs, 96 buttons
and 46 absolute encoder axis, including some rotary encoders.
All features are supported by the driver now.
Did some code refactoring along the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the wrong "return" in the loop, a capture substream won't be
released at disconnection properly if the device is capture only and has
no playback substream. This caused Oops occasionally at the device
reconnection.
Reported-by: Kim Minhyoung <minhyoung.kim@lge.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Audio Class v2 support code in 2.6.35 added checks for the
bInterfaceProtocol field. However, there are devices (usually those
detected by vendor-specific quirks) that do not have one of the
predefined values in this field, which made the driver reject them.
To fix this regression, restore the old behaviour, i.e., assume that
a device with an unknown bInterfaceProtocol field (other than
UAC_VERSION_2) has more or less UAC-v1-compatible descriptors.
[compile warning fixes by tiwai]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk to make the BOSS ME-25 work.
Many thanks to Kees van Veen.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the Roland/Cakewalk A-300PRO/A-500PRO/A-800PRO keyboards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the other logical device of the PCR-1 so that not only
the MIDI interface but also the audio interface works.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For devices with more than one control interface, let's assume the first
one contains the audio controls. Unfortunately, there is no field in any
of the descriptors to tell us whether a control interface is for audio
or MIDI controls, so a better check is not easy to implement.
On a composite device with audio and MIDI functions, for example, the
code currently overwrites chip->ctrl_intf, causing operations on the
control interface to fail if they are issued after the device probe.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Fast Track Ultra series devices did not play sound correctly
at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive
fixes this.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: sound/usb/format: silence uninitialized variable warnings
MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices
MAINTAINERS: Make Wolfson entry also cover CODEC drivers
ASoC: Only tweak WM8994 chip configuration on devices up to rev D
ASoC: Optimise DSP performance for WM8994
ALSA: hda - Fix dynamic ADC change working again
ALSA: hda - Restrict PCM parameters per ELD information over HDMI
sound: oss: sh_dac_audio.c removed duplicated #include
Gcc complains that ret might be used uninitialized:
sound/usb/format.c: In function ‘snd_usb_parse_audio_format’:
sound/usb/format.c:354: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:354: note: ‘ret’ was declared here
sound/usb/format.c:414: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:414: note: ‘ret’ was declared here
I suppose it could be uninitialized if there is ever a UAC_VERSION_3
released. Anyway this patch is worthwhile if only to silence the gcc
warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is V2 of the patch, after feedback from Clemens and Daniel.
This patch adds SuperSpeed support to the USB drivers under sound/. It adds
tests for USB_SPEED_SUPER to the appropriate places that check for the USB
speed.
This patch has been tested with our SS USB3 device emulating a set of Yamaha
speakers and a Logitech microphone, but with the descriptors modified to add
USB3 support. It has also been tested with the real speakers and microphone,
to make sure that USB2 devices still work.
Signed-off-by: Paul Zimmerman <paulz@synopsys.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Match usb ids in usb/quirks-table.h for some Hauppage HVR-950Q models
and for the HVR850 model to those ids at the end of au0828-cards.c
Thanks to nhJm449 for pointing out the problem.
Signed-off-by: John S Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the control interface is now carried in struct snd_usb_audio, we can
simplify the API a little and also drop the private ctrlif field from
struct usb_mixer_interface.
Also remove a left-over function prototype in pcm.h.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Get rid of the last occurances of _v1 suffixes, and move the version
number right after the "uac" string. Now things are consitent again.
Sorry for the forth and back, but it just looks much nicer this way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some programs like Skype trying to set capture volume automatically.
Normally it will tray, carefully step by step lover or higher, set the volume.
In real word it work not really well, because devises and vendors lie about
real audio settings.
For example most Logitech webcams have 6400 or 3500 steps for capture volume.
They do not tell that actual resolution is 384. So we have only 7 or 18 real
steps. In this patch I set real resolution only for tested devices.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanse found that in snd_usb_parse_audio_endpoints, there is a
dangling pointer dereference. When snd_usb_parse_audio_format fails,
fp is freed, and continue invoked. On the next loop, there is
"fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set
from the last iteration (but is bogus) and thus ilegally dereferenced.
Set fp to NULL before "continue".
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For RANGE requests, we should only query as much bytes as we're in fact
interested in.
For CUR requests, we shouldn't confuse the firmware with an overlong
request but just ask for 2 bytes.
This might need fixing in the future as it's not entirely clear when to
dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume
everything is coded in 16bit - this works for all firmware
implementations I've seen.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A device may report its supported sample rates in ranges rather than in
discrete triplets. The code used to only parse the MIN field instead of
properly paying attention to the MAX and RES values.
Also, handle RES values of 1 correctly and announce a continous sample
rate range in this case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Control messages directed to an interface must have the interface number
set in the lower 8 bits of wIndex. This wasn't done correctly for some
clock and mixer messages.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2 clock selectors are fortunately compatible with UAC1 audio
selector units, so we can simply reuse the same approach to get all the
linked units.
Requests to this control need a different CS value though.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a struct to parse the audio units, and return usable descriptors
for all types. There's no need to limit the result set, except for some
kind of sanity check.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bits to enable them are always 0 for UAC1 devices, so no additional
checks are required.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move more definitions from private enums to appropriate header files.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio devices which comply to the UAC2 standard can export complex clock
topologies in its descriptors and set up links between them.
The entities that are defined are
- clock sources, which define the end-leafs.
- clock selectors, which act as switch to select one out of many
possible clocks sources.
- clock multipliers, which have an input clock source, and act as clock
source again. They can be used to derive one clock from another.
All sample rate changes, clock validity queries and the like must go to
clock source elements, while clock selectors and multipliers can be used
as terminal clock source.
The following patch adds a parser for these elements and functions to
iterate over the tree and find the leaf nodes (clock sources).
The samplerate set functions were moved to the new clock.c file.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, UAC2 controls are marked read-only if any of the channels are
marked read-only in the descriptors. Change this behaviour and
- mark them writeable unless all channels are read-only
- store the read-only mask in usb_mixer_elem_info and
- check the mask again in set_cur_mix_value(), and bail out for
write-protected channels.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce two new static inline functions for a more readable parsing
of UAC2 bmaControls.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: snd-usb-caiaq: Bump version number to 1.3.21
ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
ALSA: snd-usb-caiaq: Simplify single case to an 'if'
ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
ALSA: hda: Use LPIB for a Shuttle device
ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
ALSA: hda: Use LPIB for Sony VPCS11V9E
ALSA: usb-audio: fix feature unit parser for UAC2
ALSA: asihpi - Minor code cleanup
ALSA: asihpi - Add support for new ASI8800 family
ALSA: asihpi - Fix bug preventing outstream_write preload from happening
ALSA: asihpi - Fix imbalanced lock path in hw_message
ALSA: asihpi - Remove support for old ASI8800 family
ALSA: asihpi - Add hd radio blend functions
ALSA: asihpi - Remove unused io map functions
ALSA: usb-audio: add support for UAC2 pitch control
ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
ALSA: usb-audio: fix return values
ALSA: usb-audio: parse more format descriptors with structs
sound: Add missing spin_unlock
...
Do not explicity set the default input mode. Use the hardware default
of mode 0 ('Control vinyl'), which is now available.
This reverts commit e3ca4c9.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After removing code, only one case remains. So use an 'if' instead.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This feature was undocumented on early A4DJ units. It is indicated
by lighting both the 'line' and 'phono' lamps at the same time.
Newer units document this and the newer Windows drivers enable this
for all units, so restore the functionality.
This patch simplifies the code and changes the mode mapping to match
the A8DJ, favouring simpler code and consistency over keeping the
existing mapping.
Both 'Control vinyl' and 'Phono' input modes enable the hardware
preamp. The difference is the input impedance.
This reverts commit 9a9527e.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a small off-by-one bug which causes the feature unit to announce a
wrong number of channels. This leads to illegal requests sent to the
firmware eventually.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This request is again handled differently in comparison to UAC1.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 devices have their information about pitch control stored in a
different field. Parse it, and emulate the bits for a v1 device.
A new struct uac2_iso_endpoint_descriptor is added.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-1 is not a good return value as it means -EPERM, "not permitted".
Choose -ENOTSUPP instead, which is what the code really wants to tell
its callers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The decoding/encoding is based on own reverse-engineering. Both control and
data ports are handled. Writing to control port supports SysEx events only,
as this is the only type of messages that MPD16 recognizes.
Signed-off-by: Krzysztof Foltman <wdev@foltman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is needed before the USB merge.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
For more clearance what the functions actually do,
usb_buffer_alloc() is renamed to usb_alloc_coherent()
usb_buffer_free() is renamed to usb_free_coherent()
They should only be used in code which really needs DMA coherency.
All call sites have been changed accordingly, except for staging
drivers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: Pedro Ribeiro <pedrib@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
For both UAC1 and UAC2, interrupt endpoint messages are now parsed with
structs rather that with anonymous buffer array accesses.
For UAC2, only CUR interrupt notifications are supported for now.
snd_usb_mixer_status_complete() was renamed to
snd_usb_mixer_interrupt().
Fixed one indentation flaw on the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 23caaf19b ("ALSA: usb-mixer: Add support for Audio Class v2.0")
broke support for Class1 devices due to two faulty changes. This patch
fixes it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-and-Tested-by: The Source <thesourcehim@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after
disconnection. This is due to the access to the endpoints which have
been already released at disconnection while the files are still alive.
This patch fixes the problem by checking disconnection state at
snd_usbmidi_output_drain() and by releasing urbs but keeping the
endpoint instances until really all freed.
Tested-by: Tvrtko Ursulin <tvrtko@ursulin.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files. percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.
percpu.h -> slab.h dependency is about to be removed. Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability. As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.
http://userweb.kernel.org/~tj/misc/slabh-sweep.py
The script does the followings.
* Scan files for gfp and slab usages and update includes such that
only the necessary includes are there. ie. if only gfp is used,
gfp.h, if slab is used, slab.h.
* When the script inserts a new include, it looks at the include
blocks and try to put the new include such that its order conforms
to its surrounding. It's put in the include block which contains
core kernel includes, in the same order that the rest are ordered -
alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
doesn't seem to be any matching order.
* If the script can't find a place to put a new include (mostly
because the file doesn't have fitting include block), it prints out
an error message indicating which .h file needs to be added to the
file.
The conversion was done in the following steps.
1. The initial automatic conversion of all .c files updated slightly
over 4000 files, deleting around 700 includes and adding ~480 gfp.h
and ~3000 slab.h inclusions. The script emitted errors for ~400
files.
2. Each error was manually checked. Some didn't need the inclusion,
some needed manual addition while adding it to implementation .h or
embedding .c file was more appropriate for others. This step added
inclusions to around 150 files.
3. The script was run again and the output was compared to the edits
from #2 to make sure no file was left behind.
4. Several build tests were done and a couple of problems were fixed.
e.g. lib/decompress_*.c used malloc/free() wrappers around slab
APIs requiring slab.h to be added manually.
5. The script was run on all .h files but without automatically
editing them as sprinkling gfp.h and slab.h inclusions around .h
files could easily lead to inclusion dependency hell. Most gfp.h
inclusion directives were ignored as stuff from gfp.h was usually
wildly available and often used in preprocessor macros. Each
slab.h inclusion directive was examined and added manually as
necessary.
6. percpu.h was updated not to include slab.h.
7. Build test were done on the following configurations and failures
were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my
distributed build env didn't work with gcov compiles) and a few
more options had to be turned off depending on archs to make things
build (like ipr on powerpc/64 which failed due to missing writeq).
* x86 and x86_64 UP and SMP allmodconfig and a custom test config.
* powerpc and powerpc64 SMP allmodconfig
* sparc and sparc64 SMP allmodconfig
* ia64 SMP allmodconfig
* s390 SMP allmodconfig
* alpha SMP allmodconfig
* um on x86_64 SMP allmodconfig
8. percpu.h modifications were reverted so that it could be applied as
a separate patch and serve as bisection point.
Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.
Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the EHCI driver copes with small iso packets without blowing
up, take the snd-ua101 driver out of the alpha-test stage.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix build errors when CONFIG_PM is not enabled:
sound/usb/card.c:629: error: 'usb_audio_suspend' undeclared here (not in a function)
sound/usb/card.c:630: error: 'usb_audio_resume' undeclared here (not in a function)
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device does not have audio controllers and backlit buttons only.
Input data is handled over a dedicated USB endpoint.
All functions are supported by the driver now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio Class v2.0 compliant devices have different descriptors and a
different way of setting/getting min/max/res/cur properties. This patch
adds support for them.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a number of new structs for mixer, selector, feature and
processing units and some static inline helpers to access fields which
have dynamic offsets. Use them in mixer.c to parse the descriptors. This
is necessary for the upcoming audio v2 parsers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For clearer namespace, also rename usbmixer_maps.c -> mixer_maps.c
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move all non-standard mixer controls and vendor-specific extensions to a
separate file. Some structs need to be exported now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No need for the private enum.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Split the audio.h file in two to clearly denote the differences
between the standards.
- Add many more defines to audio-v2.h. Most of them are not currently
used.
- Replaced a magic value with a proper define
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sample rate setting is done with a 4-byte long class request that
addresses the interface.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the parser to correctly handle v2 descriptors with multiple
format bits set.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for USB audio 2.0 support, change the audioformat
structure so that it uses a bitmask to specify possible formats.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_usb_substream::format field actually contains the index of the
current alternate setting, so rename it to altset_idx to avoid
confusion.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the usb audio driver by factoring out a lot of functions to
separate files. Code for procfs, quirks, urbs, format parsers etc all
got a new home now.
Moved almost all special quirk handling to quirks.c and introduced new
generic functions to handle them, so the exceptions do not pollute the
whole driver.
Renamed usbaudio.c to card.c because this is what it actually does now.
Renamed usbmidi.c to midi.c for namespace clarity.
Removed more things from usbaudio.h.
The non-standard drivers were adopted accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename snd-usb-lib to snd-usbmidi-lib as MIDI functions are the only
thing it actually contains. Introduce a new header file to only declare
these functions.
Introduced usbmixer.h for all functions exported by usbmixer.c.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As part of the USB audio code cleanup, move the non-standard ua101
driver out of the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch works around misbehaviour of Creative Creative VF0470 Live Cam
which reports 16 kHz sample rate for audio capture while actually producing
8 kHz stream.
Signed-off-by: Arseniy Lartsev <arseniy@fizlesh.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some code that is no longer needed now that the relevant parts of
the driver have been tested.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar"
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.
Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.
Now things are also nicely prefixed which makes understanding the code
easier.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usbmixer proc file contains mapping between ALSA control API and
USB mixer control units. The purpose of this file is for debugging
and a problem diagnostics.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.
The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.
Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
