| Commit message (Collapse) | Author | Age | Files | Lines |
|---|
| |
|
|
|
|
|
| |
Mia has an undocumented line-out control, and it has to be exposed.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |
|
|
|
|
|
|
| |
we cannot set the sampling rate of the device, but can only read it
from the board, so we don't need the member for it.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |
|
|
|
|
|
|
| |
the rmh bus is not used asynchronously, so it is safe to remove the
specific code pieces.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |
|
|
|
|
|
|
| |
Simplify snd_pcm_drain() implementation and avoid unneeded array-
allocation for waitqueues. Instead, one waitqueue is used for the
first draining stream, and wait until all streams finished.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
* fix/hda:
ALSA: hda - Fix MSI GX620 mixer
ALSA: hda - Fix Dell S14 pin setup
ALSA: hda - Fix IDT92HD83* codec setup
ALSA: hda - Add support for HP dv6
ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
ALSA: hda - Set default GPIO for IDT92HD71bxx
ALSA: hda - Set default GPIO for STAC/IDT codecs
ALSA: hda - Add missing model=auto entry for ALC269
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
The headphone and speaker mixer elements aren't properly set for
MSI GX620 with targa-8ch-dig quirk.
Also fixed the speaker volume control for other ALC883-targa quirks,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| | |
The pin setup for Dell S14 quirk is rather wrong for the latest driver.
Fixed pin 0x0a, 0x0b, 0x0d and 0x0f.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
Remove unnecessary (and buggy) init sequences left for IDT92HD83*
codecs in the previous fixes. The DACs are now dynamically connected,
thus shouldn't be set statically in init verbs. Also, the mono_nid
is detected dynamically, thus shouldn't be set staticaly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| |
| |
| | |
Add the quirk entry for HP dv6. Also add a workaround for the headphone
detection by setting hp_detect=1 beforehand. Without this, the driver
won't do auto-muting because BIOS doesn't give any HP pin but only a
line-out pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
It's possible that hp_detect is set even though no headphone pin is
detected. The driver issues, however, an unsol event only to hp_pins[0],
which can be invalid.
This patch adds the check of the valid pin to send an unsol event
at initialization and resume callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| | |
A smiliar fix for IDT 92HD71Bxx codecs like the previous commit for
other IDT/STAC codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines.
However, currently we don't set it unless the model is specified just
for safety reason. But, most machines do need this bit, so this safety
handling is rather annoying.
This patch enables GPIO0 setup as default for them. Many HP / Dell
laptops should work even without model override with this change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | |
| |
| |
| | |
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
* fix/asoc:
ASoC: remove unused #include <linux/version.h>
ASoC: S3C lrsync function made to work with IRQs disabled.
ASoC: Fix display of stream name in DAPM debugfs
ASoC: Clean up error handling in MPC5200 DMA setup
|
| | | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
Remove unused #include <linux/version.h>('s) in
sound/soc/codecs/ad1836.c
sound/soc/codecs/ad1938.c
sound/soc/codecs/wm8974.c
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
| | | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK
is dead due to improper initialization of CPU or CODEC, the system
gets stuck in the loop because jiffies may never get updated.
Implemented counter based wait mechanism for atleast the same
timeout period.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
| | | |
| | |
| | |
| | |
| | |
| | | |
Also display streams all the time while we're here.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
| | | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.
The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
when != if (...) { <+...x...+> }
(
x->f1 = E
|
(x->f1 == NULL || ...)
|
f(...,x->f1,...)
)
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
| |\ \ \
| | | |
| | | |
| | | |
| | | | |
* topic/ymfpci:
sound: ymfpci: increase timer resolution to 96 kHz
|
| | | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | | |
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
* topic/usb-audio:
ALSA: usb-audio - Fix types taken in min()
sound: usb-audio: do not make URBs longer than sync packet interval
sound: usb-audio: add MIDI drain callback
sound: usb-audio: use multiple output URBs
sound: usb-audio: use multiple input URBs
sound: usb-audio: Xonar U1 digital output support
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
Fix the compile warning due to different integer types used in min():
sound/usb/usbaudio.c: In function 'init_substream_urbs':
sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
When draining, instead of waiting for fifty milliseconds, just wait for
the currently active URBs to complete. This cuts the usual waiting time
down to one USB frame, or zero in the common case when there is no URB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to send multiple packets in one USB
frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to receive multiple packets in one
USB frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
Add support for the Asus Xonar U1. This device is mostly class compliant, but
the digital output requires a vendor-specific request.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | | |
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
|
| | | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | | |
Some USB devices give bogus dB information and it screws up PA.
It's better to detect a broken value and correct it in the driver
before exposing the value to the outside.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | | |
Use the new TLV_DB_MINMAX type instead of TLV_DB_SCALE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | | |
Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides. For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
* topic/soundcore-preclaim:
sound: make OSS device number claiming optional and schedule its removal
sound: request char-major-* module aliases for missing OSS devices
chrdev: implement __[un]register_chrdev()
|
| | | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases. This feature is redundant as chrdev already has such
mechanism. This preemptive claiming prevents alternative OSS
implementation.
The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.
This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel. The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.
As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.
For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded. The custom module aliases don't
have any inherent benefit than backward compatibility.
sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | |_|/ / /
| |/| | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | |
| | | | | | |
[un]register_chrdev() assume minor range 0-255. This patch adds __
prefixed versions which take @minorbase and @count explicitly.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
* topic/snd-printk:
ALSA: Fixed a typo of printk()
ALSA: Add debug module option
ALSA: core - strip too long file names in snd_print*()
|
| | | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
Fixed a silly typo of printk() included in the previous patch...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | |
| | | | | | | |
When modules are built with M= option, they pass long file paths to
__FILE__. This results in ugly outputs of snd_print*() when
CONFIG_SND_VERBOSE_PRINTK is set.
This patch adds a check of the path and strips the leading path dirs
if the file name is an absolute path to improve the readability of logs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \ \
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | | |
* topic/pcm-estrpipe-in-pm:
ALSA: pcm - Tell user that stream to be rewound is suspended
|
| | | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | | | | | | | |
Return STRPIPE instead of EBADF when userspace attempts to rewind
of forward a stream that was suspended in meanwhile, so that it
can be recovered by snd_pcm_recover().
This was causing Pulseaudio to unload the ALSA sink module under a race
condition when it attempted to rewind the stream right after resume from
suspend, before writing to the stream which would cause it to revive the
stream otherwise. Tested to work with Pulseaudio patched to attempt to
snd_pcm_recover() upon receiving an error from snd_pcm_rewind().
Signed-off-by: Lubomir Rintel <lkundrak@v3.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \ \ \
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | | |
* topic/pcm-drain-nonblock:
ALSA: pcm - Increase protocol version
ALSA: pcm - Fix drain behavior in non-blocking mode
|
| | | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | | |
Increase the PCM protocol version to indicate the drain ioctl behavior
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | | |
The current PCM core has the following problems regarding PCM draining
in non-blocking mode:
- the current f_flags isn't checked in snd_pcm_drain(), thus changing
the mode dynamically via snd_pcm_nonblock() after open doesn't work.
- calling drain in non-blocking mode just return -EAGAIN error, but
doesn't provide any way to sync with draining.
This patch fixes these issues.
- check file->f_flags in snd_pcm_drain() properly
- when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state
but quits ioctl immediately without waiting the whole drain; the
caller can sync the drain manually via poll()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \ \ \ \
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | | |
* topic/oxygen:
sound: oxygen: work around MCE when changing volume
|
| | | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | | |
When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it. On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.
To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |\ \ \ \ \ \ \ \ \ \
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | | |
* topic/oss:
ALSA: allocation may fail in snd_pcm_oss_change_params()
sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
sound: fix OSS MIDI output data loss
|
| | | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | | |
Allocation may fail, show if it did.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | |
| | | | | | | | | | | |
Since !LI_CCFG_* evaluates to 0, this did not change anything to
cfgval and ctlval.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| | | |_|_|_|/ / / / /
| |/| | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | |
| | | | | | | | | | |
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call. Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
