This patch adds support for ASUS - Xonar DSX sound cards. Tested on
openSUSE 12.2 with kernel:
Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux
Works:
- play sounds
- adjust volume on master channel.
- mute .
Since Xonar DS uses the same chip, everything that works for DS should
work for DSX as well.
Signed-off-by: Sergiu Giurgiu <sgiurgiu11@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Starting audio or seeking in various music players causes
setup_dig_out_stream() to be called, which resets the SPDIF stream,
which caused one DAC (but not another) to make a clicking noise every
time.
This patch ensures the reset only happens when it needs to, which is
when the format changes, and makes the code a little more readable.
Signed-off-by: Laurence Darby <ldarby@tuffmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
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>
Fix races at PCM disconnection:
- while a PCM device is being opened or closed
- while the PCM state is being changed without lock in prepare,
hw_params, hw_free ops
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev_<level> calls take less code than dev_printk(KERN_<LEVEL>
and reducing object size is good.
Coalesce multiline formats for easier grep.
Coalesce segmented single line formats too.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
Commit 4eeaaeaea (ALSA: core: add hooks for audio timestamps) added the
new audio_tstamp field to struct snd_pcm_status. However, struct
timespec requires 64-bit alignment, so the 64-bit compiler would insert
32 bits of padding before this field, which broke SNDRV_PCM_IOCTL_STATUS
with error messages like this:
kernel: unknown ioctl = 0x80984120
To solve this, insert the padding explicitly so that it can be taken
into account when calculating the ABI structure size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should really use "fck" when asking for the functional clock and not
"dmic_fck".
This way we can ensure that multiple dmic modules can exist in the system.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also drop the includes that are no longer needed and just
cause problems for the ARM common zImage.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
[tony@atomide.com: updated to drop unneeded headers]
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The BIOS on HP dv5 doesn't have the DMI string to guide the setup of
mute led GPIO and polarity. Associate this laptop with the hp-inv-led
model.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: Vinícius Angiolucci <angiolucci@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A couple of driver fixes, one that improves the interoperability of
WM8994 with controllers that are sensitive to extra BCLK cycles and some
build break fixes for ux500.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
A couple of driver fixes, one that improves the interoperability of
WM8994 with controllers that are sensitive to extra BCLK cycles and some
build break fixes for ux500.
The check of the return value from vortex_wtdma_bufshft() in
vortex_interrupt() is useless as it executes no code. Actually the
return value is intentionally ignored because the delta calculation
for wavetable doesn't work always correctly. For avoiding the
confusion, a comment is added and the superfluous if () is removed.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
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>
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>
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>
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>
Rather than always assuming the maximum possible BCLK rate will be
required generate BCLKs for stereo if either one or two channels is
enabled. In order to support this we also need to ensure that only
the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors by using correct kconfig symbol name:
sound/pci/ice1712/psc724.c:417:5: error: 'struct snd_ice1712' has no member named 'pm_resume'
sound/pci/ice1712/psc724.c:418:5: error: 'struct snd_ice1712' has no member named 'pm_suspend_enabled'
[Fixed another #ifdef CONFIG_PM in the same file, too, by tiwai]
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
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>
Reuse code from clocksource to handle wall clock counter.
Since wrapparound occurs, the audio timestamp is reinitialized
to zero on a trigger. Synchronized linked devices will
start counting from same reference to avoid any drift.
Max buffer time is limited to 178 seconds to make sure
wall clock counter does not overflow
Wallclock timestamps are disabled on capture streams
until we figure out how to handle digital inputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable stream 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>
snd_miro_probe is a static function that is only called twice in the file
that defines it. At each call site, its argument is freed using
snd_card_free. Thus, there is no need for snd_miro_probe to call
snd_card_free on its argument on any of its error exit paths.
Because snd_card_free both reads the fields of its argument and kfrees its
argments, the results of the second snd_card_free should be unpredictable.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r@
identifier f,free,a;
parameter list[n] ps;
type T;
expression e;
@@
f(ps,T a,...) {
... when any
when != a = e
if(...) { ... free(a); ... return ...; }
... when any
}
@@
identifier r.f,r.free;
expression x,a;
expression list[r.n] xs;
@@
* x = f(xs,a,...);
if (...) { ... free(a); ... return ...; }
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.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>
There are small races opened in the check of running bit and the timer
lock. Instead of adding yet more flag, just protect the whole racy
codes with the existing cable->lock. As a bonus, we can get rid of
timer_lock now.
Reported-and-tested-by: Omair Mohammed Abdullah <omair.m.abdullah@linux.intel.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>
Quite a few mixer applications do not handle deactivated controls
correctly. This patch adds such controls to snd-dummy to make
crash^H^H^H^H^Htesting these apps easier.
To make the testing of deactivated mixer controls easier (and for people
with common hardware, possible), add a control that deactivates some
other controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
By some reason, Toshiba laptop doesn't like the EAPD turned up for the
headphone pin. Add a fix up code to force to turn down EAPD for NID
0x15.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=569991
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
checkpatch.pl discourages the use of spaces at the beginning of lines.
Some of the CTL_ELEM defines were not properly indented.
This patch replaces the leading spaces by tabs. No functionality is
changed, the commit is purely cosmetic.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the documentation, AES32 cards use a different bit position
for reporting the sync_in status.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In contrast to AES32, MADI uses the first status register to report the
sync_in status.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MADI and MADIface used to report the autosync_sample_rate. This
functionality was lost in commit
0dca179306, this commit now adds it back.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Missing breaks lead to a fall-through, thus causing the wrong
autosync_sample_rate to be reported.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to missing breaks and the resulting fall-through, card subtype
selection was effectively missing, thus causing the wrong sync check
functions to be called.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a follow-up to a97bda7d29, report the
external sample rate as system_sample_rate when in slave mode.
For PCIe MADI cards, the DDS value automatically contains the external
sample rate, but the PCI version needs this manual workaround.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DDS value is the actual physical sample rate. We set it indirectly
when selecting 44100, 48000 and so on via snd_hdspm_hw_params or
hdspm_set_clock_source.
This commit now allows the DDS value to be altered at runtime, thus
speeding up or slowing down the physical sample rate. This is required
for MADI's varispeed that allows for ±12.5% speed adjustment from the
"selected" rate (32kHz, 44100kHz, 48kHz and so on).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking
station.
Without this patch, if I plug my headphones into the jack on the
computer, everything works fine. The computer speakers mute and the
audio is played in the headphones. However, if I plug into the docking
station headphone jack the computer speakers are muted but there is no
audio in the headphones.
Addresses https://bugs.launchpad.net/bugs/1060372
Signed-off-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include linux/io.h in ice1712.h since inb() and outb() are used in
inline functions there. Remove the redundant inclusion of that file
in other places at the same time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add chip details for E-mu 1010 PCIe card. It has the same
chip as found in E-mu 1010b but it uses different PCI id.
Signed-off-by: Maxim Kachur <mcdebugger@duganet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
