Dan Rosenberg [Wed, 23 Mar 2011 15:42:57 +0000 (11:42 -0400)]
sound/oss/opl3: validate voice and channel indexes
User-controllable indexes for voice and channel values may cause reading
and writing beyond the bounds of their respective arrays, leading to
potentially exploitable memory corruption. Validate these indexes.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Rosenberg [Wed, 23 Mar 2011 14:53:41 +0000 (10:53 -0400)]
sound/oss: remove offset from load_patch callbacks
Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of
uninitialized value, and signedness issue
The offset passed to midi_synth_load_patch() can be essentially
arbitrary. If it's greater than the header length, this will result in
a copy_from_user(dst, src, negative_val). While this will just return
-EFAULT on x86, on other architectures this may cause memory corruption.
Additionally, the length field of the sysex_info structure may not be
initialized prior to its use. Finally, a signed comparison may result
in an unintentionally large loop.
On suggestion by Takashi Iwai, version two removes the offset argument
from the load_patch callbacks entirely, which also resolves similar
issues in opl3. Compile tested only.
v3 adjusts comments and hopefully gets copy offsets right.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
Similar to commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch
avoids unnecessary volume control indices for more
Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side"
controls.
These indices cause these volume controls to be ignored by PulseAudio and
vmaster and should be removed whenever possible.
Cc: stable@kernel.org Reported-by: Jan Losinski <losinski@wh2.tu-dresden.de> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 21 Mar 2011 11:00:00 +0000 (12:00 +0100)]
ALSA: usb - Remove trailing spaces from USB card name strings
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.
Xiaochen Wang [Fri, 18 Mar 2011 08:29:25 +0000 (16:29 +0800)]
sound: read i_size with i_size_read()
Convert direct read of inode->i_size to using i_size_read().
i_size_read is guaranteed to return a valid value and
its caller does not need to use addtional locking.
Signed-off-by: Xiaochen Wang <wangxiaochen0@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Wed, 16 Mar 2011 18:18:53 +0000 (18:18 +0000)]
ASoC: Remove bogus check for register validity in debugfs write
Since not all registers need to be cached and the cache is entirely
optional anyway we shouldn't be checking that a register is in the
cached range. If the register is invalid then the actual I/O code
can determine that and report an error.
Similarly, the step size can and should be enforced by the lower level
code if it's important.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Dan Rosenberg [Thu, 17 Mar 2011 22:32:24 +0000 (18:32 -0400)]
ALSA: sound/pci/asihpi: check adapter index in hpi_ioctl
The user-supplied index into the adapters array needs to be checked, or
an out-of-bounds kernel pointer could be accessed and used, leading to
potentially exploitable memory corruption.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 18 Mar 2011 06:31:53 +0000 (07:31 +0100)]
ALSA: aloop - Fix possible IRQ lock inversion
loopback_pos_update() can be called in the timer callback, thus the lock
held should be irq-safe. Otherwise you'll get AB/BA deadlock together
with substream->self_group.lock.
Add an AMDTP stream error state that occurs when we fail to queue
another packet. In this case, the stream is stopped, and the error can
be reported when the application tries to restart the PCM stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
For correct cache coherency on some architectures, DMA buffers must be
allocated in a different cache line than data that is concurrently used
by the CPU.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Tue, 15 Mar 2011 06:55:02 +0000 (07:55 +0100)]
ALSA: firewire-lib: use no-info SYT for packets without SYT sample
In non-blocking mode, the SYT_INTERVAL is larger than the number of
audio frames in each packet, so there are packets that do not contain
any frame to which the SYT could be applied. For these packets, the
SYT must not be the timestamp of the next valid SYT frame, but the
special no-info SYT value.
This fixes broken playback on the FireWave at 44.1 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a driver for two playback-only FireWire devices based on the OXFW970
chip.
v2: better AMDTP API abstraction; fix fw_unit leak; small fixes
v3: cache the iPCR value
v4: FireWave constraints; fix fw_device reference counting;
fix PCR caching; small changes and fixes
v5: volume/mute support; fix crashing due to pcm stop races
v6: fix build; one-channel volume for LaCie
v7: use signed values to make volume (range checks) work; fix function
block IDs for volume/mute; always use channel 0 for LaCie volume
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Acked-by: Stefan Richter <stefanr@s5r6.in-berlin.de> Tested-by: Jay Fenlason <fenlason@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Thu, 10 Mar 2011 19:43:35 +0000 (13:43 -0600)]
ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecs
This patch replaces use of the harcoded arrays of pins, muxes, digital
mics and adcs with the auto-generated ones using codec parsing and
auto-discovers all actually connected digital mic pins on 92HD8X-like
codecs
This patch also adds the support for d-mic on pin 0x20.
Vitaliy Kulikov [Thu, 10 Mar 2011 01:47:43 +0000 (19:47 -0600)]
ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs
When the mux for digital mic is different from the mux for other mics,
the current auto-parser doesn't handle them in a right way but provides
only one mic. This patch fixes the issue.
Takashi Iwai [Thu, 10 Mar 2011 13:11:59 +0000 (14:11 +0100)]
ALSA: hda - Move default input-src selection to init part
Move the default input-src selection code for alc268/269 to the init
part instead of the parser. The input-src selection might be overwritten
by init verbs.
Takashi Iwai [Thu, 10 Mar 2011 11:51:11 +0000 (12:51 +0100)]
ALSA: hda - Initialize special cases for input src in init phase
Currently some special handling for the unusual case like dual-ADCs
or a single-input-src is done in the tree-parse time in
set_capture_mixer(). But this setup could be overwritten by static
init verbs.
This patch moves the initialization into the init phase so that
such input-src setup won't be lost.
Paul Bolle [Fri, 11 Mar 2011 10:24:52 +0000 (11:24 +0100)]
ALSA: intel8x0m: wait a bit before warm reset check
At every resume a laptop I use prints this message (at KERN_ERR level):
ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2]
The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what
seems to be happening is that the register involved indicated a warm
reset for some time (as the ICH_AC97WARM bit was set) but by the time
the warning is printed, and that same register is checked again, that
bit is already cleared and only the ICH_AC97COLD bit is still set.
It turns out a warm reset needs some time to settle, but it is currently
checked right away. The test therefore fails the first time it is done
and schedule_timeout_uninterruptible() will be called. Once we return
from that jiffies is already (far) past end_time on this laptop, so we
exit the loop, print a warning, and exit the function while the warm
reset actually succeeded.
A way to fix this is to call usleep_range() after writing to the
register involved. A handful of tests suggest 500 usecs is a safe value.
(This might punish the "finish cold reset" case, but on this laptop such
a cold reset apparently never happens, so I can't say for sure.)
While we're at it drop the extra single tick from end_time, as it looks
rather silly.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Oliver Neukum [Fri, 11 Mar 2011 13:51:12 +0000 (14:51 +0100)]
ALSA: usbaudio: implement USB autosuspend
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>
Marek Belisko [Wed, 9 Mar 2011 20:46:20 +0000 (21:46 +0100)]
ASoC: mini2440: Fix uda134x codec problem.
ASoC audio for mini2440 platform in current kenrel doesn't work.
First problem is samsung_asoc_dma device is missing in initialization.
Next problem is with codec. Codec is initialized but never probed
because no platform_device exist for codec driver. It leads to errors
during codec binding to asoc dai. Next problem was platform data which
was passed from board to asoc main driver but not passed to codec when
called codec_soc_probe().
Following patch should fix issues. But not sure if in correct way.
Please review.
Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM
Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: hda: Prevent writing ICH6_PCIREG_TCSEL on AMD systems
azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL
although this looks to be only defined on Intel systems and has a
different meaning on AMD systems. On AMD systems the PCI interrupt pin
control register is modified instead.
Since the meaning of offset 0x44 in device specific configuration space is
unknown for devices by other vendors, we only exclude AMD systems to
retain the current behaviour.
Signed-off-by: Adam Lackorzynski <adam@os.inf.tu-dresden.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88x
Without this change, a volume control named "Surround" or "Side" would
get an unnecessary index, causing it to be ignored by the vmaster and
PulseAudio.
Takashi Iwai [Wed, 9 Mar 2011 16:33:48 +0000 (17:33 +0100)]
sound: Use sound_register_*() for additional OSS minor devices
Since OSS driver creates the device entries for /dev/audio* and
/dev/dspW* by itself without coping with sound_core, it leads to
conflicts with others and let sysfs spewing warnings.
This patch rewrites the registration part of OSS driver to use
the standard method also for additional minor devices.
Reported-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl) Tested-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl) Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve EP93xx I2S clocks management.
Some freqs values are set not exact as they requested for MCLK and
original code was not able to find divisors for SCLK and LRCLK.
This code just picks up nearest value from 3 possible variants.
This patch makes 44100 and 192000 rates working and fixes
capture function (by selecting SCLK/LRCLK=64 where possible).
All other rates should work as before.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx
machine driver.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC.
This is done to make capture work properly.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 9 Mar 2011 11:33:09 +0000 (11:33 +0000)]
ASoC: Fix double addition of prefixes due to widget prefixing
We're not only prefixing all controls, we're also prefixing the widget
names in the runtime data. This causes us to add the prefix twice - once
when using the widget name to generate the control name and once when
adding the control.
Really we shouldn't be prefixing the widget names at all, the matching
code should be handing this as we always know which DAPM context a
widget came from and always display the widget name in terms of a DAPM
context. However, we're quite close to the merge window and that's
relatively invasive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Reported-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 8 Mar 2011 19:29:53 +0000 (19:29 +0000)]
ASoC: Use the correct DAPM context when cleaning up final widget set
Now we've got multi-component we need to make sure that the DAPM context
(and hence register I/O context) we use to apply the pending updates at
the end of a DAPM sequence is the one we were processing rather than the
one that was used to initate the state change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Tue, 8 Mar 2011 18:52:08 +0000 (18:52 +0000)]
ASoC: Simplify WM9081 speaker startup by using widgets for spaker output
Now we have a register write minimisation code in DAPM we don't need to
worry about the ordering of the enable and disable of the PGA and the
output stage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Wed, 9 Mar 2011 09:25:00 +0000 (11:25 +0200)]
ASoC: omap: rx51: Enable McBSP2 sidetone
McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.
Sidetone feature can be tested with following commands:
(set up codec input and output paths)
# Enable and configure sidetone
amixer -D hw:0 set 'McBSP2 Sidetone' on
amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
# Do not loop audio via CPU
arecord -f dat >/dev/null |aplay /dev/zero
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Tue, 8 Mar 2011 17:23:24 +0000 (17:23 +0000)]
ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 8 Mar 2011 00:17:56 +0000 (00:17 +0000)]
ASoC: Warn rather than set a silly constraint when we can't do symmetry
Symmetric rate configuration can fail if the second stream starting tries
to apply the symmetric constraint before the first stream has got far
enough to pick a rate. Rather than try to enforce a nonsensical rate of
0Hz log a warning and allow the application to carry on. Things might go
wrong later on but the user will know about it and there's unlikely to be
lasting damage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 20:58:11 +0000 (20:58 +0000)]
ASoC: Provide CODEC clocking operations and API calls
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 16:38:44 +0000 (16:38 +0000)]
ASoC: Add DAPM widget and path data to CODEC driver structure
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Clemens Ladisch [Mon, 7 Mar 2011 12:24:30 +0000 (13:24 +0100)]
ALSA: control: clean up snd_ctl_hole_check()
The return value of snd_ctl_hole_check() is used only to detect whether
to continue the loop in snd_ctl_find_hole() or not, so we can simplify
the code by changing this return type to a boolean. Also rename this
function to better show what it actually does.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 7 Mar 2011 12:22:50 +0000 (13:22 +0100)]
ALSA: control: fix numid conflict check for new controls
The purpose of the snd_ctl_hole_check() function is to find conflicts
between the numerical IDs of the new control and those of any existing
controls. However, it would fail to detect an existing control whose
count is smaller than the new control's count and whose interval of IDs
is entirely contained in the interval of the new control's IDs.
To fix this, use the correct formula to detect overlapping intervals,
which happens to simplify the condition.
This problem was not encountered so far because ALSA does not yet allow
drivers to allocate specific control IDs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Linus Torvalds [Tue, 8 Mar 2011 04:46:39 +0000 (20:46 -0800)]
Merge branch 's5p-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung
* 's5p-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung:
ARM: S3C64XX: Update regulator names for debugfs compatiblity on SMDK6410
ARM: S3C64XX: Fix build with WM1190 disabled and WM1192 enabled on SMDK6410
ARM: S3C64XX: Reduce output of s3c64xx_dma_init1()
ARM: S3C64XX: Tone down SDHCI debugging
ARM: S3C64XX: Add clock for i2c1
ARM: S3C64XX: Staticise non-exported GPIO to interrupt functions
ARM: SAMSUNG: Include devs.h in dev-uart.c to prototype devices
ARM: S3C64XX: Fix keypad setup to configure correct number of rows
ARM: S3C2440: Fix usage gpio bank j pin definitions on GTA02
ARM: S5P64X0: Fix number of GPIO lines in Bank F
ARM: S3C2440: Select missing S3C_DEV_USB_HOST on GTA02
Linus Torvalds [Tue, 8 Mar 2011 04:43:55 +0000 (20:43 -0800)]
Merge branch 'drm-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/airlied/drm-2.6
* 'drm-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/airlied/drm-2.6:
drm: index i shadowed in 2nd loop
drm/nv50-nvc0: prevent multiple vm/bar flushes occuring simultanenously
drm/nouveau: fix regression causing ttm to not be able to evict vram
drm/i915: Rebind the buffer if its alignment constraints changes with tiling
drm/i915: Disable GPU semaphores by default
drm/i915: Do not overflow the MMADDR write FIFO
Revert "drm/i915: fix corruptions on i8xx due to relaxed fencing"
Dmitry Shmidt [Thu, 3 Mar 2011 22:40:10 +0000 (17:40 -0500)]
mmc: sdio: Allow sdio operations in other threads during sdio_add_func()
This fixes a bug introduced by 807e8e40673d ("mmc: Fix sd/sdio/mmc
initialization frequency retries") that prevented SDIO drivers from
performing SDIO commands in their probe routines -- the above patch
called mmc_claim_host() before sdio_add_func(), which causes a deadlock
if an external SDIO driver calls sdio_claim_host().
Fix tested on an OLPC XO-1.75 with libertas on SDIO.
Signed-off-by: Dmitry Shmidt <dimitrysh@google.com> Reviewed-and-Tested-by: Chris Ball <cjb@laptop.org> Signed-off-by: Chris Ball <cjb@laptop.org>
Dave Airlie [Mon, 7 Mar 2011 21:18:35 +0000 (07:18 +1000)]
Merge remote branch 'ickle/drm-intel-fixes' into drm-fixes
* ickle/drm-intel-fixes:
drm/i915: Rebind the buffer if its alignment constraints changes with tiling
drm/i915: Disable GPU semaphores by default
drm/i915: Do not overflow the MMADDR write FIFO
Revert "drm/i915: fix corruptions on i8xx due to relaxed fencing"
Linus Torvalds [Mon, 7 Mar 2011 21:14:19 +0000 (13:14 -0800)]
Merge branch 'for-linus' of git://git390.marist.edu/pub/scm/linux-2.6
* 'for-linus' of git://git390.marist.edu/pub/scm/linux-2.6:
[S390] tape: deadlock on system work queue
[S390] keyboard: integer underflow bug
[S390] xpram: remove __initdata attribute from module parameters
The per-vm mutex doesn't prevent this completely, a flush coming from the
BAR VM could potentially happen at the same time as one for the channel
VM. Not to mention that if/when we get per-client/channel VM, this will
happen far more frequently.
Signed-off-by: Ben Skeggs <bskeggs@redhat.com> Signed-off-by: Dave Airlie <airlied@redhat.com>
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Ryan Mallon <ryan@bluewatersys.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Extend range of supported sample rates for CS4271 CODEC.
Extend range of supported sample rates for CS4271 CODEC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
projects / karo-tx-linux.git / log