Commit Graph

648 Commits

Author SHA1 Message Date
Mark Brown
913d7b4cc0 ASoC: Add support for WM8960 capless mode
The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.

Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-03 17:08:43 +00:00
Mark Brown
b6877a477d ASoC: Move WM8960 platform data into include/sound
Avoids machine files having to peer into sound/soc which is a bit
rude and icky.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-03 17:08:42 +00:00
Peter Ujfalusi
258020d088 ASoC: core: Add delay operation to snd_soc_dai_ops
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-03 17:08:41 +00:00
Jassi Brar
14dc5734bd ASoC: Allow mulitple usage count of codec and cpu dai
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-26 11:17:48 +00:00
jassi brar
6423c1875c ASoC: Remove runtime field from DAI
In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22 14:15:30 +00:00
jassi brar
d273ebe77a ASoC: Pass dai_link as argument to platform suspend and resume
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22 14:14:58 +00:00
Mark Brown
6c5f1fed49 ASoC: Make pmdown_time a long
Fixes a warning.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-17 14:37:20 +00:00
Mark Brown
96dd362284 ASoC: Make pmdown_time a per-card setting
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-16 19:14:52 +00:00
Mark Brown
3a66d3877e ASoC: Add WM2000 driver
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-12 10:31:06 +00:00
Mark Brown
a3032b47c4 ASoC: Add a cache_sync bit to the CODEC structure
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache.  This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-04 10:40:45 +00:00
Mark Brown
8c961bcca1 ASoC: Allow CODECs to ask soc-cache to suppress physical writes
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active.  Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-03 18:03:37 +00:00
Guennadi Liakhovetski
6c2fb6a8d8 ASoC: add helper macros to declare struct soc_enum instances
Several shortcuts for popular uses of some of SOC_ENUM_* and
SOC_VALUE_ENUM_* macros.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-25 14:51:02 +00:00
Guennadi Liakhovetski
8484c63f4b ASoC: add simplified versions of widget macros
Many macros from include/sound/soc-dapm.h take an array and a number of
elements in it as arguments, whereas most users use static arrays and use
"x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of
those macros, calling ARRAY_SIZE() internally.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-25 14:50:45 +00:00
Mark Brown
a96ca33873 ASoC: Support turning off bias when the CODEC is idle
Currently ASoC always maintains the bias of the CODEC while the system
is active.  With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.

As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias.  The distinction between STANDBY and OFF is still
maintained.  This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-01-21 12:04:08 +00:00
Peter Ujfalusi
6aceabb459 ASoC: tlv320dac33: Burst mode BCLK divider configuration
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-20 11:47:49 +00:00
Guennadi Liakhovetski
84740ac19a ASoC: fix compile breakage - add a missing header include
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-19 12:36:40 +00:00
Mark Brown
163849ea9b Merge branch 'for-2.6.33' into for-2.6.34 2010-01-12 12:59:05 +00:00
Ilkka Koskinen
2138301e16 ASoC: tpa6130a2: Support for tpa6140's regulators
tpa6140a2 uses different names for the regulators.

Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-11 17:13:11 +00:00
Mark Brown
b35a28af0a ASoC: Add initial WM8955 CODEC driver
The WM8955 is a low power, high quality stereo DAC with integrated
headphone and loudspeaker amplifiers, designed to reduce external
component requirements in portable digital audio applications. This is
an initial driver implementing support for the majority of the
functionality in the device, currently OUT3 is not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-12-18 13:06:47 +00:00
Linus Torvalds
4ef58d4e2a Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (42 commits)
  tree-wide: fix misspelling of "definition" in comments
  reiserfs: fix misspelling of "journaled"
  doc: Fix a typo in slub.txt.
  inotify: remove superfluous return code check
  hdlc: spelling fix in find_pvc() comment
  doc: fix regulator docs cut-and-pasteism
  mtd: Fix comment in Kconfig
  doc: Fix IRQ chip docs
  tree-wide: fix assorted typos all over the place
  drivers/ata/libata-sff.c: comment spelling fixes
  fix typos/grammos in Documentation/edac.txt
  sysctl: add missing comments
  fs/debugfs/inode.c: fix comment typos
  sgivwfb: Make use of ARRAY_SIZE.
  sky2: fix sky2_link_down copy/paste comment error
  tree-wide: fix typos "couter" -> "counter"
  tree-wide: fix typos "offest" -> "offset"
  fix kerneldoc for set_irq_msi()
  spidev: fix double "of of" in comment
  comment typo fix: sybsystem -> subsystem
  ...
2009-12-09 19:43:33 -08:00
Mark Brown
a91eb199e4 ASoC: Initial WM8904 CODEC driver
The WM8904 is a high performance ultra-low power stereo CODEC
optimised for portable audio applications, with features including
a class W amplifier, FLL with free running mode, Mobile ReTune and
ground referenced headphone and line outputs.

Support for some features, most particularly the digital microphone
interface, is not yet present.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-12-04 16:50:53 +00:00
Mark Brown
dd1b3d53c2 ASoC: Export snd_soc_update_bits_unlocked()
Allows custom controls to use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-12-04 16:07:06 +00:00
Takashi Iwai
86e1d57e4f Merge branch 'topic/hda' into for-linus 2009-12-04 16:22:45 +01:00
Takashi Iwai
baf9226667 Merge branch 'topic/asoc' into for-linus 2009-12-04 16:22:41 +01:00
Takashi Iwai
57648cd52b Merge branch 'topic/misc' into for-linus 2009-12-04 16:22:37 +01:00
André Goddard Rosa
af901ca181 tree-wide: fix assorted typos all over the place
That is "success", "unknown", "through", "performance", "[re|un]mapping"
, "access", "default", "reasonable", "[con]currently", "temperature"
, "channel", "[un]used", "application", "example","hierarchy", "therefore"
, "[over|under]flow", "contiguous", "threshold", "enough" and others.

Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-12-04 15:39:55 +01:00
Jean Delvare
83cf0a9b86 comment typo fix: sybsystem -> subsystem
Signed-off-by: Jean Delvare <jdelvare@suse.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-12-04 15:39:49 +01:00
Takashi Iwai
75639e7ee1 Merge branch 'topic/beep-rename' into topic/core-change 2009-12-01 15:58:10 +01:00
Takashi Iwai
980f31c46b Merge branch 'topic/ice1724-quartet' into topic/hda 2009-12-01 15:57:01 +01:00
Mark Brown
c0fa59df72 ASoC: Add BCLK calculation utility for TDM mode too
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-25 19:55:46 +00:00
Krzysztof Helt
9dc9120c77 ALSA: opti-miro: expose ACI mixer to outside drivers
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-23 09:41:55 +01:00
Krzysztof Helt
9aeba62971 ALSA: opti-miro: make miro.h header available outside the alsa directory
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-23 09:41:46 +01:00
Krzysztof Helt
b753e03e5e ALSA: cs4236: update control names
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.

Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.

Also, delete one misnamed cs4231 register define.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-18 07:23:16 +01:00
Joonyoung Shim
c871a05315 ASoC: Add jack_status_check callback function for GPIO jacks
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-12 16:45:53 +00:00
Mark Brown
7aae816dae ASoC: Add bit clock rate calculator utility functions
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-11-12 16:45:48 +00:00
Clemens Ladisch
7584af10cf sound: rawmidi: record a substream's owner process
Record the pid of the task that opened a RawMIDI substream.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-10 16:32:38 +01:00
Clemens Ladisch
e7373b702f sound: pcm: record a substream's owner process
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-10 16:32:20 +01:00
Clemens Ladisch
25d27eded1 control: use reference-counted pid
Instead of storing the PID number, take a reference to the task's pid
structure.  This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-06 14:32:06 +01:00
Clemens Ladisch
31cef7076e control: remove snd_konctrol_volatile::owner_pid field
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-06 14:32:03 +01:00
Krzysztof Helt
d114cd84a1 ALSA: cs4236: detect chip in one pass
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.

Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-05 18:10:25 +01:00
Rafael Ignacio Zurita
9dcaa7b25f ALSA: sh: add SuperH DAC audio driver for ALSA V4
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).

Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.

Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-04 09:17:40 +01:00
Mark Brown
fe3e78e073 ASoC: Factor out snd_soc_init_card()
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-03 22:14:43 +00:00
Peter Ujfalusi
c8bf93f0fe ASoC: Codec driver for Texas Instruments tlv320dac33 codec
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.

TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.

The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.

Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).

b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:02:04 +01:00
Mark Brown
d2058b0cd0 ASoC: Remove snd_soc_suspend_device()
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:01:43 +01:00
Peter Ujfalusi
493b67efff ASoC: TPA6130A2 amplifier driver
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.

The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.

The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"

From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':

        {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
        {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},

Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-09 18:50:37 +01:00
Mark Brown
907bc6c7fc Merge branch 'for-2.6.32' into for-2.6.33 2009-10-06 16:01:27 +01:00
Mark Brown
d2b247a8be ASoC: Add virtual enumeration support for DAPM muxes
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 15:57:02 +01:00
Takashi Iwai
7c824f4b69 ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 07:22:58 +02:00
Peter Ujfalusi
88439ac793 ASoC: add support for multiple cards/codecs in debugfs
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:

debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
                                                  /dapm_pop_time
                                                  /dapm/{widgets}

With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-01 12:13:04 +01:00
Krzysztof Helt
acd4710091 ALSA: sscape: convert to firmware loader framework
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.

An additional firmware initialization code has been moved from the OSS
driver.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 07:51:56 +02:00