Commit Graph

73 Commits

Author SHA1 Message Date
Mark Brown ac6d7c48e3 Merge remote-tracking branch 'asoc/topic/compress' into asoc-next 2014-01-20 11:50:41 +00:00
Liam Girdwood 2360702530 ASoC: DPCM: make some DPCM API calls non static for compressed usage
The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-17 17:56:21 +00:00
Mark Brown 2f43a23ab9 Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai 2014-01-16 12:42:57 +00:00
Mark Brown a9b68d3b90 Merge remote-tracking branch 'asoc/topic/core' into for-tiwai 2014-01-16 12:42:53 +00:00
Lars-Peter Clausen 55dcdb5051 ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper
Instead of open-coding the intersecting of two rate masks (and getting slightly
wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect()
helper function.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-14 20:42:34 +00:00
Lars-Peter Clausen 817873f4b1 ASoC: pcm: Properly initialize hw->rate_max
If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll
end up with the rate_max field of the runtime hardware set to 0.  (Note that it
is still possible for the components to constrain the supported sample rates
using other methods, e.g. setting a list constraint) If rate_max is 0 this means
that the sound card doesn't support any rates at all, which is not the desired
result. So initialize rate_max to UINT_MAX. For symmetry reasons also set
rate_min to 0.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-14 20:41:57 +00:00
Mark Brown 64a9aa9cf5 Linux 3.13-rc3
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Merge tag 'v3.13-rc3' into asoc-pcm

Linux 3.13-rc3
2014-01-14 20:41:53 +00:00
Lars-Peter Clausen 16d7ea9167 ASoC: Allow PCMs to restrict the supported formats
Some DMA cores might add additional restrictions on which in memory audio
formats can be supported by the compound sound card. If the PCM component
specifies a set of formats it supports (by setting the formats field to non 0)
take these into account when calculating the format set for the compound sound
card.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-09 14:31:59 +00:00
Liam Girdwood 1e9de42f43 ASoC: dpcm: Explicitly set BE DAI link supported stream directions
Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).

Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-07 17:57:48 +00:00
Jean-Francois Moine dcf0fa27a5 ASoC: pcm: Fix lack of platform bespoke_trigger() call
When the platform driver has no ops, the platform function
bespoke_trigger() is no more called.

The problem was introduced by the commit c5914b0aae
	"ASoC: pcm: Check for ops before deferencing them"

Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-07 17:11:07 +00:00
Lars-Peter Clausen 002220a90d ASoC: dpcm: Allow PCMs to omit the set of supported formats
Allow PCMs that do not impose any restrictions on the supported formats to set
the formats field to 0, Instead of assuming that this means that the PCM does
not support any formats (which doesn't make much sense), assume that it supports
all formats. This brings the behavior of DPCM closer to that of non-DPCM.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-06 17:42:42 +00:00
Lars-Peter Clausen 08ae9b456d ASoC: dpcm: Add helper function for initializing runtime pcm
We have the same code for initializing the runtime pcm on both the playback and
the capture path. Factor this out into a common helper function.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-06 17:36:48 +00:00
Mark Brown 729b47a007 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2014-01-02 13:01:50 +00:00
Mark Brown 07ac582c85 Merge remote-tracking branch 'asoc/topic/symmetry' into asoc-core
Conflicts (Trivial add/delete):
	sound/soc/soc-pcm.c
2013-12-04 11:54:29 +00:00
Nicolin Chen 7f62b6ee76 ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()
The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-04 11:46:40 +00:00
Nicolin Chen 0b4bbae85e ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close()
This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-04 11:45:29 +00:00
Lars-Peter Clausen 62e5f676f6 ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints
If there are symmetry constraints between the playback and the capture channel
set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-02 11:27:14 +00:00
Lars-Peter Clausen 78e45c99f6 ASoC: pcm: Always honor DAI min and max sample rate constraints
snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for
the PCM stream based on the rates specified in the rates field. Since we call
snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially
overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause
the minimum or maximum rate to be set to a value outside the range of one of the
components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or
SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via
SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To
fix this first calculate the minimum and maximum rates using
snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified
in the snd_soc_pcm_stream structs.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27 16:43:12 +00:00
Lars-Peter Clausen 17b6c19b34 ASoC: pcm: Fix rate_max calculation
In order to make sure that the sample rate is in the supported range of both
components the maximum rate of the card should be the minimum of the maximum
rate of each components. There is one special case to consider though, if
max_rate is set to 0 this means there is no maximum specified, so use
min_not_zero() macro which will give use the desired result.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27 16:42:37 +00:00
Nicolin Chen d3383420c9 ASoC: soc-pcm: move DAIs parameters cleaning into hw_free()
We're now applying soc_hw_params_symmetry() to reject unmatched parameters
while we clear parameters in soc_pcm_close(). So here's a use case might be
broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav

In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()
->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we
only clear parameters in soc_pcm_close(). The parameters would be remained
in the system even if the playback of 44100.wav is finished.

Thus, this patch is trying to move parameters cleaning into hw_free() so that
the system can continue to serve this kind of use case.

Also, since we set them in hw_params(), it should be better to clear them in
hw_free() for symmetry.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-24 13:32:50 +00:00
Nicolin Chen 3635bf09a8 ASoC: soc-pcm: add symmetry for channels and sample bits
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.

In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.

Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.

Thus, this patch applys symmetry for channels and sample bits.

Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-24 13:32:50 +00:00
Nicolin Chen 988e8cc41e ASoC: Add pinctrl PM to components of active DAIs
It's quite popular that more drivers are using pinctrl PM, for example:
(Documentation/devicetree/bindings/arm/primecell.txt). Just like what
runtime PM does, it would deactivate and activate pin group depending
on whether it's being used or not.

And this pinctrl PM might be also beneficial to cpu dai drivers because
they might have actual pinctrl so as to sleep their pins and wake them
up as needed.

To achieve this goal, this patch sets pins to the default state during
resume or startup; While during suspend and shutdown, it would set pins
to the sleep state.

As pinctrl PM would return zero if there is no such pinctrl sleep state
settings, this patch would not break current ASoC subsystem directly.

[ However, there is still an exception that the patch can not handle,
that is, when cpu dai driver does not have pinctrl property but another
device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example.
SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio
Multiplexer, has it). In this case, this kind of cpu dai driver needs to
find a way to obtain the pinctrl property as its own, by moving property
from AUDMUX to SSI, or creating a pins link/dependency between these two
devices, or using a more decent way after we figure it out. ]

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-04 09:08:59 -08:00
Russell King - ARM Linux 2062b4c5d2 ASoC: dpcm: improve robustness
Avoid oopsing if there is no backend stream associated with a front end
stream.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-10-31 10:57:47 -07:00
Mark Brown c5914b0aae ASoC: pcm: Check for ops before deferencing them
Ensure that we always check that an ops structure is present before we
try to use it, improving the robustness of the system.

Reported-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-10-31 10:57:47 -07:00
Jarkko Nikula 7cc302d231 ASoC: pcm: Remove extra spaces from dev_ prints
dev_ prints are already prefixed by ": " before format string so there is no
need for extra spaces.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-30 18:06:33 +01:00
Mark Brown 90d561bed9 Merge remote-tracking branch 'asoc/topic/fsl' into tmp 2013-09-01 21:15:52 +01:00
Fabio Estevam d6bead020d ASoC: soc-pcm: Allow to specify unidirectional dai_link
Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.

The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-01 16:29:27 +01:00
Mark Brown 60529c65ec Merge branch 'topic/core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-dapm 2013-07-29 18:40:55 +01:00
Lars-Peter Clausen c3f48ae6fd ASoC: dapm: Pass snd_soc_card directly to soc_dpcm_runtime_update()
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-07-24 14:56:19 +01:00
Mark Brown d4e1a73acd ASoC: pcm: Use the power efficient workqueue for delayed powerdown
There is no need to use a normal per-CPU workqueue for delayed power downs
as they're not timing or performance critical and waking up a core for them
would defeat some of the point.

Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Viresh Kumar <viresh.kumar@linaro.org>
2013-07-18 11:58:28 +01:00
Mark Brown 70fe99d8db Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2013-06-17 17:20:16 +01:00
Mark Brown 4616274d33 ASoC: dapm: Treat DAI widgets like AIF widgets for power
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.

Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.

Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2013-06-07 15:54:50 +01:00
Mark Brown 056790923e ASoC: pcm: Require both CODEC and CPU support when declaring stream caps
When declaring playback and capture capabilities check for both CODEC
side and CPU side support rather than only checking for CODEC side
support.  While it is unusual some CPUs do have unidirectional DAIs.

Reported-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2013-06-03 09:55:00 +01:00
Lars-Peter Clausen bd477c31ca ASoC: core: Add helper function to initialize the runtime pcm
We use the same code to initialize the runtime pcm based on the
snd_soc_pcm_stream struct on both the playback and capture path. Factor this
code into a helper function to make things a bit more tidy.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-05-14 16:52:48 +04:00
Lars-Peter Clausen 90996f43b3 ASoC: core: Move snd_soc_set_runtime_hwparams() to soc-pcm.c
snd_soc_set_runtime_hwparams() is the only PCM related function that lives in
soc-core.c. All other PCM related functions live in soc-pcm.c, so move
snd_soc_set_runtime_hwparams() over as well for a bit more consistency.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-05-14 16:52:43 +04:00
Mark Brown d289323286 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2013-02-11 11:06:30 +00:00
Mark Brown da18396f94 ASoC: core: Allow digital mute for capture
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.

The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.

Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
2013-02-08 11:08:44 +00:00
Dan Carpenter fffc0ca29f ASoC: pcm: delete some dead code
I've removed several unreachable returns.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:05:58 +00:00
Patrick Lai 08b27848da ASoC: pcm: allow backend hardware to be freed in pause state
When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.

Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-20 15:59:46 +00:00
Mark Brown 326b06a8a9 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2012-12-15 23:56:43 +09:00
Misael Lopez Cruz 9bffb1fb7c ASoC: Prevent pop_wait overwrite
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.

In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.

One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):

aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE

Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-15 23:45:09 +09:00
Mark Brown b1bc7b3c9c ASoC: pcm: Report error code when we fail to init platform
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-27 19:35:59 +00:00
Liam Girdwood 103d84a3cb ASoC: pcm: Standardise ASoC PCM messages
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.

Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-21 11:39:58 +09:00
Liam Girdwood 3ac3f5ca91 ASoC: dpcm: Allow FE to be opened without valid BE routes.
Some userspace will open a PCM device and then configure mixers
for routing before triggering. This patch allows userspace to do
this sequence.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-06 19:06:58 +01:00
Liam Girdwood 5cb9b74822 ASoC: pcm: Clean up logging in soc_new_pcm()
Use dev_ style logging throughout soc_new_pcm()

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-06 18:59:15 +01:00
Liam Girdwood 35ea0655a1 ASoC: dpcm: Fix dpcm_get_be() to check that DAI is BE
Make sure that the dpcm_get_be() only returns BE DAI links.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-06-08 06:56:38 +08:00
Mark Brown 45c0a188ca ASoC: pcm: Staticise non-exported functions
They pollute the global namespace and cause sparse to complain.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-05-10 10:37:19 +01:00
Mark Brown b3bba9a1a8 ASoC: pcm: Fix DPCM for aux_devs
When we instantiate an aux_dev we use a fake rtd as part of the process
which doesn't have a dai_link associated with it. Fix the dpcm startup
code to cope with this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-05-08 12:29:15 +01:00
Liam Girdwood f57b8488bc ASoC: dpcm: Fixup debugFS for DPCM state.
Remove writable debugFS permission, use simple_open() and
fix indentation.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:38:47 +01:00
Liam Girdwood be3f3f2ce6 ASoC: pcm: Add pcm operation for pcm ioctl.
Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.

This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:43 +01:00