Commit Graph

13231 Commits

Author SHA1 Message Date
Matteo Frigo
3737e2be50 ALSA: ice1724: Use linear scale for AK4396 volume control.
The AK4396 DAC has a linear-scale attentuator, but
sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is
not quite right.  This patch restores the correct scale, borrowing
from the ak4396 code in sound/pci/oxygen/oxygen.c.

Signed-off-by: Matteo Frigo <athena@fftw.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-12 16:17:41 +02:00
Catalin Iacob
c302d6133c ALSA: hda_intel: add position_fix quirk for Asus K53E
Commit c20c5a841c changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.

My hardware is Cougar Point which the commit log of
c20c5a841c mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.

Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-11 14:28:45 +02:00
Dan Carpenter
81cb324675 ALSA: compress_core: fix open flags test in snd_compr_open()
O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always
false and it will never do compress capture.  The test for O_WRONLY is
also slightly off.  The original test would consider "->flags =
(O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid.

I've also removed the pr_err() because that could flood dmesg.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-11 14:27:40 +02:00
Takashi Iwai
07dc59f098 ALSA: hda - Fix Oops at codec reset/reconfig
snd_hda_codec_reset() calls restore_pincfgs() where the codec is
powered up again, which eventually tries to resume and initialize via
the callbacks of the codec.  However, it's the place just after codec
free callback, thus no codec callbacks should be called after that.
On a codec like CS4206, it results in Oops due to the access in init
callback.

This patch fixes the issue by clearing the codec callbacks properly
after freeing codec.

Reported-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-10 10:26:23 +02:00
Takashi Iwai
1213a205f9 ALSA: usb-audio: Fix bogus error messages for delay accounting
The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
  delay: estimated 0, actual 352
  delay: estimated 353, actual 705

These come from the sanity check in retire_playback_urb().  Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent.  And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.

For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.

Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06 15:00:15 +02:00
Takashi Iwai
ab548d2dba ALSA: hda - Fix missing Master volume for STAC9200/925x
With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with
vmaster hook in patch_sigmatel.c], the former Master volume control
was converted to PCM.  This was supposed to be covered by the vmaster
control.  But due to the lack of "PCM" slave definition, this didn't
happen properly.  The patch fixes the missing entry.

Reported-by: Andrew Shadura <bugzilla@tut.by>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06 10:10:11 +02:00
Daniel Mack
2e4a263ca8 ALSA: snd-usb: fix cross-interface streaming devices
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.

Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:04:53 +02:00
Daniel Mack
245baf983c ALSA: snd-usb: fix calls to next_packet_size
In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.

However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.

As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.

Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:48 +02:00
Daniel Mack
fbcfbf5f67 ALSA: snd-usb: restore delay information
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.

This patch adds them back, restoring the correct delay information
behaviour.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:08 +02:00
Pavel Roskin
03d2f44e96 ALSA: snd-usb: use list_for_each_safe for endpoint resources
snd_usb_endpoint_free() frees the structure that contains its argument.

Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 18:17:45 +02:00
Daniel Mack
015618b902 ALSA: snd-usb: Fix URB cancellation at stream start
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.

Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:46:27 +02:00
Takashi Iwai
c36b5b054a ALSA: hda - Don't trust codec EPSS bit for IDT 92HD83xx & co
These codecs seem reporting EPSS but require longer delay for the
proper D3 transition.  For example, D3_STOP_CLOCK_OK bit won't be set
correctly even after D3.

In this patch, codec->epss flag is overridden for avoid the
misbehavior.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 09:26:16 -07:00
Takashi Iwai
983f6b9381 ALSA: hda - Avoid unnecessary parameter read for EPSS
EPSS parameter should be static, so we can read it once and remember.
This also allows more easily to override the wrong EPSS capability
reported from a codec by changing the flag in the codec
initialization step.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 09:25:57 -07:00
David Henningsson
042b92c185 ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakers
This fixes an issue with a machine where there were no speakers,
but GPIO0 had to be data=1 for the headphone to be functioning.

I'm not sure if we need a more advanced patch to solve all possible cases,
but if so, this patch would still provide a minor optimisation.

BugLink: https://bugs.launchpad.net/bugs/1040077
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-22 16:26:05 +02:00
Ondrej Zary
53e1719f3d ALSA: snd-als100: fix suspend/resume
snd_card_als100_probe() does not set pcm field in struct snd_sb.
As a result, PCM is not suspended and applications don't know that they need
to resume the playback.

Tested with Labway A381-F20 card (ALS120).

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-21 07:29:40 +02:00
Takashi Iwai
535b6c51fe ALSA: hda - Fix leftover codec->power_transition
When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.

Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 21:25:22 +02:00
Takashi Iwai
f0b433e9f3 ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
 unremarkable in a global context.  The biggest batch are for the newly
 added Arizona drivers.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Additional updates for 3.6

A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context.  The biggest batch are for the newly
added Arizona drivers.
2012-08-20 21:26:04 +02:00
Takashi Iwai
fa2f5bf096 Merge branch 'topic/ca0132-fix' into for-linus
This is a series of fixes for CA0132, especially the missing SPDIF I/O
and the mixer build errors.
2012-08-20 11:38:31 +02:00
David Henningsson
c41999a239 ALSA: hda - don't create dysfunctional mixer controls for ca0132
It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:33:23 +02:00
Julia Lawall
c86b93628e ALSA: sound/ppc/snd_ps3.c: fix error return code
Initialize ret before returning on failure, as done elsewhere in the
function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:01:14 +02:00
Julia Lawall
b17cbdd85f ALSA: sound/pci/rme9652/hdspm.c: fix error return code
Convert a nonnegative error return code to a negative one, as returned
elsewhere in the function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:00:51 +02:00
Julia Lawall
ae970eb45d ALSA: sound/pci/sis7019.c: fix error return code
Initialize rc before returning on failure, as done elsewhere in the
function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 10:57:51 +02:00
Julia Lawall
4d8ce1c996 ALSA: sound/pci/ctxfi/ctatc.c: fix error return code
Initialize err before returning on failure, as done elsewhere in the
function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 10:57:30 +02:00
Julia Lawall
0c23e46eb4 ALSA: sound/atmel/ac97c.c: fix error return code
In the first case, the second test of whether retval is negative is
redundant.  It is dropped and the previous and subsequent tests are
combined.

In the second case, add an initialization of retval on failure of ioremap.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 10:56:01 +02:00
Julia Lawall
aaf265c22e ALSA: sound/atmel/abdac.c: fix error return code
Initialize retval before returning from a failed call to ioremap.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 10:53:13 +02:00
Dan Carpenter
94f3ec6b22 sound: oss/sb_audio: prevent divide by zero bug
Speed comes from get_user() in audio_ioctl().  We use it to set the "s"
variable before clamping it to valid values so it could lead to a divide
by zero bug.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 10:24:21 +02:00
Mark Brown
28c42c2830 ASoC: wm9712: Fix inverted capture volume
The capture volume increases with the register value so it shouldn't be
flagged as inverted.

Reported-by: Christoph Fritz <chf.fritz@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-17 22:43:18 +01:00
Mark Brown
ccf795847a ASoC: wm9712: Fix microphone source selection
Currently the microphone input source is not selectable as while there is
a DAPM widget it's not connected to anything so it won't be properly
instantiated. Add something more correct for the input structure to get
things going, even though it's not hooked into the rest of the routing
map and so won't actually achieve anything except allowing the relevant
register bits to be written.

Reported-by: Christop Fritz <chf.fritz@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-08-17 22:42:14 +01:00
Mark Brown
939d5044b1 ASoC: wm5102: Remove DRC2
It will be removed from future device revisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-17 22:38:27 +01:00
David Henningsson
5e68fb3cab ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
Instead of blindly initializing a volume knob widget, first check
that there actually is a volume knob widget.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-16 14:14:56 +02:00
Takashi Iwai
e9ba389c5f ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
A PCM capture stream on usb-audio causes a scheduling-while-atomic
BUG, as reported in the bugzilla entry below.  It's because
snd_usb_endpoint_start() is called at first at trigger START for a
capture stream, and this function contains the left-over EP
deactivation codes.  The problem doesn't happen for a playback stream
because the function is called at PCM prepare time, which can sleep.

This patch fixes the BUG by moving the EP deactivation code into the
PCM prepare callback.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-16 08:04:07 +02:00
Takashi Iwai
3bdcff70b6 ALSA: lx6464es: Add a missing error check
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-14 17:42:11 +02:00
David Henningsson
265d931a9e ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch
Some Conexant devices (e g CX20590) have no mute capability on
their Beep widgets.
This patch makes sure we don't try setting mutes on those widgets.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-14 10:22:31 +02:00
Mark Brown
12022a7853 ASoC: jack: Always notify full jack status
Don't just notify for the bits we've updated, notify the full state of the
jack otherwise users might get confused by misleading reports.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13 20:47:58 +01:00
Mark Brown
17c3f7e8f3 ASoC: wm5110: Add missing input PGA routes
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13 13:27:30 +01:00
Mark Brown
14ebd8a8c1 ASoC: wm5102: Add missing input PGA routes
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13 13:27:29 +01:00
Wang Xingchao
088c820b73 ALSA: hda - fix Copyright debug message
As spec said, 1 indicates no copyright is asserted.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-13 10:02:01 +02:00
Sachin Kamat
61f5d61ef9 ASoC: Samsung: Fix build error
Fixes the following build error:
In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0,
		from arch/arm/plat-samsung/include/plat/dma-ops.h:17,
		from arch/arm/plat-samsung/include/plat/dma.h:128,
		from sound/soc/samsung/pcm.c:23:
arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8:
			error: redefinition of ‘struct s3c2410_dma_client’
arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here
make[3]: *** [sound/soc/samsung/pcm.o] Error 1

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-10 18:07:20 +01:00
Mengdong Lin
e037cb4a54 ALSA : hda - bug fix on checking the supported power states of a codec
The return value of snd_hda_param_read() is -1 for an error, otherwise
it's the supported power states of a codec.

The supported power states is a 32-bit value. Bit 31 will be set to 1
if the codec supports EPSS, thus making "sup" negative. And the bit
28:5 is reserved as "0".
So a negative value other than -1 shall be further checked.

Please refer to High-Definition spec 7.3.4.12 "Supported Power
States", thanks!

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10 14:11:58 +02:00
David Henningsson
14bc9c6dc6 ALSA: hda - Fix panned "Beep Playback Switch"
When "Beep Playback Switch" had a different value on left and right
channels (such as muting left but not right, or vice versa), this
could result in the right channel being ignored.

This patch enables beep to be sounding from right channel only, and
also give correct result back to userspace (e g amixer).

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10 14:10:20 +02:00
Dan Carpenter
de64c0ee7d ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()
This function returns its own error codes instead of normal negative
error codes.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10 12:11:21 +02:00
Mark Brown
fb099cb712 ASoC: core: Upgrade the severity of probe deferral errors to dev_err()
In the past when ASoC had a custom probe deferral mechanism people
complained about the logspam it generated and didn't want to know about
the fact that we were doing probe deferral so all the error messages for
it were at dev_dbg(), making diagnostics hard. Now that we have probe
deferral as an accepted thing and it's generating log messages anyway
there's no need to worry about this so upgrade the severity of all the
probe deferral sources to dev_err() so that they are displayed by default.

Also add one for missing aux_devs since there wasn't one.

Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09 19:34:04 +01:00
James Ralston
144dad99ef ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH
This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH

Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09 18:42:42 +02:00
Takashi Iwai
d34e4e00ad ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM
When CONFIG_PM is set but CONFIG_PM_SLEEP is unset,
SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to
compile warnings.

For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM.

Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09 15:47:15 +02:00
Chris Rattray
15676937e6 ASoC: wm8994: Add missing dapm routes for WM8958 rev A
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09 14:21:47 +01:00
Mark Brown
52c0eee332 ASoC: wm8962: Don't duplicate bias level management in resume
The core will bring the bias level up for us since we use idle_bias_off,
duplicating this may be harmful.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09 14:11:10 +01:00
Scott Jiang
8b5eae137b ASoC: bfin: fix memory leak in sport3 controller driver
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09 14:08:59 +01:00
Vaibhav Bedia
0d62427572 ASoC: Davinci: McASP: Flush the FIFO before enabling
FIFO should be flushed before it is enabled for the first time.
This fixes the I/O errors reported by the ASoC core on a fresh boot

Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09 14:05:47 +01:00
David Henningsson
94c142a160 ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V
To turn off pin control for the pin was tested, and helped against
this issue.

BugLink: https://bugs.launchpad.net/bugs/1034779
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09 11:00:40 +02:00
Takashi Iwai
8e13fc1c5f ALSA: hda - Add missing SPDIF I/O setup for CA0132
CA0132 driver had some codes to handle the S/PDIF I/O, but the actual
setups of pins and converters were missing.  Now the pins are added.

Also, fixed a few points triggering invalid codec verbs and mixer
elements since the digital I/O audio widgets on CA0132 have no amp.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-08 17:27:50 +02:00