Commit graph

13247 commits

Author SHA1 Message Date
Takashi Iwai
5d037f9064 ASoC: Updates for 3.6
A bigger set of updates than I'm entirely comfortable with - things
 backed up a bit due to travel.  As ever the majority of these are small,
 focused updates for specific drivers though there are a couple of core
 changes.  There's been good exposure in -next.
 
 The AT91 patch fixes a build break.
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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: Updates for 3.6

A bigger set of updates than I'm entirely comfortable with - things
backed up a bit due to travel.  As ever the majority of these are small,
focused updates for specific drivers though there are a couple of core
changes.  There's been good exposure in -next.

The AT91 patch fixes a build break.
2012-09-15 08:24:42 +02:00
Bo Shen
985b11fa80 ASoC: wm8904: correct the index
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-09-14 18:02:38 +01:00
Takashi Iwai
64f1e00d8e ALSA: hda - Yet another position_fix quirk for ASUS machines
ASUS X53S also suffers from the same issue as in commit c302d6133.
Use POS_FIX_POSBUF for this hardware, too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=47461

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-13 16:56:13 +02:00
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
Stephen Warren
a32826e4ae ASoC: tegra: fix maxburst settings in dmaengine code
The I2S controllers are programmed with an "attention" level of 4 DWORDs.
This must match the configuration passed to the DMA driver, so that when
they burst in data, they don't overflow the available FIFO space. Also,
the burst size is relevant to the destination for playback, and source
for capture, not vice-versa as originally written.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-09-07 09:52:02 +08:00
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
Dylan Reid
57b2d68863 ASoC: samsung dma - Don't indicate support for pause/resume.
The pause and resume operations indicate that the stream can be
un-paused/resumed from the exact location they were paused/suspended.
This is not true for this driver, the pause and suspend triggers share
the same code path with stop, they flush all pending DMA transfers.
This drops all pending samples.  The pause_release/resume triggers are
the same as start, except that prepare won't be called beforehand,
nothing will be enqueued to the DMA engine and nothing will happen (no
audio).  Removing the pause flag will let apps know that it isn't
supported.  Removing the resume flag will cause user space to call
prepare and start instead of resume, so audio will continue playing when
the system wakes up.

Before removing the pause and resume flags, I tested this on an exynos
5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a
write error.  Suspend/resume testing led to the same result.  Removing
the two flags fixes suspend/resume (since snd_pcm_prepare is called
again). And leads to a proper reporting of pause not supported.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-09-06 18:55:59 +08:00
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
Fabio Estevam
37f45cc54c ASoC: mc13783: Remove mono support
Playing a mono track on a mc13783 codec results in incorrect playback rate.

Remove mono support so that a mono track can be played correctly.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-09-06 08:17:12 +08:00
Heather Lomond
4758be37c0 ASoC: arizona: Fix typo in 44.1kHz rates
Signed-off-by: Heather Lomond <hlomond@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-09-06 06:58:44 +08:00
Prasad Joshi
fd4fb262b3 ASoC: spear: correct the check for NULL dma_buffer pointer
The if condition
	if (!buf && !buf->area)

checks if the buf pointer is NULL and then dereferences it again to
check if the buffer area is NULL, resulting in possible NULL
dereference.

Signed-off-by: Prasad Joshi <prasadjoshi.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-31 14:24:52 -07:00
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
Stephen Warren
c921928661 sound: tegra_alc5632: remove HP detect GPIO inversion
Both the schematics and practical testing show that the HP detect GPIO
is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio
should not specify to invert the signal.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # v3.4 v3.5
2012-08-28 10:14:01 -07:00
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
Mark Brown
4e872a4682 ASoC: dapm: Don't force card bias level to be updated
Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means
that any DAPM context being updated will have the bias level automatically
set, including the card. We can't safely do this as the card callbacks are
called for each device context and so the management of the card bias is
more complex. Several multi-component cards rely on this behaviour.

Skip updates during the asynchronous run entirely. We should really do them
in the synchronous section but it's not 100% clear which values to pick as
the different DAPM contexts may have different bias levels.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-25 13:51:09 +01:00
Mark Brown
d8c3bb911f ASoC: dapm: Make sure we update the bias level for CODECs with no op
Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated)
ensures that we update non-CODEC DAPM contexts but means that if a
CODEC has no set_bias_level() operation it'll not be updated. Fix
that.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-25 13:50:21 +01:00
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
Julia Lawall
bc72d26bdb ASoC: am3517evm: fix error return code
It was forgotten to initialize ret to the result of calling
snd_soc_dai_set_sysclk, unlike at the other calls in the same 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>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-20 20:44:36 +01:00
Julia Lawall
b18e93a493 ASoC: ux500_msp_i2s: better use devm functions and fix error return code
Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap
(and use resource_size for the third argument).  These changes make it
possible to remove the error-handling code at the end of
ux500_msp_i2s_init_msp, and all of the gotos become direct returns.

In the case of the second call to devm_kzalloc, the return variable ret was
not initialized.  Here it is changed to a direct return of -ENOMEM.

A simplified version of the semantic match that finds the second 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-20 20:44:28 +01:00
Julia Lawall
db8b624d55 ASoC: imx-sgtl5000: fix error return code
Initialize ret on the second call to imx_audmux_v2_configure_port so that
the subsequent test checks that result and not the previous one.

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: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-20 20:44:19 +01:00
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