Commit Graph

13263 Commits

Author SHA1 Message Date
Wei Yongjun 3a4a7ef567 ALSA: opl4: use list_move_tail instead of list_del/list_add_tail
Using list_move_tail() instead of list_del() + list_add_tail().

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-05 09:14:45 +02:00
Daniel Mack 2b58fd5b31 ALSA: snd-usb: Add quirks for Playback Designs devices
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:

 - They need a 20ms delay after each class compliant request as the
   hardware ACKs the USB packets before the device is actually ready
   for the next command. Sending data immediately will result in buffer
   overflows in the hardware.
 - The devices send bogus feedback data at the start of each stream
   which confuse the feedback format auto-detection.

This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.

In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-04 11:31:14 +02:00
Marko Friedemann c05fce586d ALSA: USB: Support for (original) Xbox Communicator
Added support for Xbox Communicator to USB quirks.

Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-03 10:14:25 +02:00
Wei Yongjun 1f3b14072b ALSA: fix possible memory leak in snd_mixer_oss_build_input()
uinfo has been allocated in this function and should be
freed before leaving from the error handling cases.

spatch with a semantic match is used to found this problem.
(http://coccinelle.lip6.fr/)

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-03 10:08:28 +02:00
Josh Triplett a03e4a66c7 ALSA: Remove the last mention of SNDRV_MAIN_OBJECT_FILE
SNDRV_MAIN_OBJECT_FILE hasn't done anything since the pre-git days, and
the only remaining reference occurs as a #define in sound/last.c.  Drop
that last mention of it.

Signed-off-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-03 10:07:23 +02:00
Takashi Iwai d819387ef7 ALSA: hda - Clean up redundant FG checks
Just refactoring, no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 07:58:28 -07:00
Takashi Iwai 08fa20ae20 ALSA: hda - Yet another fix for D3 stop-clock refcounting
The call of pm_notify callback in snd_hda_codec_free() should be with
the check of the current state whether pm_notify(false) is called or
not, instead of codec->power_on check.

For improving the code readability and fixing this inconsistency,
codec->d3_stop_clk_ok is renamed to codec->pm_down_notified, and this
flag is set only when runtime PM down is called.  The new name reflects
to a more direct purpose of the flag.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 07:50:28 -07:00
Takashi Iwai 5a798394c8 ALSA: cs5530: Fix resource leak in error path
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44741

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 13:21:00 -07:00
Takashi Iwai fbaf6a5a35 ALSA: korg1212: Fix reverted min/max ADC sense range
k1212MinADCSens and k1212MaxADCSens are defined wrongly.
The max must be greater than the min by obvious reason.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46561

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:57:38 -07:00
Takashi Iwai 4b927345a3 ALSA: hda - Optimize bitfield usage in struct hda_codec
Move up a few bitfields to be packed into a single int.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:50:40 -07:00
Takashi Iwai 83012a7ccb ALSA: hda - Clean up CONFIG_SND_HDA_POWER_SAVE
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its
behavior can be well controlled via the default value and module
parameter.  Let's just replace it with the standard CONFIG_PM.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:50:13 -07:00
Takashi Iwai 432c641e01 ALSA: hda - Fix D3 clock stop check for codecs with own set_power_state op
When a codec provides its own set_power_state op, the D3-clock-stop
isn't checked correctly.  And the recent changes for repeating the
state-setting operation isn't applied to such a codec, too.

This patch fixes these issues by moving the call of codec's own op to
the place where the generic power-set operation is done, and move the
power-state synchronization code out of
snd_hda_set_power_state_to_all() so that it can be called always at
the end of power-up/down sequence, and updates the D3 clock-stop flag
properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:48:55 -07:00
Takashi Iwai 68467f51c1 ALSA: hda - Fix runtime PM leftover refcounts
When the HD-audio is removed, it leaves the refcounts when codecs are
powered up (usually yes) in the destructor.  For fixing the unbalance,
and cleaning up the code mess, this patch changes the following:
- change pm_notify callback to take the explicit power on/off state,
- check of D3 stop-clock and keep_link_on flags is moved to the caller
  side,
- call pm_notify callback in snd_hda_codec_new() and snd_hda_codec_free()
  so that the refcounts are proprely updated.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:48:49 -07:00
Takashi Iwai 48ee7cb8b4 ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substream
The two entries are duplicated in struct snd_usb_endpoint.
Seems forgotten in the last clean-up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 16:30:02 -07:00
David Flater 1338fc97d0 ALSA: emu8000: fix emu8000 DRAM sized 512 KiB too small
v2:  Fixed result still wrong in the case of 512 KiB DRAM.  Oops.

Applicable to 3.5.3 mainline.

In emu8000.c, size_dram determines the amount of memory on the sound card by
doing write/readback tests starting at 512 KiB and incrementing by 512 KiB.
On success, detected_size is updated to the successful address and testing
continues.  On failure, the loop is immediately exited.  The resulting
detected_size is 512 KiB too small except in two special cases:

1. If there is no memory, the initial 0 value of detected_size is used, which
   is correct.
2. If the address space wraps around, detected_size is updated before the
   bailout, so the result is correct.

The patch corrects all cases and was tested with an AWE64 Gold.  Before:
  EMU8000 [0x620]: 3584 Kb on-board memory detected
  asfxload 4GMGSMT.SF2 (4174814 B) fails.
After:
  EMU8000 [0x620]: 4096 Kb on-board memory detected
  asfxload 4GMGSMT.SF2 succeeds.

I do not have a card with 512 KiB to test with, but by forcibly enabling the
added conditional I verified on the AWE64 Gold that it detects 512 KiB
(successfully reading from the first memory location) and does not hang the
card.

C.f. Bug 46451 https://bugzilla.kernel.org/show_bug.cgi?id=46451

Signed-off-by: David Flater <dave@flaterco.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 19:58:12 +02:00
Takashi Iwai a184d4e459 Merge branch 'for-linus' into for-next
Need to merge the fixes regarding EPSS.

Conflicts:
	sound/pci/hda/hda_codec.c
2012-08-28 09:26:59 -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
Takashi Iwai 5d908ab941 ALSA: hda - Make clear built-in driver optimization
Use unsigned int to make clear that the codes required only for
modules will be reduced by the compiler optimization.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 09:12:09 -07:00
Markus Bollinger 8c3f1b1cbc ALSA: pcxhr: Add 8 new sound cards
add new sound cards VX442HR VX442e PCX442HR PCX442e VX822HR VX822e PCX822HR and PCX822e

Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-27 16:10:29 +02:00
Takashi Iwai 56244d0868 ALSA: cmi8328: Fix build error with CONFIG_GAMEPORT=n
sound/isa/cmi8328.c: In function 'snd_cmi8328_remove':
  sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function)
  sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in
  make[3]: *** [sound/isa/cmi8328.o] Error 1

Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-24 07:54:16 +02:00
Mengdong Lin 5d6147f101 ALSA: hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
The patch to support runtime PM introduced a bug:
Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk'
'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But
there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE.

This patch is to fix the bug.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-24 07:22:42 +02:00
Mengdong Lin b8dfc46241 ALSA: hda - add runtime PM support
Runtime PM can bring more power saving:
- When the controller is suspended, its parent device will also have a chance
  to suspend.
- PCI subsystem can choose the lowest power state the controller can signal
  wake up from. This state can be D3cold on platforms with ACPI PM support.
And runtime PM can provide a gerneral sysfs interface for a system policy
manager.

Runtime PM support is based on current HDA power saving implementation. The user
can enable runtime PM on platfroms that provide acceptable latency on transition
from D3 to D0.

Details:
- When both power saving and runtime PM are enabled:
  -- If a codec supports 'stop-clock' in D3, it will request suspending the
     controller after it enters D3 and request resuming the controller before
     back to D0. Thus the controller will be suspended only when all codecs are
     suspended and support stop-clock in D3.
  -- User IO operations and HW wakeup signal can resume the controller back to
     D0.
- If runtime PM is disabled, power saving just works as before.
- If power saving is disabled, the controller won't be suspended because the
  power usage counter can never be 0.

More about 'stop-clock' feature:
If a codec can support targeted pass-through operations in D3 state when there
is no BCLK present on the link, it will set CLKSTOP flag in the supported power
states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec
section 7.3.3.10 Power state and 7.3.4.12 Supported Power State.

[Fixed CONFIG_PM_RUNTIME dependency in hda_intel.c by tiwai]

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-23 14:21:32 +02:00
Takashi Iwai 8a5354140a ALSA: hda - Call snd_hda_jack_report_sync() generically in hda_codec.c
Instead of calling the jack sync in the init callback of each codec,
call it generically at initialization and resume.  By calling it at
the last of resume sequence, a possible race between the jack sync and
the unsol event enablement in the current code will be closed, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-22 16:48:17 +02:00
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 f993348746 ALSA: introduce snd-cmi8328: C-Media CMI8328 driver
Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as
AudioExcel AV500.

It supports PCM playback and capture (full-duplex) through wss_lib, gameport,
OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected
to the MPU401 port and Aux 1 input internally which works too.
The CDROM interface is not supported (as the drivers for these CDROMs were
removed from the kernel some time ago).

A separate driver is needed because CMI8328 is completely different chip to
CMI8329/CMI8330. It's configured by magic registers (there's no PnP). Sound is
provided by a real WSS codec (CS4231A) and the SB part is just a SB Pro
emulation (for DOS games, useless for Linux).

When SB is enabled, the CMI8328 chip disables access to the WSS codec,
emulates SoundBlaster on one side and outputs sound data to the codec - so SB
and WSS can't work together with this card. The WSS codec can do full duplex
by itself so there's no need for crazy things like snd-cmi8330 does
(combining SB and WSS parts into one driver).

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-21 07:30:46 +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 ddf83485d7 Merge branch 'for-linus' into for-next
Conflicts:
	sound/pci/hda/hda_codec.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 22:14:26 +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 099d53c308 ALSA: hda - Add missing ifdef CONFIG_SND_HDA_POWER_SAVE to tracepoints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 18:04:40 +02:00
Andi Kleen 8dea9d382a ALSA: lto, sound: Fix export symbols for !CONFIG_MODULES
The new LTO EXPORT_SYMBOL references symbols even without CONFIG_MODULES.
Since these functions are macros in this case this doesn't work.
Add a ifdef to fix the build.

Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:53:10 +02:00
Takashi Iwai 65fcd41d37 ALSA: hda - Check the power state when power_save option is changed
... by calling the newly introduced snd_hda_power_sync().

I had to reimplement a wheel for adding the trigger at changing the
parameter -- the parameter set ops is overwritten to pass the integer
parameter, then trigger the power-state sync.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:46:06 +02:00
Takashi Iwai c376e2c72b ALSA: hda - Implement snd_hda_power_sync() helper function
Added a new helper function snd_hda_power_sync() to trigger the
power-saving manually.  It's an inline function call to
snd_hda_power_save() helper function.

Together with this addition, snd_hda_power_up*() and
snd_hda_power_down() functions are inlined to a call of the same
snd_hda_power_save() helper function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:46:05 +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
Takashi Iwai b244d33560 ALSA: hda - Add tracepoints at snd_hda_power_up/down entrances.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:24:29 +02:00
Ondrej Zary 6f0fa66051 ALSA: snd-ad1816a: Implement suspend/resume
Implement suspend/resume support for AD1816 chips.
Tested with Terratec SoundSystem Base-1.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:12:56 +02:00
Ondrej Zary c86b6b452a ALSA: snd-ad1816a: remove useless struct snd_card_ad1816a
struct snd_card_ad1816a is only set but the values are never used then.
Removing it allows struct snd_card's private_data to be used for
struct snd_ad1816a, simplifying the code.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 11:10:39 +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