profile/ivi/kernel-adaptation-intel-automotive.git
12 years agoALSA: snd-usb: Add quirks for Playback Designs devices
Daniel Mack [Tue, 4 Sep 2012 08:23:07 +0000 (10:23 +0200)]
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>
12 years agoALSA: USB: Support for (original) Xbox Communicator
Marko Friedemann [Mon, 3 Sep 2012 08:12:40 +0000 (10:12 +0200)]
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>
12 years agoALSA: fix possible memory leak in snd_mixer_oss_build_input()
Wei Yongjun [Sun, 2 Sep 2012 14:10:27 +0000 (22:10 +0800)]
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>
12 years agoALSA: Remove the last mention of SNDRV_MAIN_OBJECT_FILE
Josh Triplett [Mon, 3 Sep 2012 03:04:17 +0000 (20:04 -0700)]
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>
12 years agoALSA: hda - Clean up redundant FG checks
Takashi Iwai [Fri, 31 Aug 2012 14:54:38 +0000 (07:54 -0700)]
ALSA: hda - Clean up redundant FG checks

Just refactoring, no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Yet another fix for D3 stop-clock refcounting
Takashi Iwai [Fri, 31 Aug 2012 14:46:56 +0000 (07:46 -0700)]
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>
12 years agoALSA: cs5530: Fix resource leak in error path
Takashi Iwai [Thu, 30 Aug 2012 20:21:00 +0000 (13:21 -0700)]
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>
12 years agoALSA: korg1212: Fix reverted min/max ADC sense range
Takashi Iwai [Thu, 30 Aug 2012 14:57:38 +0000 (07:57 -0700)]
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>
12 years agoALSA: hda - Optimize bitfield usage in struct hda_codec
Takashi Iwai [Tue, 28 Aug 2012 23:39:25 +0000 (16:39 -0700)]
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>
12 years agoALSA: hda - Clean up CONFIG_SND_HDA_POWER_SAVE
Takashi Iwai [Fri, 24 Aug 2012 16:38:08 +0000 (18:38 +0200)]
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>
12 years agoALSA: hda - Fix D3 clock stop check for codecs with own set_power_state op
Takashi Iwai [Tue, 28 Aug 2012 16:59:20 +0000 (09:59 -0700)]
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>
12 years agoALSA: hda - Fix runtime PM leftover refcounts
Takashi Iwai [Tue, 28 Aug 2012 16:14:29 +0000 (09:14 -0700)]
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>
12 years agoALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substream
Takashi Iwai [Tue, 28 Aug 2012 23:27:26 +0000 (16:27 -0700)]
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>
12 years agoALSA: emu8000: fix emu8000 DRAM sized 512 KiB too small
David Flater [Tue, 28 Aug 2012 02:25:21 +0000 (22:25 -0400)]
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>
12 years agoMerge branch 'for-linus' into for-next
Takashi Iwai [Tue, 28 Aug 2012 16:26:59 +0000 (09:26 -0700)]
Merge branch 'for-linus' into for-next

Need to merge the fixes regarding EPSS.

Conflicts:
sound/pci/hda/hda_codec.c

12 years agoALSA: hda - Don't trust codec EPSS bit for IDT 92HD83xx & co
Takashi Iwai [Tue, 28 Aug 2012 16:20:13 +0000 (09:20 -0700)]
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>
12 years agoALSA: hda - Avoid unnecessary parameter read for EPSS
Takashi Iwai [Tue, 28 Aug 2012 16:18:01 +0000 (09:18 -0700)]
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>
12 years agoALSA: hda - Make clear built-in driver optimization
Takashi Iwai [Fri, 24 Aug 2012 16:40:10 +0000 (18:40 +0200)]
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>
12 years agoALSA: pcxhr: Add 8 new sound cards
Markus Bollinger [Fri, 24 Aug 2012 12:54:57 +0000 (14:54 +0200)]
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>
12 years agoALSA: cmi8328: Fix build error with CONFIG_GAMEPORT=n
Takashi Iwai [Fri, 24 Aug 2012 05:52:03 +0000 (07:52 +0200)]
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>
12 years agoALSA: hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
Mengdong Lin [Fri, 24 Aug 2012 04:06:30 +0000 (12:06 +0800)]
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>
12 years agoALSA: hda - add runtime PM support
Mengdong Lin [Thu, 23 Aug 2012 09:32:30 +0000 (17:32 +0800)]
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>
12 years agoALSA: hda - Call snd_hda_jack_report_sync() generically in hda_codec.c
Takashi Iwai [Wed, 22 Aug 2012 14:40:24 +0000 (16:40 +0200)]
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>
12 years agoALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakers
David Henningsson [Wed, 22 Aug 2012 14:10:43 +0000 (16:10 +0200)]
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>
12 years agoALSA: introduce snd-cmi8328: C-Media CMI8328 driver
Ondrej Zary [Mon, 20 Aug 2012 20:39:51 +0000 (22:39 +0200)]
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>
12 years agoALSA: snd-als100: fix suspend/resume
Ondrej Zary [Mon, 20 Aug 2012 19:50:13 +0000 (21:50 +0200)]
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>
12 years agoMerge branch 'for-linus' into for-next
Takashi Iwai [Mon, 20 Aug 2012 20:14:26 +0000 (22:14 +0200)]
Merge branch 'for-linus' into for-next

Conflicts:
sound/pci/hda/hda_codec.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Fix leftover codec->power_transition
Takashi Iwai [Mon, 20 Aug 2012 19:25:22 +0000 (21:25 +0200)]
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>
12 years agoMerge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound...
Takashi Iwai [Mon, 20 Aug 2012 19:26:04 +0000 (21:26 +0200)]
Merge tag 'asoc-3.6' of git://git./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.

12 years agoALSA: hda - Add missing ifdef CONFIG_SND_HDA_POWER_SAVE to tracepoints
Takashi Iwai [Mon, 20 Aug 2012 16:04:40 +0000 (18:04 +0200)]
ALSA: hda - Add missing ifdef CONFIG_SND_HDA_POWER_SAVE to tracepoints

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: lto, sound: Fix export symbols for !CONFIG_MODULES
Andi Kleen [Sun, 19 Aug 2012 02:56:22 +0000 (19:56 -0700)]
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>
12 years agoALSA: hda - Check the power state when power_save option is changed
Takashi Iwai [Tue, 14 Aug 2012 15:13:32 +0000 (17:13 +0200)]
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>
12 years agoALSA: hda - Implement snd_hda_power_sync() helper function
Takashi Iwai [Tue, 14 Aug 2012 15:12:47 +0000 (17:12 +0200)]
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>
12 years agoMerge branch 'topic/ca0132-fix' into for-linus
Takashi Iwai [Mon, 20 Aug 2012 09:38:31 +0000 (11:38 +0200)]
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.

12 years agoALSA: hda - don't create dysfunctional mixer controls for ca0132
David Henningsson [Mon, 20 Aug 2012 09:17:00 +0000 (11:17 +0200)]
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>
12 years agoALSA: hda - Add tracepoints at snd_hda_power_up/down entrances.
Takashi Iwai [Mon, 20 Aug 2012 08:22:25 +0000 (10:22 +0200)]
ALSA: hda - Add tracepoints at snd_hda_power_up/down entrances.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: snd-ad1816a: Implement suspend/resume
Ondrej Zary [Sun, 19 Aug 2012 21:27:26 +0000 (23:27 +0200)]
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>
12 years agoALSA: snd-ad1816a: remove useless struct snd_card_ad1816a
Ondrej Zary [Sun, 19 Aug 2012 21:27:19 +0000 (23:27 +0200)]
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>
12 years agoALSA: sound/ppc/snd_ps3.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:59 +0000 (09:02 +0200)]
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>
12 years agoALSA: sound/pci/rme9652/hdspm.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:54 +0000 (09:02 +0200)]
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>
12 years agoALSA: sound/pci/sis7019.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:55 +0000 (09:02 +0200)]
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>
12 years agoALSA: sound/pci/ctxfi/ctatc.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:56 +0000 (09:02 +0200)]
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>
12 years agoALSA: sound/atmel/ac97c.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:57 +0000 (09:02 +0200)]
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>
12 years agoALSA: sound/atmel/abdac.c: fix error return code
Julia Lawall [Sun, 19 Aug 2012 07:02:58 +0000 (09:02 +0200)]
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>
12 years agoALSA: fix pcm.h kernel-doc warning and notation
Randy Dunlap [Sun, 19 Aug 2012 00:43:05 +0000 (17:43 -0700)]
ALSA: fix pcm.h kernel-doc warning and notation

Fix kernel-doc warning in <sound/pcm.h> and add function name to make
the kernel-doc notation complete.

Warning(include/sound/pcm.h:1081): No description found for parameter 'substream'

Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agosound: oss/sb_audio: prevent divide by zero bug
Dan Carpenter [Sat, 18 Aug 2012 15:55:15 +0000 (18:55 +0300)]
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>
12 years agoASoC: wm9712: Fix inverted capture volume
Mark Brown [Tue, 31 Jul 2012 17:37:28 +0000 (18:37 +0100)]
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>
12 years agoASoC: wm9712: Fix microphone source selection
Mark Brown [Thu, 16 Aug 2012 21:36:04 +0000 (22:36 +0100)]
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
12 years agoASoC: wm5102: Remove DRC2
Mark Brown [Thu, 16 Aug 2012 12:08:23 +0000 (13:08 +0100)]
ASoC: wm5102: Remove DRC2

It will be removed from future device revisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
David Henningsson [Thu, 16 Aug 2012 12:11:09 +0000 (14:11 +0200)]
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>
12 years agoALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
Takashi Iwai [Wed, 15 Aug 2012 10:32:00 +0000 (12:32 +0200)]
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>
12 years agoALSA: hda - Add 3stack-automute model to AD1882 codec
Takashi Iwai [Mon, 13 Aug 2012 09:09:35 +0000 (11:09 +0200)]
ALSA: hda - Add 3stack-automute model to AD1882 codec

Added a simple support of automute for the front HP jack to AD1882
stack model.  Such an addition is basically an exception -- we really
want to avoid the static quirk codes, but AD1882 parser isn't still
ready for moving to the BIOS auto-parser yet.  So, as a quick fix, I
merged it for now.

In near future, we really need the big clean up of patch_analog.c to
move on to the auto-parser...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: PCI: Replace CONFIG_PM with CONFIG_PM_SLEEP
Takashi Iwai [Tue, 14 Aug 2012 16:12:04 +0000 (18:12 +0200)]
ALSA: PCI: Replace CONFIG_PM with CONFIG_PM_SLEEP

Otherwise we may get compile warnings due to unused functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Fix possible compile warnings regarding CONFIG_PM
Takashi Iwai [Tue, 14 Aug 2012 16:10:09 +0000 (18:10 +0200)]
ALSA: hda - Fix possible compile warnings regarding CONFIG_PM

Replace with a proper ifdef check of CONFIG_PM_SLEEP in hda_intel.c.
But other places in HD-audio driver are still marked with CONFIG_PM,
since these can be called for power-saving even without
CONFIG_PM_SLEEP.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: lx6464es: Add a missing error check
Takashi Iwai [Tue, 14 Aug 2012 15:42:11 +0000 (17:42 +0200)]
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>
12 years agoALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch
David Henningsson [Mon, 13 Aug 2012 15:10:46 +0000 (17:10 +0200)]
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>
12 years agoASoC: jack: Always notify full jack status
Mark Brown [Mon, 13 Aug 2012 15:28:36 +0000 (16:28 +0100)]
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>
12 years agoASoC: wm5110: Add missing input PGA routes
Mark Brown [Fri, 10 Aug 2012 14:40:22 +0000 (15:40 +0100)]
ASoC: wm5110: Add missing input PGA routes

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoASoC: wm5102: Add missing input PGA routes
Mark Brown [Fri, 10 Aug 2012 14:40:12 +0000 (15:40 +0100)]
ASoC: wm5102: Add missing input PGA routes

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoALSA: hda - fix Copyright debug message
Wang Xingchao [Mon, 13 Aug 2012 06:11:10 +0000 (14:11 +0800)]
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>
12 years agoALSA: hda - show ICT/KAE control bits
Wang Xingchao [Mon, 13 Aug 2012 07:43:49 +0000 (15:43 +0800)]
ALSA: hda - show ICT/KAE control bits

Enable two debug options for S/PDIF Converter Control.
KAE: Keep Alive Enable; ICT: IEC Coding Type.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoASoC: Samsung: Fix build error
Sachin Kamat [Wed, 8 Aug 2012 06:04:43 +0000 (11:34 +0530)]
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>
12 years agoALSA : hda - bug fix on checking the supported power states of a codec
Mengdong Lin [Fri, 10 Aug 2012 12:11:58 +0000 (14:11 +0200)]
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>
12 years agoALSA: hda - Fix panned "Beep Playback Switch"
David Henningsson [Fri, 10 Aug 2012 11:29:32 +0000 (13:29 +0200)]
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>
12 years agoALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()
Dan Carpenter [Fri, 10 Aug 2012 09:22:58 +0000 (12:22 +0300)]
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>
12 years agoMAINTAINERS: Add entries for Wolfson Arizona class devices
Mark Brown [Sat, 23 Jun 2012 10:25:43 +0000 (11:25 +0100)]
MAINTAINERS: Add entries for Wolfson Arizona class devices

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoASoC: core: Upgrade the severity of probe deferral errors to dev_err()
Mark Brown [Thu, 9 Aug 2012 17:44:37 +0000 (18:44 +0100)]
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>
12 years agoALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH
James Ralston [Thu, 9 Aug 2012 16:38:59 +0000 (09:38 -0700)]
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>
12 years agoMerge branch 'topic/hda-probe-defer' into for-next
Takashi Iwai [Thu, 9 Aug 2012 15:41:44 +0000 (17:41 +0200)]
Merge branch 'topic/hda-probe-defer' into for-next

Fix a build error when CONFIG_SND_HDA_PATCH_LOADER isn't set.

12 years agoALSA: hda - Fix forgotten ifdef CONFIG_SND_HDA_PATCH_LOADER
Takashi Iwai [Thu, 9 Aug 2012 15:40:46 +0000 (17:40 +0200)]
ALSA: hda - Fix forgotten ifdef CONFIG_SND_HDA_PATCH_LOADER

The firmware callback must be protected by that ifdef.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoMerge branch 'topic/hda-probe-defer' into for-next
Takashi Iwai [Thu, 9 Aug 2012 14:30:13 +0000 (16:30 +0200)]
Merge branch 'topic/hda-probe-defer' into for-next

This branch fixes the stall during probing the HD-audio driver when
the specified "patch" firmware doesn't exist.  It's basically a long-
standing issue, but mostly harmless until the recent rework of
firmware loader base code.

12 years agoALSA: hda - Deferred probing with request_firmware_nowait()
Takashi Iwai [Thu, 9 Aug 2012 11:49:23 +0000 (13:49 +0200)]
ALSA: hda - Deferred probing with request_firmware_nowait()

For processing the firmware handling properly for built-in kernels,
implement an asynchronous firmware loading with
request_firmware_nowait().  This means that the codec probing is
deferred when the patch option is specified.

Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Load firmware in hda_intel.c
Takashi Iwai [Thu, 9 Aug 2012 10:33:28 +0000 (12:33 +0200)]
ALSA: hda - Load firmware in hda_intel.c

This is a preliminary work for the deferred probing for
request_firmware() errors at init.

This patch moves the call of request_firmware() to hda_intel.c, and
call it in the earlier stage of probing rather than
azx_probe_continue().

Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM
Takashi Iwai [Thu, 9 Aug 2012 13:47:15 +0000 (15:47 +0200)]
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>
12 years agoASoC: wm8994: Add missing dapm routes for WM8958 rev A
Chris Rattray [Thu, 9 Aug 2012 09:10:54 +0000 (10:10 +0100)]
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>
12 years agoASoC: wm8962: Don't duplicate bias level management in resume
Mark Brown [Mon, 30 Jul 2012 17:23:35 +0000 (18:23 +0100)]
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>
12 years agoASoC: bfin: fix memory leak in sport3 controller driver
Scott Jiang [Thu, 9 Aug 2012 22:08:40 +0000 (18:08 -0400)]
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>
12 years agoASoC: Davinci: McASP: Flush the FIFO before enabling
Vaibhav Bedia [Wed, 8 Aug 2012 15:10:31 +0000 (20:40 +0530)]
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>
12 years agoALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V
David Henningsson [Thu, 9 Aug 2012 08:56:12 +0000 (10:56 +0200)]
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>
12 years agoALSA: hda - Add missing SPDIF I/O setup for CA0132
Takashi Iwai [Wed, 8 Aug 2012 15:26:54 +0000 (17:26 +0200)]
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>
12 years agoALSA: hda - Use the standard PCM ops for CA0132
Takashi Iwai [Wed, 8 Aug 2012 15:20:18 +0000 (17:20 +0200)]
ALSA: hda - Use the standard PCM ops for CA0132

Now with the workaround using codec->pcm_format_first flag, we can
clean up the home-baked codes in patch_ca0132.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items
Takashi Iwai [Wed, 8 Aug 2012 15:15:55 +0000 (17:15 +0200)]
ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Add codec->pcm_format_first flag
Takashi Iwai [Wed, 8 Aug 2012 15:12:52 +0000 (17:12 +0200)]
ALSA: hda - Add codec->pcm_format_first flag

Introduced a new flag to set up the PCM stream format at first before
the stream_id and channel tag.  Some codecs (e.g. CA0132) seem
preferring this over stream_id -> format order.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoASoC: imx-ssi: Remove mono support
Fabio Estevam [Tue, 7 Aug 2012 19:51:34 +0000 (16:51 -0300)]
ASoC: imx-ssi: Remove mono support

Playing a mono track results in incorrect playback rate, ie, the audio
is played at a faster rate.

Remove mono support in the driver by setting 'channes_min' to dual-channel
and this allows mono tracks to be played correctly.

Reported-by: Gaëtan Carlier <gcembed@gmail.com>
Tested-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoASoC: mxs: Fix the name of the SoC family
Fabio Estevam [Wed, 8 Aug 2012 03:47:21 +0000 (00:47 -0300)]
ASoC: mxs: Fix the name of the SoC family

SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
12 years agoALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad
David Henningsson [Wed, 8 Aug 2012 06:43:37 +0000 (08:43 +0200)]
ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad

The same ID is twice in the quirk table, so the second one is not used.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y
Takashi Iwai [Tue, 7 Aug 2012 16:09:23 +0000 (18:09 +0200)]
ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y

When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in
hda_auto_parser.c looks really ugly like:

  ALSA sound/pci/hda/hda_auto_parser.c:331    mono: mono_out=0x0
  ALSA sound/pci/hda/hda_auto_parser.c:334    dig-out=0x12/0x0
  ALSA sound/pci/hda/hda_auto_parser.c:335    inputs:
  ALSA sound/pci/hda/hda_auto_parser.c:339  Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339  Line=0x10
  ALSA sound/pci/hda/hda_auto_parser.c:341
  ALSA sound/pci/hda/hda_auto_parser.c:343    dig-in=0x13

Better to put one item at each line.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoASoC: omap-mcbsp: Fix 6pin mux configuration
Peter Ujfalusi [Tue, 7 Aug 2012 12:37:47 +0000 (15:37 +0300)]
ASoC: omap-mcbsp: Fix 6pin mux configuration

The check for the mux_signal callback was wrong which prevents us to
configure the 6pin port's FSR/CLKR signal mux.

Reported-by: CF Adad <cfadad@rocketmail.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org (3.4+)
12 years agoALSA: hda - remove redundant auto quirks for conexant 506x
David Henningsson [Tue, 7 Aug 2012 12:03:30 +0000 (14:03 +0200)]
ALSA: hda - remove redundant auto quirks for conexant 506x

Now that the auto model is the default, these quirks are redundant
and can be removed.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - remove quirk for Dell Vostro 1015
David Henningsson [Tue, 7 Aug 2012 12:03:29 +0000 (14:03 +0200)]
ALSA: hda - remove quirk for Dell Vostro 1015

This computer is confirmed working with model=auto on kernel 3.2.
Also, parsing fails with hda-emu with the current model.

Cc: stable@kernel.org (3.2+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - add dock support for Thinkpad X230
Felix Kaechele [Mon, 6 Aug 2012 21:02:01 +0000 (23:02 +0200)]
ALSA: hda - add dock support for Thinkpad X230

As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to
correctly set up the pins for the dock port.

Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - Fix regression of HDMI codec probing
Takashi Iwai [Mon, 6 Aug 2012 12:49:36 +0000 (14:49 +0200)]
ALSA: hda - Fix regression of HDMI codec probing

The commit c4bfe94a causes a regression on some codecs at probing.
Since this was just a workaround to shut up a kernel warning, it'd be
better to revert and fix properly.  So we ended up with re-adding the
cleanup callback.

Tested-and-reported-by: Matt Horan <matt@matthoran.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: hda - add dock support for Thinkpad T430s
Philipp A. Mohrenweiser [Mon, 6 Aug 2012 11:14:18 +0000 (13:14 +0200)]
ALSA: hda - add dock support for Thinkpad T430s

Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow
sound in docking station.

Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3

Cc: stable@kernel.org
Signed-off-by: Philipp A. Mohrenweiser <phiamo@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: print small buffers via %*ph[C]
Andy Shevchenko [Thu, 2 Aug 2012 13:52:41 +0000 (16:52 +0300)]
ALSA: print small buffers via %*ph[C]

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoMerge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound...
Mark Brown [Fri, 3 Aug 2012 22:01:54 +0000 (23:01 +0100)]
Merge tag 'asoc-3.6' of git://git./linux/kernel/git/broonie/sound into for-3.6

ASoC: Additional updates for 3.6

A few updates for issues discovered during the merge window, the main
one being the fix for the issues with defaulting to use of regmap
without properly checking if there was I/O in place already.

12 years agoMerge branch 'topic/next' into for-next
Takashi Iwai [Fri, 3 Aug 2012 10:59:38 +0000 (12:59 +0200)]
Merge branch 'topic/next' into for-next

12 years agoALSA: isa: Move snd_legacy_find_free_ioport to initval.h
Ondrej Zary [Wed, 1 Aug 2012 14:05:39 +0000 (16:05 +0200)]
ALSA: isa: Move snd_legacy_find_free_ioport to initval.h

Move snd_legacy_find_free_ioport() function back to initval.h as it is used
by two drivers.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: emu10k1: Avoid access to invalid pages when period=1
Takashi Iwai [Fri, 3 Aug 2012 10:51:21 +0000 (12:51 +0200)]
ALSA: emu10k1: Avoid access to invalid pages when period=1

When period=1, the driver tries to allocate a bit bigger buffer than
requested by the user due to the irq latency tolerance.  This may lead
to accesses over the actually allocated pages.

This patch adds a check of the page index and assigns the silent page
when it's over the given buffer size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoALSA: PCM: Fix possible memory leaks in the error path
Takashi Iwai [Fri, 3 Aug 2012 10:48:32 +0000 (12:48 +0200)]
ALSA: PCM: Fix possible memory leaks in the error path

When the first page allocation failed for sgbuf, it leaks the records
that have been formerly allocated.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
12 years agoLinux 3.6-rc1 v3.6-rc1
Linus Torvalds [Thu, 2 Aug 2012 23:38:10 +0000 (16:38 -0700)]
Linux 3.6-rc1