Commit Graph

7601 Commits

Author SHA1 Message Date
Liam Girdwood
d8b55d2cd0 ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
Fix build warning about unused ops and add ops
to the sdp4430 DAI link.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-19 14:14:51 +01:00
Jorge Eduardo Candelaria
871a05a78b ASoC: TWL6040: Enable earphone path in codec
Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-19 14:07:23 +01:00
Jorge Eduardo Candelaria
7254e2bddc ASoC: SDP4430: Add support for Earphone speaker
Enable earphone speaker in sdp4430 machine driver.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-19 14:07:15 +01:00
Misael Lopez Cruz
5e64d6aadd ASoC: SDP4430: Add sdp4430 machine driver
Add ASoC support for TI SDP4430.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-19 14:04:40 +01:00
Grant Likely
61c7a080a5 of: Always use 'struct device.of_node' to get device node pointer.
The following structure elements duplicate the information in
'struct device.of_node' and so are being eliminated.  This patch
makes all readers of these elements use device.of_node instead.

(struct of_device *)->node
(struct dev_archdata *)->prom_node (sparc)
(struct dev_archdata *)->of_node (powerpc & microblaze)

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
2010-05-18 16:10:44 -06:00
Linus Torvalds
1014cfe2fb Merge branch 'core-locking-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip
* 'core-locking-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
  lockdep: Reduce stack_trace usage
  lockdep: No need to disable preemption in debug atomic ops
  lockdep: Actually _dec_ in debug_atomic_dec
  lockdep: Provide off case for redundant_hardirqs_on increment
  lockdep: Simplify debug atomic ops
  lockdep: Fix redundant_hardirqs_on incremented with irqs enabled
  lockstat: Make lockstat counting per cpu
  i8253: Convert i8253_lock to raw_spinlock
2010-05-18 08:17:35 -07:00
Dan Williams
0b28330e39 Merge branch 'ioat' into dmaengine 2010-05-17 16:30:58 -07:00
Linus Walleij
058276303d DMAENGINE: extend the control command to include an arg
This adds an argument to the DMAengine control function, so that
we can later provide control commands that need some external data
passed in through an argument akin to the ioctl() operation
prototype.

[dan.j.williams@intel.com: fix up some missed conversions]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
2010-05-17 16:30:42 -07:00
Geert Uytterhoeven
ff2db7c5ab m68k: amiga - Sound platform device conversion
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
2010-05-17 21:37:44 +02:00
Peter Ujfalusi
2d4cdd6fc9 ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-17 20:34:15 +01:00
Felipe Balbi
7fd1d74bfc ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.

Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-17 20:34:10 +01:00
Julia Lawall
550a8b691c ALSA: sound/pci/asihpi: Use kzalloc
Use kzalloc rather than the combination of kmalloc and memset.

The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression x,size,flags;
statement S;
@@

-x = kmalloc(size,flags);
+x = kzalloc(size,flags);
 if (x == NULL) S
-memset(x, 0, size);
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:12:44 +02:00
Wu Fengguang
3eaead579e ALSA: hdmi - dont fail on extra nodes
The number of HDMI nodes is expected to go up in future.
So don't fail hard on seeing extra converter/pin nodes.

We can still operate safely on the nodes within
MAX_HDMI_CVTS/MAX_HDMI_PINS.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:12:13 +02:00
Wu Fengguang
e48b00870f ALSA: intelhdmi - add id for the CougarPoint chipset
Add id for Intel CougarPoint HDMI audio codec.

CougarPoint provides 3 Audio Converters.
Increase MAX_HDMI_CVTS/MAX_HDMI_PINS accordingly.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:11:59 +02:00
Wu Fengguang
41da2e0a01 ALSA: intelhdmi - user friendly codec name
Use the full chipset codename as codec name.
They are more user friendly than the spec abbrs.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:11:48 +02:00
Wu Fengguang
e9abf85fe1 ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
This is necessary to support >=3 HDMI playback devices
starting from the CougarPoint codec.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:11:39 +02:00
Dan Carpenter
1be1d76b8a ALSA: asihpi: incorrect range check
The entity_type_to_size[] array has LAST_ENTITY_TYPE (11) number of elements,
not LAST_ENTITY_ROLE (17).  This only affects the debug output.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:10:34 +02:00
Dan Carpenter
2448b14715 ALSA: asihpi: testing the wrong variable
There is a typo here.  We want to test "*dst" not "dst".

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:10:13 +02:00
Dan Carpenter
b0fb75ad5c ALSA: es1688: add pedantic range checks
Smatch complains that if (dev == SNDRV_CARDS) we're one past the end of
the array.  That's unlikely to happen in real life, I suppose.

Also smatch complains about "strcpy(card->shortname, pcm->name);"
The "pcm->name" buffer is 80 characters and "card->shortname" is 32
characters.  If you follow the call paths it turns out we never actually
use more than 16 characters so it's not a problem.  But anyway, let's
make it easy for people auditing this in the future.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-17 08:09:51 +02:00
Sergey Lapin
d98508a121 OMAP: McBSP: Add 32-bit mode support
This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.

Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-14 11:14:24 +01:00
Takashi Iwai
105ce39ca4 Merge branch 'fix/hda' into for-linus 2010-05-13 10:07:15 +02:00
Takashi Iwai
8213466596 ALSA: ice1724 - Fix ESI Maya44 capture source control
The capture source control of maya44 was wrongly coded with the bit
shift instead of the bit mask.  Also, the slot for line-in was
wrongly assigned (slot 5 instead of 4).

Reported-by: Alex Chernyshoff <alexdsp@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-12 16:43:32 +02:00
Peter Ujfalusi
36aeff6146 ASoC: TWL4030: Add control for digimic Left Right swap
The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-12 09:58:26 +01:00
Takashi Iwai
9fe17b5d47 ALSA: pcm - Use pgprot_noncached() for MIPS non-coherent archs
MIPS non-coherent archs need the noncached pgprot in mmap of PCM buffers.
But, since the coherency needs to be checked dynamically via
plat_device_is_coherent(), we need an ugly check dependent on MIPS
in ALSA core code.

This should be cleaned up in MIPS arch side (e.g. creating
dma_mmap_coherent()) in near future.

Tested-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-12 10:32:42 +02:00
Clemens Ladisch
6a45f78225 ALSA: virtuoso: fix Xonar D1/DX front panel microphone
Commit 65c3ac885c in 2.6.33 accidentally
left out the initialization of the AC97 codec FMIC2MIC bit, which broke
recording from the front panel microphone.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-12 10:28:36 +02:00
Takashi Iwai
2a6ce6e5fd ALSA: hda - Add hp-dv4 model for IDT 92HD71bx
It turned out that HP dv series have inconsistent the mute-LED GPIO
mapping among various models.  dv4/7 seem to use GPIO 0 while dv 5/6
seem to use GPIO 3.  The previous commit
  26ebe0a289
  ALSA: hda - Fix mute-LED GPIO pin for HP dv series
breaks dv5/6.

This patch adds the new quirk model, hp-dv4, to handle HP dv4/7
separately from HP dv5/6.

Tested-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com> (for dv6-1110ax)
Acked-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-12 10:20:42 +02:00
Daniel Mack
e213e9cf70 ALSA: sound/usb: add preliminary support for UAC2 interrupts
For both UAC1 and UAC2, interrupt endpoint messages are now parsed with
structs rather that with anonymous buffer array accesses.

For UAC2, only CUR interrupt notifications are supported for now.

snd_usb_mixer_status_complete() was renamed to
snd_usb_mixer_interrupt().

Fixed one indentation flaw on the way.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-11 22:44:07 +02:00
Haojian Zhuang
baffe1699c [ARM] pxa: add namespace on ssp
In order to prevent code ambiguous, add namespace on functions in ssp driver.

Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2010-05-11 17:25:06 +02:00
Eric Miao
866d091dcb [ARM] pxa: remove incorrect select PXA_SSP in Kconfig
PXA_SSP is actually used by drivers like drivers/spi/pxa2xx_spi.c and
sound/soc/pxa/pxa-ssp.c, not by boards. Remove those incorrect 'select'
from Kconfig and make SOC_PXA_SSP to select.

Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2010-05-11 17:24:58 +02:00
Haojian Zhuang
54c39b420f [ARM] pxa: move ssp into common plat-pxa
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2010-05-11 17:24:58 +02:00
Eric Miao
83f2889643 [ARM] pxa: merge regs-ssp.h into ssp.h
No need to separate them as they should be together from the begining.

Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2010-05-11 17:24:58 +02:00
Mark Brown
6a2f1ee1f9 ASoC: Don't restart unconfigured WM8994 FLLs
If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-11 14:18:52 +01:00
Mark Brown
6adb26bd03 ASoC: Reorder power down sequence for WM hubs devices
Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-11 14:18:41 +01:00
Mark Brown
3254d28500 ASoC: Add additional WM hubs DC servo trace
Log the values we're getting back from the DC servo and the values we
write to it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-11 14:18:34 +01:00
Mark Brown
fd5722e5cd ASoC: Add register write logging for WM8994
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-11 14:18:18 +01:00
Jaroslav Kysela
f48f606d9f [ALSA] snd-hda-intel: Improve azx_position_ok()
Add back the zero return value (activate workqueue) when
bdl_pos_adj is nonzero for position check.

Do the position related check only for first next period
using wallclk counter.

Return -1 value (ignore interrupt) when period_bytes
variable is zero.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-05-11 12:17:55 +02:00
Peter Ujfalusi
d11bb4a925 ASoC: core: Fix for the volume limiting when invert is in use
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-11 09:34:11 +01:00
Jaroslav Kysela
e54637205b [ALSA] snd-hda-intel: use WALLCLK register to check for early irqs
Use 24Mhz WALLCLK register to ignore too early interrupts and
wrong interrupt status. The bad timing confuses the higher ALSA
layer and causes audio skipping. More information about behaviour
and debugging can be found in kernel bz#15912.

https://bugzilla.kernel.org/show_bug.cgi?id=15912

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-05-11 10:21:46 +02:00
Takashi Iwai
26ebe0a289 ALSA: hda - Fix mute-LED GPIO pin for HP dv series
Old HP dv series seem to use the GPIO pin 0 for controlling the mute LED
although the pin is a large package, where the newer models use GPIO 3
in such a case.  For fixing the regression from the previous kernels,
set spec->gpio_led statically for these model quirks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-11 08:36:29 +02:00
Shahin Ghazinouri
beaffc3993 ALSA: hda - Fixes distorted recording on US15W chipset
The HDA controller in US15W (Poulsbo) reports inaccurate position values
for capture streams when using the LPIB read method, resulting in
distorted recordings.

However, using the position buffer is broken for playback streams,
resulting in a fallback to the LPIB method with the current driver.
This patch works around the issue by independently detecting the read
position method for capture and playback streams.

The patch will not have any effect if the position fix method is
explicitly set.

[Code simplified by tiwai]

Signed-off-by: Shahin Ghazinouri <shahin.ghazinouri@pelagicore.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-11 08:21:33 +02:00
Daniel T Chen
0ebf9e3692 ALSA: hda: Fix 0 dB for Lenovo models using Conexant CX20549 (Venice)
Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-May/027525.html

As reported on the mailing list, we also need to cap to the 0 dB offset
for Lenovo models, else the sound will be distorted.

Reported-and-Tested-by: Tim Starling <tstarling@wikimedia.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-11 08:18:31 +02:00
Mark Gross
ed77134bfc PM QOS update
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation.  I did this because request more
accurately represents what it actually does.

Also, I added a string based ABI for users wanting to use a string
interface.  So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface.  (someone asked me for it and I don't think
it hurts anything.)

This patch updates some documentation input I got from Randy.

Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
2010-05-10 23:08:19 +02:00
Linus Torvalds
94b849aaf6 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: Revert "ALSA: hda/realtek: quirk for D945GCLF2 mainboard"
  ALSA: hda - add support for Lenovo ThinkPad X100e in conexant codec
  ALSA: hda - fix DG45ID SPDIF output
2010-05-10 09:48:27 -07:00
Takashi Iwai
5433137336 Merge branch 'fix/hda' into topic/hda 2010-05-10 17:24:03 +02:00
Stefan Lippers-Hollmann
482c453315 ALSA: Revert "ALSA: hda/realtek: quirk for D945GCLF2 mainboard"
This reverts commit 7aee674665.

As it doesn't seem to be universally valid for all mainboard revisions of
the D945GCLF2 and breaks snd-hda-intel/ snd-hda-codec-realtek on the Intel
Corporation "D945GCLF2" (LF94510J.86A.0229.2009.0729.0209) mainboard.

00:1b.0 Audio device [0403]: Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01)

Signed-off-by: Stefan Lippers-Hollmann <s.l-h@gmx.de>
Cc: <stable@kernel.org> [2.6.33]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 17:16:10 +02:00
Pierre-Louis Bossart
1965c441ec ALSA: hda: enable SPDIF output for Conexant 5051/Lenovo docking stations
This patch enables the SPDIF output pin by default. It also enables
it for quirks related to Levono docking stations (x200 and 25041,
identified with the same 17aa:20f2 ID). Even though not all Lenovo
docking stations have SPDIF connectors, enabling the pin by default
shouldn't be a problem for anyone.
Other quirks remain unmodified.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 17:00:01 +02:00
Mark Brown
896060c76b ASoC: Use more idiomatic driver name for WM8731
Make dev_() prints much prettier.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-10 11:04:34 +01:00
Mark Brown
06ae99888e ASoC: Refactor WM8731 regulator management into bias management
This allows more flexible integration with subsystem features.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-10 11:04:22 +01:00
Mark Brown
3efab7dcc0 ASoC: Allow DAI links to be kept active over suspend
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked.  This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link.  It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:37:13 +01:00
Mark Brown
452a5fd679 ASoC: Allow active paths from the GSM modem while the GTA02 is suspended
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:37:04 +01:00
Mark Brown
1547aba993 ASoC: Support leaving paths enabled over system suspend
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.

Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.

When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:36:48 +01:00
Mark Brown
9949788b79 ASoC: Refactor DAPM suspend handling
Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.

Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:36:36 +01:00
Mark Brown
50ae8384cd ASoC: Remove unused DAPM suspend flag
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:35:55 +01:00
Mark Brown
29e189c29d ASoC: Remove unneeded suspend bias managment from CODEC drivers
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:35:25 +01:00
Andrej Gelenberg
0217f1499c ALSA: hda - add support for Lenovo ThinkPad X100e in conexant codec
Ideapad quirks working for my ThinkPad X100e (microphone is not tested).

Signed-off-by: Andrej Gelenberg <andrej.gelenberg@udo.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 10:28:12 +02:00
Dominik Brodowski
317b6d6300 pcmcia: dev_node removal (write-only drivers)
dev_node_t was only used to transport some minor/major numbers
from the PCMCIA device drivers to deprecated userspace helpers.
However, only a few drivers made use of it, and the userspace
helpers are deprecated anyways. Therefore, get rid of dev_node_t .

As a first step, remove any usage of dev_node_t from drivers which
only wrote to this typedef/struct, but did not make use of it.

CC: linux-bluetooth@vger.kernel.org
CC: Harald Welte <laforge@gnumonks.org>
CC: linux-mtd@lists.infradead.org
CC: linux-wireless@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
2010-05-10 10:23:14 +02:00
Dominik Brodowski
eb14120f74 pcmcia: re-work pcmcia_request_irq()
Instead of the old pcmcia_request_irq() interface, drivers may now
choose between:

- calling request_irq/free_irq directly. Use the IRQ from *p_dev->irq.

- use pcmcia_request_irq(p_dev, handler_t); the PCMCIA core will
  clean up automatically on calls to pcmcia_disable_device() or
  device ejection.

- drivers still not capable of IRQF_SHARED (or not telling us so) may
  use the deprecated pcmcia_request_exclusive_irq() for the time
  being; they might receive a shared IRQ nonetheless.

CC: linux-bluetooth@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-wireless@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: linux-usb@vger.kernel.org
CC: linux-ide@vger.kernel.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
2010-05-10 10:23:13 +02:00
Dominik Brodowski
a7debe789d pcmcia: pass FORCED_PULSE parameter in pcmcia_request_configuration()
As it's only used there it makes no sense relying on pcmcia_request_irq().

CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
2010-05-10 10:23:12 +02:00
Takashi Iwai
670ff6abd6 ALSA: opl4 - Fix a wrong argument in proc write callback
The commit 24e4a1211f
    ALSA: info - Use standard types for info callbacks
introduced a wrong type to snd_opl4_mem_proc_write() for pos argument.
Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 10:21:32 +02:00
Krzysztof Helt
a20971b201 ALSA: Merge es1688 and es968 drivers
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.

Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.

Also, a new PnP id is added for the card I acquired (the change
was tested on this card).

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 09:49:30 +02:00
Krzysztof Helt
396fa82726 ALSA: es1688: allocate snd_es1688 structure as a part of snd_card structure
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 09:48:59 +02:00
Takashi Iwai
02a2ad4029 Merge branch 'fix/misc' into topic/misc 2010-05-10 09:48:47 +02:00
Ville Syrjälä
1bde78bc25 ALSA: maestro3: Clear interrupts before enabling them
Avoid spurious interrupts when initializing the device.

Signed-off-by: Ville Syrjälä <syrjala@sci.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-08 11:51:13 +02:00
Ville Syrjälä
6895b5262e ALSA: es1968: Clear interrupts before enabling them
Avoid spurious interrupts when initializing the device.

Signed-off-by: Ville Syrjälä <syrjala@sci.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-08 11:51:06 +02:00
Daniel Mack
5e68888356 ALSA: sound/usb: fix UAC1 regression
Commit 23caaf19b ("ALSA: usb-mixer: Add support for Audio Class v2.0")
broke support for Class1 devices due to two faulty changes. This patch
fixes it.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-and-Tested-by: The Source <thesourcehim@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-08 11:39:44 +02:00
Jassi Brar
d0bbc24d2a ASoC: SMDK64XX: Switch to IISv4 CPU driver
Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 16:46:06 +01:00
Jassi Brar
af56b1c27b ASoC: S3C64XX: IISv4: Add CPU driver
Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 16:45:41 +01:00
Peter Ujfalusi
bd843edf81 ASoC: tpa6130a2: Fix for the custom kcontrol functions
Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 16:42:40 +01:00
Peter Ujfalusi
826e962c46 Revert "ASoC: tpa6130a2: Support for limiting gain"
This reverts commit 6f3991152f.

Since core has now support for limiting the volume on controls this
patch is not needed.  Furthermore, this patch actually prevents the core
to set new volume on the TPA.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 16:42:23 +01:00
Peter Ujfalusi
637d3847ba ASoC: core: Support for limiting the volume
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)

If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:

snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);

This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 16:41:33 +01:00
Mark Brown
3057876498 Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35 2010-05-07 16:38:26 +01:00
Wu Fengguang
4d26f44657 ALSA: hda - fix DG45ID SPDIF output
This reverts part of commit 52dc438606, in order to fix a regression:
broken SPDIF output on Intel DG45FC motherboard (IDT 92HD73E1X5 codec).

	--- DG45FC-IDT-codec-2.6.32  (SPDIF OK)
	+++ DG45FC-IDT-codec-2.6.33  (SPDIF broken)

	 Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital
	   Pincap 0x00000010: OUT
	-  Pin Default 0x40f000f0: [N/A] Other at Ext N/A
	-    Conn = Unknown, Color = Unknown
	-    DefAssociation = 0xf, Sequence = 0x0
	-  Pin-ctls: 0x00:
	+  Pin Default 0x014510a0: [Jack] SPDIF Out at Ext Rear
	+    Conn = Optical, Color = Black
	+    DefAssociation = 0xa, Sequence = 0x0
	+  Pin-ctls: 0x40: OUT
	   Connection: 3
	      0x25* 0x20 0x21
	 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital
	   Pincap 0x00000010: OUT
	-  Pin Default 0x01451140: [Jack] SPDIF Out at Ext Rear
	+  Pin Default 0x074510b0: [Jack] SPDIF Out at Ext Rear Panel
	     Conn = Optical, Color = Black
	-    DefAssociation = 0x4, Sequence = 0x0
	-    Misc = NO_PRESENCE
	-  Pin-ctls: 0x40: OUT
	+    DefAssociation = 0xb, Sequence = 0x0
	+  Pin-ctls: 0x00:
	   Connection: 3
	      0x26* 0x20 0x21

Cc: <stable@kernel.org>
Cc: Alexey Fisher <bug-track@fisher-privat.net>
Tested-by: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-07 10:24:53 +02:00
Benjamin Herrenschmidt
1ed31d6db9 Merge commit 'origin/master' into next 2010-05-07 11:29:25 +10:00
Takashi Iwai
aeb29a82de Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-05-06 17:06:27 +02:00
Peter Ujfalusi
2f005471e2 ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power
Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control

It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.

The codec reset values are considered safe in all environmnts.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-06 14:58:29 +01:00
Peter Ujfalusi
6f3991152f ASoC: tpa6130a2: Support for limiting gain
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-06 14:58:20 +01:00
Jarkko Nikula
5193d62f18 ASoC: tlv320aic3x: Add platform data and reset gpio handling
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-06 14:58:02 +01:00
Jarkko Nikula
49100c9835 ASoC: omap: Add basic audio support for Nokia RX-51/N900
This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-06 09:50:11 +01:00
Takashi Iwai
ef5dbbccbb ALSA: hda - Remove superfluous external amp setup for ALC888
We had a fixed external amp setup enabled for ALC888, but this seems
unnecessary.  The amps are controlled rather by GPIOs.
Let's remove it now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-06 08:40:25 +02:00
Takashi Iwai
20d157aef2 Merge branch 'fix/hda' into topic/hda 2010-05-06 08:39:43 +02:00
Linus Torvalds
38c9e91bc3 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
  ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
  ALSA: take tu->qlock with irqs disabled
  ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
  ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
  ALSA: hda - fix array indexing while creating inputs for Cirrus codecs
  ALSA: es968: fix wrong PnP dma index
2010-05-05 07:54:22 -07:00
Jassi Brar
8a7c251871 ASoC: S3C: I2S: Move set_sysclk to common code
Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:15:14 +01:00
Jassi Brar
9e991a4bf3 ASoC: S3C: I2Sv2: New field for controller feature
In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c

While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:14:21 +01:00
Jassi Brar
d47ef9c79d ASoC: S3C64XX: I2S: Use s3c2412 defines
Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:13:48 +01:00
Jassi Brar
5728242789 ASoC: S3C: I2Sv2: Unify i2s_get_clock callback
Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:13:20 +01:00
Jassi Brar
21a7ad08e2 ASoC: S3C: I2Sv2: Discard redundant field iis_clk
No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:12:29 +01:00
Jassi Brar
d79696ff44 ASoC: S3C2412: I2S: Return correct source clock
Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:11:52 +01:00
Jassi Brar
ce76f9fd34 ASoC: S3C2412: I2S: Debug IMS field
The IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:11:29 +01:00
Jassi Brar
b720d56294 ASoC: SAMSUNG: I2S: Add bit definitions
Define more bit definitions in the order of mainline
support for the SoC.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:11:02 +01:00
Jassi Brar
d07e7ce9b6 ASoC: S3C: I2Sv2: Move defines closer to driver
The header for I2Sv2
   linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-05 15:10:39 +01:00
Mark Brown
985d8c4c9e ASoC: Add debug output tracing all cache register writes
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-05 15:10:17 +01:00
Takashi Iwai
69b5de8475 Merge branch 'fix/hda' into for-linus 2010-05-05 10:08:30 +02:00
Daniel T Chen
8f0f5ff677 ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
BugLink: https://launchpad.net/bugs/541802

The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_cxt5045() for all Packard Bell models.

Reported-by: Valombre
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 10:01:15 +02:00
Anisse Astier
231f50bc0e ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper
HP and Mic support.

Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 10:00:00 +02:00
Dan Carpenter
bfe70783ca ALSA: take tu->qlock with irqs disabled
We should disable irqs when we take the tu->qlock because it is used in
the irq handler.  The only place that doesn't is
snd_timer_user_ccallback().  Most of the time snd_timer_user_ccallback()
is called with interrupts disabled but the the first ti->ccallback()
call in snd_timer_notify1() has interrupts enabled.

This was caught by lockdep which generates the following message:

> =================================
> [ INFO: inconsistent lock state ]
> 2.6.34-rc5 #5
> ---------------------------------
> inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage.
> dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes:
> (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer]
> {HARDIRQ-ON-W} state was registered at:
>   [<c1048de9>] __lock_acquire+0x654/0x1482
>   [<c1049c73>] lock_acquire+0x5c/0x73
>   [<c125ac3e>] _raw_spin_lock+0x25/0x34
>   [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer]
>   [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer]

Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 09:57:08 +02:00
Daniel T Chen
c536668138 ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
BugLink: https://launchpad.net/bugs/549267

The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.

Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 09:52:41 +02:00
Daniel T Chen
4442dd4613 ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
BugLink: https://launchpad.net/bugs/573284

The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.

Reported-by: Andy Couldrake <acouldrake@googlemail.com>
Tested-by: Andy Couldrake <acouldrake@googlemail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 09:51:15 +02:00
Brian J. Tarricone
8dd34ab111 ALSA: hda - fix array indexing while creating inputs for Cirrus codecs
This fixes a problem where cards show up as only having a single mixer
element, suppressing all sound output.

Signed-off-by: Brian J. Tarricone <brian@tarricone.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-05 09:45:33 +02:00
Peter Ujfalusi
e5e5b31e8c ASoC: tpa6130a2: TLV mapping for tpa6140a2
Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.

Provide different mixer control for the chips with correct
TLV mapping.

User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140

The way machine drivers are using this amplifier remained
the same.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-04 20:55:01 +01:00
Peter Ujfalusi
ad05c03b1c ASoC: tlv320dac33: Support for turning off the codec
Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).

After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).

The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.

There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, but we does not
   need to execute the playback related configuration
2. Playback caused  BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, and also we need
   to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
   Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   still on.

Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.

Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:55:54 +01:00
Peter Ujfalusi
0b61d2b9f2 ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure
As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.

The substream must be easily available in other places than
pcm_* callbacks.

Manage a pointer in _startup, and _shutdown for this.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:55:48 +01:00
Peter Ujfalusi
239fe55c7f ASoC: tlv320dac33: Revised module loading, and DAC33 ID read
Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:55:48 +01:00
Peter Ujfalusi
ef909d6729 ASoC: tlv320dac33: Optimize power up, and restore
On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:55:48 +01:00
Peter Ujfalusi
1b7c9afbfb ASoC: TWL4030: Remove OUTL/R outputs
OUTL/R are leftovers from the original driver, and they
are no longer needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:47:30 +01:00
Peter Ujfalusi
7b4c734eea ASoC: TWL4030: AIF/APLL fix in DAPM domain
This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.

We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */

/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */

/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */

The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).

The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-03 12:47:29 +01:00
Ingo Molnar
53ba4f2fa7 Merge commit 'v2.6.34-rc6' into core/locking 2010-05-03 09:17:01 +02:00
Geert Uytterhoeven
b0b4ce38a5 MIPS: TXx9: Add missing MODULE_ALIAS definitions for TXx9 platform devices
This enables autoloading of the TXx9 sound driver on RBTX4927.

Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
To: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Cc: Linux MIPS Mailing List <linux-mips@linux-mips.org>
Patchwork: http://patchwork.linux-mips.org/patch/1101/
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
2010-04-30 20:52:40 +01:00
Mark Brown
39b8eab7e7 ASoC: Add WM9090 amplifier driver
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.

Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control.  The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-30 16:12:44 +01:00
Liam Girdwood
cf134d5bfb ASoC: tlv320dac33 - disable regulators at i2c remove()
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-28 13:27:18 +01:00
Liam Girdwood
1849235876 ASoC: zoom2 - update DAPM pins
Remove bogus twl4030 pins

Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-28 13:27:18 +01:00
Liam Girdwood
1beb91f004 ASoC: pandora - update DAPM pins
Remove bogus TWL4030 pins.

Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-28 13:27:18 +01:00
Mark Brown
dde3a7e9cb ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option
The SYSCLK source is automatically managed when configuring the PLL.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-28 11:33:04 +01:00
Takashi Iwai
cb7b76961f Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-04-27 15:35:59 +02:00
Jarkko Nikula
07779fdd1a ASoC: tlv320aic3x: Add basic regulator support
This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-27 11:19:23 +01:00
Jarkko Nikula
db13802e51 ASoC: tlv320aic3x: Change bias management semantics
Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-27 11:08:06 +01:00
Jarkko Nikula
d3235c4ac1 ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_level
These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-27 11:08:06 +01:00
Jarkko Nikula
c6de6e0300 ASoC: tlv320aic3x: Remove unused version string
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-27 11:08:05 +01:00
Vladimir Zapolskiy
b28528a124 ASoC: UDA134X: Add UDA1345 CODEC support
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-26 15:28:18 +01:00
Mark Brown
5e5e2bef28 ASoC: Warn on low WM8994 AIFCLK
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:26:13 +01:00
Mark Brown
759512fbac ASoC: Correct inversion of speaker mixer PCM switch
Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:24:28 +01:00
Peter Ujfalusi
f57d2cfaad ASoC: tlv320dac33: FIFO caused delay reporting
Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.

Bypass mode: FIFO is bypassed, report 0 as delay

Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.

Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.

Phase1: when we received the alarm threshold, but our workqueue has
        not been executed (safeguard phase). Just count the played out
        samples since ts1 and subtract it from the alarm threshold
        value.
Phase2: During nSample burst (after writing to nSample register), count
        the played out samples since ts1, count the samples received
        since ts2 (in a burst). Estimate the FIFO depth using these and
        alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
        samples since ts1. Estimate the FIFO depth using the nSample
        configuration and the alarm threshold value.

Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received

Interrupts are coming when DAC33 FIFO depth reaches upper threshold.

Phase1: Draining phase (after the burst), counting the played out
        samples since ts1, and subtract it from the upper threshold
        value.
Phase2: During burst operation. Using the pre calculated time needed to
        play out samples from the buffer during the drain period (from
        upper to lower threshold), move the time window to cover the
        estimated time from the burst start to the current time.
        Calculate the samples played out since lower threshold and also
        the samples received during the same time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:16:39 +01:00
Peter Ujfalusi
76f471274d ASoC: tlv320dac33: Calculate the interface speed during bursts
When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:16:33 +01:00
Peter Ujfalusi
4260393e71 ASoC: tlv320dac33: Change magic numbers used in Mode7
Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:16:28 +01:00
Peter Ujfalusi
55abb59c9a ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode
There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:16:23 +01:00
Peter Ujfalusi
f4d5932806 ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1
Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-26 15:16:18 +01:00
Krzysztof Helt
867f1845c5 ALSA: es968: fix wrong PnP dma index
There is only one dma for the ESS ES968 based board.
Its index is 0 and not 1.

This make the es968 card working.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-26 09:05:44 +02:00
Mark Brown
3a278a0c65 ASoC: Allow reporting of NULL jacks
Follow the core jack implementation and allow reporting on the status
of NULL jacks, avoiding the need to check in detection implementations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-23 17:07:10 +01:00
Barry Song
ba0a24e738 ASoC: ad193x: fix typo, delete redundant space
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-23 16:14:57 +01:00
Barry Song
d6bdc0f7fe ASoC: ad193x: fix wrong register setting in ad193x_set_dai_fmt
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-23 16:14:02 +01:00
Takashi Iwai
227c4edb72 Merge branch 'fix/misc' into for-linus 2010-04-23 17:10:48 +02:00
Takashi Iwai
1f10cd34d9 Merge branch 'fix/hda' into for-linus 2010-04-23 17:10:44 +02:00
Hans de Goede
5a5e02e509 ALSA: snd-es1968: Make hardware volume buttons an input device (rev2)
The hardware volume handling code in essence just detects key presses, and
then does some hardcoded modification of the master volume based on which key
is pressed.

Clearly the right thing to do here is just report these keypresses to
userspace and let userspace decide what to with them.

This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-23 17:09:59 +02:00
Hans de Goede
eb581adf25 ALSA: snd-maestro3: Make hardware volume buttons an input device (rev2)
While working on the sound suspend / resume problems with my laptop
I noticed that the hardware volume handling code in essence just detects
key presses, and then does some hardcoded modification of the master volume
based on which key is pressed.

This made me think that clearly the right thing to do here is just report
these keypresses to userspace and let userspace decide what to with them.

This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.

As an added bonus the keys now work identical to volume keys on a (usb)
keyboard with multimedia keys, providing visual feedback of the volume
level change, and a better range of the volume control (with a properly
configured desktop environment).

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-23 17:09:46 +02:00
Daniel T Chen
5c1bccf645 ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558
BugLink: https://launchpad.net/bugs/568600

The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.

This change is necessary for 2.6.32.11 and 2.6.33.2 alike.

Reported-by: Andy Ross <andy@plausible.org>
Tested-by: Andy Ross <andy@plausible.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-23 08:01:42 +02:00
Daniel T Chen
0e0280dc2b ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203
BugLink: https://launchpad.net/bugs/459083

The OR has verified with 2.6.32.11 and the latest alsa-driver stable
daily snapshot that position_fix=1 is necessary for the external mic
to work and for PulseAudio not to crash constantly.

This patch is necessary also for 2.6.32.11 and 2.6.33.2.

Reported-by: <imwithid@yahoo.com>
Tested-by: <imwithid@yahoo.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-23 08:00:43 +02:00
Jiri Kosina
6c9468e9eb Merge branch 'master' into for-next 2010-04-23 02:08:44 +02:00
Hans de Goede
20133d4cd3 ALSA: snd-meastro3: Document hardware volume control a bit
While working on a fix for the volume being muted on the allegro in my
Compaq EVO N600C after suspend, I've learned a few things about the hardware
volume control worth documenting. The actual fix for the suspend / resume
issue is in the next patch in this set.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 16:53:38 +02:00
Takashi Iwai
6458a54423 Merge branch 'fix/misc' into topic/misc 2010-04-22 16:53:24 +02:00
Hans de Goede
715aa67533 ALSA: snd-meastro3: Ignore spurious HV interrupts during suspend / resume
Ignore spurious HV interrupts during suspend / resume, this avoids
mistaking them for a mute button press. This is not very pretty but
it seems the only way to fix the master volume control gets muted
after suspend issue I'm seeing. Note that the es1968 driver is doing
exactly the same.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 16:53:10 +02:00
Hans de Goede
7efbfd1ae9 ALSA: snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
Without this quirk sound stops working after suspend resume. With this quirk,
one still needs to manually unmute the master volume control after a suspend /
/ resume cycle. That is fixed in another patch in this set.

Note that this patch was submitted to the alsa bug tracker a long time ago:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 16:52:39 +02:00
Daniel T Chen
3353541fe5 ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526
BugLink: https://launchpad.net/bugs/567494

The OR has verified that the existing model quirk, ALC880_UNIWILL,
is insufficient for audible playback and capture by default. Instead,
the ALC880_F1734 model quirk needs to be used.

This change is necessary for both 2.6.32.11 and 2.6.33.2.

Reported-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Tested-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 14:58:15 +02:00
Daniel T Chen
aac78daf8f ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio XPS 1645
BugLink: https://launchpad.net/bugs/553002

The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.

This change is necessary for 2.6.32.11 and 2.6.33.2 alike.

Reported-by: Robert Chambers
Tested-by: Robert Chambers
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 09:14:32 +02:00
Eliot Blennerhassett
719f82d398 ALSA: Add support of AudioScience ASI boards
Added the support of AudioScience ASI boards.
The driver has been tested for years on alsa-driver external tree,
now finally got merged to the kernel.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-22 07:21:53 +02:00
Mark Brown
7add84aa77 ASoC: Allow unspecified source when stopping WM8994 FLLs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-22 02:29:01 +09:00
Mark Brown
ee839a2127 ASoC: Tone down debugging for WM8994 class W
It's a little verbose during path changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-21 01:41:28 +09:00
Mark Brown
7d48a6acbc ASoC: Set full range of WM8994 FLL Fratio values
Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-21 01:41:27 +09:00
Mark Brown
136ff2a272 ASoC: Support FLL input clock selection on WM8994
The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-21 01:41:26 +09:00
Phil Carmody
4f6f22d7be ASoC: da7210: Fencepost error in reg cache read
An index equal to the array size may not be accessed.

Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-21 01:41:26 +09:00
Takashi Iwai
d4a8ca2461 ASoC: missing conversions to snd_soc_codec_*_drvdata()
Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-20 08:29:19 +02:00
Takashi Iwai
b7d2526f5c ALSA: hda - Fix resume from StR of HP 2510p with docking-station
When HP laptop with AD1981 codec is suspended and the docking-station
is connected before the resume, the outputs get confused, and wrongly
routed still to the speaker.  This is because of a change in 2.6.34-rc1
ea52bf260e
    ALSA: hda: Add powerdown for Analog Devices HDA codecs

The problem was the added resume callback that doesn't consider the
modified init hook.  The fix is simply remove the resume callback here
and make the resume normally.  This doesn't change any behavior intended
in the commit above (for shutting down the sound at suspend) but only
fixes the resume.

Reported-and-tested-by: Frans Pop <elendil@planet.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-19 18:11:29 +02:00
Mark Brown
b2c812e22d ASoC: Add indirection for CODEC private data
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.

To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-17 10:46:22 +09:00
Mark Brown
890c681275 Merge branch 'for-2.6.34' into for-2.6.35 2010-04-17 10:45:54 +09:00
Takashi Iwai
cf0dbba515 Merge remote branch 'alsa/devel' into topic/misc 2010-04-16 15:20:06 +02:00
Jaroslav Kysela
ca4c2adaf2 ALSA: usb/mixer - use get_iface_desc() rather than direct structure
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-04-16 10:37:50 +02:00
Jaroslav Kysela
f09d045e2a Merge branch 'topic/usb' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel 2010-04-16 10:37:41 +02:00
Takashi Iwai
923125c650 Merge branch 'fix/hda' into for-linus 2010-04-16 10:03:48 +02:00
Takashi Iwai
872d65f674 Merge branch 'fix/misc' into for-linus 2010-04-16 10:03:42 +02:00
Takashi Iwai
d336905e00 Merge branch 'fix/asoc' into for-linus 2010-04-16 10:03:36 +02:00
Sascha Hauer
8392609969 ASoC: imx-ssi: do not call hrtimer_disable in trigger function
Doing so causes a deadlock, so just signal the timer to stop
using an atomic variable.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-16 01:02:35 +09:00
Brian Waters
1cff399ecd ALSA: i2c: Fixed 8 checkpatch errors
Fixed 8 checkpatch errors (ERROR: do not use assignment in if condition)
in sound/i2c/i2c.c.

Signed-off-by: Brian Waters <brianmwaters@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-15 10:13:54 +02:00
Jens Taprogge
7b2bfdbc0d ALSA: hda - Add initial support for Thinkpad T410s HDA codec
attached please find a patch that adds support for at least the T410s
HDA codec.  Most likely it will also add support for the T410 and T510
based models.

The patch was derived from Ideapad support.  Support for the laptop's and
docking-station output connectors as well as the docking-station microphone
connector and the laptops internal devices has been tested.  Since it has been
developed without a data-sheet available, support for digital outputs and the
laptop's microphone input may well be incorrect.

Microphone mute functionality is not included:
The microphone mute button seems to be reported through thinkpad_acpi key
0000101b.  The mute button LED seems to be wired to thinkpad_acpi led
number 15.

Signed-off-by: Jens Taprogge <jens.taprogge@taprogge.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-15 09:10:29 +02:00
Takashi Iwai
039f0f3a5b Merge branch 'fix/hda' into topic/hda 2010-04-15 09:09:02 +02:00
Takashi Iwai
8815cd030f ALSA: hda - Add position_fix quirk for Biostar mobo
The Biostar mobo seems to give a wrong DMA position, resulting in
stuttering or skipping sounds on 2.6.34.  Since the commit
7b3a177b0d, "ALSA: pcm_lib: fix "something
must be really wrong" condition", makes the position check more strictly,
the DMA position problem is revealed more clearly now.

The fix is to use only LPIB for obtaining the position, i.e. passing
position_fix=1.  This patch adds a static quirk to achieve it as default.

Reported-by: Frank Griffin <ftg@roadrunner.com>
Cc: Eric Piel <Eric.Piel@tremplin-utc.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-15 09:02:41 +02:00
Joerg Schirottke
d1501ea844 ALSA: hda - add a quirk for Clevo M570U laptop
Added the matching model for Clevo laptop M570U.

Signed-off-by: Joerg Schirottke <master@kanotix.com>
Tested-by: Maximilian Gerhard <maxbox@directbox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-15 08:37:41 +02:00
Sascha Hauer
565a79f74a ASoC: imx-ssi: increase minimum periods to 4
Currently the notification of elapsed periods is not very exact.
Increase minimum periods to 4 as suggested by Liam Girdwood.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-15 10:29:49 +09:00
Takashi Iwai
b265faed8c Merge branch 'fix/hda' into topic/hda 2010-04-14 14:39:21 +02:00
Takashi Iwai
3d83e577a8 ALSA: hda - Avoid invalid "Independent HP" control for VIA codecs
Some VIA codecs have no multiple source selection for headphone pins,
thus it's useless (and wrong) to create "Independent HP" control on them.

This patch adds the check of connections to skip the control in such a
case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-14 14:36:23 +02:00
Takashi Iwai
b331439dfd ALSA: hda - Fix control element allocations in VIA codec parser
The commit 5b0cb1d850
    ALSA: hda - add more NID->Control mapping
breaks the control element allocation by returning a wrong value.
Let's fix it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-14 14:35:11 +02:00
Takashi Iwai
02f4865fa4 ALSA: core - Define llseek fops
Set no_llseek to llseek file ops of each sound component (but for hwdep).
This avoids the implicit BKL invocation via generic_file_llseek() used
as default when fops.llseek is NULL.

Also call nonseekable_open() at each open ops to ensure the file flags
have no seek bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-13 12:01:21 +02:00
Takashi Iwai
73029e0ff1 ALSA: info - Implement common llseek for binary mode
The llseek implementation is identical for existing driver implementations,
so let's merge to the common layer.  The same code for the text proc file
can be used even for the binary proc file.

The driver can provide its own llseek method if needed.  Then the common
code will be skipped.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-13 12:01:20 +02:00
Takashi Iwai
d97e1b7823 ALSA: info - Check file position validity in common layer
Check the validity of the file position in the common info layer before
calling read or write callbacks in assumption that entry->size is set up
properly to indicate the max file size.

Removed the redundant checks from the callbacks as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-13 12:01:14 +02:00
Takashi Iwai
24e4a1211f ALSA: info - Use standard types for info callbacks
Use loff_t, size_t and ssize_t for arguments of info callbacks
to follow the standard procfs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-13 11:57:14 +02:00
Takashi Iwai
067e4a5d23 Merge branch 'topic/bkl' into topic/core-cleanup 2010-04-13 11:24:34 +02:00
Takashi Iwai
96d9e9c039 Merge branch 'fix/misc' into topic/misc 2010-04-13 11:14:43 +02:00
Philby John
b68b58fd6a ALSA: aaci - Fix alignment faults on ARM Cortex introduced by commit 29a4f2d3
The commit 29a4f2d3 used writel() at offset 0x26 which is
half-word aligned causing unaligned exceptions on a
Cortex-A8. The original patch solved the "aaci-pl041 fpga:04:
ac97 read back fail" issue on a soft reset. Reading from any
arbitrary aaci register seems to solve this issue.

Signed-off-by: Philby John <pjohn@mvista.com>
Acked-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-13 09:46:55 +02:00
Marek Vasut
d21e0f4cd1 ASoC: Zipit Z2 WM8750 ASoC driver
This patch adds support for sound through the WM8750 codec on Zipit Z2.
Also, this patch incorporates support for detecting headset jack
insertion through the jack detection API.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-12 11:33:16 +01:00
Bill Gatliff
e135443e21 ASoC: Use SNDRV_PCM_RATE_8000_96000 macro for WM8731
Signed-off-by: Bill Gatliff <bgat@billgatliff.com>
Acked-by: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-12 11:33:04 +01:00
Takashi Iwai
ff818c24c2 ALSA: hda - Add fix-up for Sony VAIO with ALC269
Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF
ground or Hi-Z to make the headphone working.  Other than that, model=auto
works fine, so let's use model=auto with a specific fix-up table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-12 08:59:25 +02:00
Takashi Iwai
7fa90e873f ALSA: hda - Enhance fix-up table for Realtek codecs
A few enhancement / fixes for fix-up table of some Realtek codecs:
 - Apply fix-ups only for the auto model
 - Apply additional verbs after normal init verbs
 - Add a debug print to show the fix-up application

This is basically a preliminary work for the next fix for Sony VAIO.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-12 08:58:48 +02:00
Takashi Iwai
27762b2ce1 Merge branch 'fix/misc' into topic/usb 2010-04-10 21:34:56 +02:00
Takashi Iwai
29aac005ff ALSA: usb - Fix Oops after usb-midi disconnection
usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after
disconnection.  This is due to the access to the endpoints which have
been already released at disconnection while the files are still alive.

This patch fixes the problem by checking disconnection state at
snd_usbmidi_output_drain() and by releasing urbs but keeping the
endpoint instances until really all freed.

Tested-by: Tvrtko Ursulin <tvrtko@ursulin.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-10 21:34:24 +02:00
Takashi Iwai
60508abe9b Merge branch 'fix/hda' into topic/hda 2010-04-09 17:36:19 +02:00
Takashi Iwai
7f311a4691 ALSA: hda - Fix initial capture source connections of ALC880/260
The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin.  This can be a problem when mode=auto
and there is only one input pin.  Then user can't change the connection
at all.

This patch adds the code to initialize the input pin connection of these
codecs.

Reference: Novell bnc#594363
	https://bugzilla.novell.com/show_bug.cgi?id=594363

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-09 17:35:42 +02:00
Marek Vasut
6ca0c22ef8 ASoC: WM8750: Convert to new API
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.

Also, this patch fixes the Jive and Spitz machine.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-09 12:17:42 +01:00
Kailang Yang
226b1ec8c1 ALSA: hda - Fix setup for ALC269vb amic and dmic models
Corrected HP and mic pins for ALC269vb amic and dmic models.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-09 11:01:20 +02:00
Kailang Yang
531d8791ac ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21
ALC269vb has an alternative HP pin 0x21 in addition.
Fix the parser to recognize it.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-09 10:57:33 +02:00
Takashi Iwai
4cf19b848f ALSA: Remove BKL from open multiplexer
Use a local mutex instead of BKL.  This should suffice since each device
type has also its open_mutex.
Also, a bit of clean-up of the legacy device auto-loading code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-09 10:28:36 +02:00
Sascha Hauer
43a3cec013 ASoC: imx-ssi: Use a hrtimer in FIQ mode
Using a regular timer results in poll times < 1 jiffie with small
buffers, so we loaded the timer with the actual jiffie value. We can
be more accurate using a hrtimer. Also, we have to call
snd_pcm_period_elapsed after playing period_bytes and not
runtime->period_size (which is in samples and not in bytes).

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-08 15:21:05 +01:00
Sascha Hauer
671999cb5d ASoC: imx-pcm-dma-mx2: restart DMA after an error
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-08 15:21:01 +01:00
Sascha Hauer
206b60e189 ASoC: imx-ssi: honor IMX_SSI_DMA flag
When checking if we are DMA capable we have to check for the
IMX_SSI_DMA flag which is already set from platform_data instead
of setting it again when we want to do DMA.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@Slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-08 15:20:57 +01:00
Huang Weiyi
78e4fd26ef ASoC: wm2000: remove unused #include <linux/version.h>
Remove unused #include <linux/version.h>('s) in
  sound/soc/codecs/wm2000.c

Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-08 15:16:00 +01:00
Takashi Iwai
5b5cd553e3 ALSA: info - Remove BKL
Use the fine-grained mutex for the assigned info object, instead.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 18:33:57 +02:00
Takashi Iwai
d05468b72a ALSA: pcm - Remove BKL from async callback
It's simply calling fasync_helper().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 18:29:46 +02:00
Linus Torvalds
84db18bbeb Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: mixart: range checking proc file
  ALSA: hda - Fix a wrong array range check in patch_realtek.c
  ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
  ALSA: hda - Enable amplifiers on Acer Inspire 6530G
  ASoC: Only do WM8994 bias off transition from standby
  ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
  ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
  ASoC: Support second DC servo readback method for wm_hubs
  ASoC: Avoid wraparound in wm_hubs DC servo correction
  ALSA: echoaudio - Eliminate use after free
  ALSA: i2c: cleanup: change parameter to pointer
  ALSA: hda - Add MSI blacklist for Aopen MZ915-M
  ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
  ALSA: hda - Update document about MSI and interrupts
  ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
  ALSA: hda - Add missing printk argument in previous patch
  ASoC: Fix passing platform_data to ac97 bus users and fix a leak
  ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
  ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
  ASoC: wm8994: playback => capture
2010-04-07 08:42:25 -07:00
Maurus Cuelenaere
7ad7b218f4 ALSA: hda: Add support for Medion WIM2160
This adds support for the Medion WIM2160 soundcard.
There's no PCI quirk added because it has the same PCI id as the
Medion MD2.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 14:56:55 +02:00
Takashi Iwai
25e8d9b67b ALSA: hda - Remove left-over debug printk in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 14:53:00 +02:00
d binderman
6237cdac5d powerpc/aoa: gpio-pmf.c: 3 * redundant code
Signed-off-by: David Binderman <dcb314@hotmail.com>
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
2010-04-07 18:00:36 +10:00
Takashi Iwai
55b371d4ac Merge branch 'fix/hda' into for-linus 2010-04-07 09:54:46 +02:00
Takashi Iwai
7445c995b0 Merge branch 'fix/asoc' into for-linus 2010-04-07 09:54:41 +02:00
Takashi Iwai
1172234cbe Merge branch 'fix/misc' into for-linus 2010-04-07 09:54:33 +02:00