Commit Graph

7575 Commits

Author SHA1 Message Date
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
Takashi Iwai 489008cd58 ALSA: hda - Fix ALC882 DAC connections in auto mode
Assign DACs properly to each output.  Currently, the front output is bound
to HP/speaker outputs blindly, but they should be assigned to individual
DACs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 09:06:00 +02:00
Takashi Iwai 92ab7b8f38 Merge branch 'fix/hda' into topic/hda 2010-04-07 08:38:47 +02:00
Takashi Iwai 68c7ccb8f8 ALSA: powermac - Fix obsoleted machine_is_compatible()
machine_is_compatible() was renamed to of_machine_is_compatible().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-07 07:45:46 +02:00
Dan Carpenter b0cc58a25d ALSA: mixart: range checking proc file
The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".

Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".

Signed-off-by: Dan Carpenter <error27@gmail.com>
Cc: <stable@kernel.org>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-06 18:52:39 +02:00
Takashi Iwai f9700d5a45 ALSA: hda - Fix a wrong array range check in patch_realtek.c
The commit 6a4f2ccb46 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections.  Fixed now.

Reference: bko#15689
	https://bugzilla.kernel.org/show_bug.cgi?id=15689

Reported-by: Adrian Ulrich <kernel@blinkenlights.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-05 23:36:16 +02:00
Mark Brown 53a61d967a Merge branch 'for-2.6.34' into for-2.6.35
Conflicts due to context changes next to the backported DMA data change:
	include/sound/soc.h
2010-04-05 19:19:32 +01:00
Mark Brown 8876698406 ASoC: Implement interrupt based WM8994 microphone detection
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.

Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-05 19:18:12 +01:00
Daniel Mack 5f712b2b73 ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.

All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.

[Note that this is a backported version for 2.6.34.
 Upstream commit is fd23b7dee]

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-05 19:14:11 +01:00
Tony Vroon d12841827a ALSA: hda - Enable amplifiers on Acer Inspire 6530G
After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-05 18:29:48 +02:00
Mark Brown d522ffbfb9 ASoC: Only do WM8994 bias off transition from standby
Otherwise we may try to power down multiple times when the using
idle bias off and the driver is removed.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-05 16:20:49 +01:00
Mark Brown 4dcc93d0ed ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo
operations has been deprecated and with some more recente revisions
may perform incorrectly, especially when only analogue bypass paths
are in use. Switch to using readback from the DC servo command
register instead, which is supported for all devices. Without this
unacceptably long timeouts may be observed in some circumstances.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-05 16:20:02 +01:00
Mark Brown ae9d8607fe ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
If we need to offset correct the DC servo then don't use runtime
recalibration since that is likely to introduce further offsets
which will be evident on powerdown.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-05 16:19:29 +01:00
Mark Brown 8437f7006b ASoC: Support second DC servo readback method for wm_hubs
More recent Wolfson hubs devices add the ability to read back the DC
servo calibration information from the register used to write offsets,
and later still ones remove the old readback registers. Add support
for the new scheme, and use it for WM8994 device revisions that
support it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-05 16:19:09 +01:00
Mark Brown 3fa49e3ad9 ASoC: Avoid wraparound in wm_hubs DC servo correction
If the correction wraps around then a substantial offset would be
introduced.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-05 16:17:39 +01:00
Risto Suominen f1b1f75e25 ALSA: powermac - Add debug log
Add some debug log in tumbler.c.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:26:34 +02:00
Risto Suominen b6d7335001 ALSA: powermac - Lineout detection on G4 DA
Lineout (Pro Speaker) detection on PowerMac G4 Digital Audio (Tumbler).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:25:55 +02:00
Risto Suominen 819ef70b13 ALSA: powermac - Reverse HP detection on G4 DA
Reverse headphone detection bit on PowerMac G4 Digital Audio (Tumbler).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:25:02 +02:00
Julia Lawall a0fd4345f9 ALSA: echoaudio - Eliminate use after free
Use the call to snd_card_free in the error handling code at the end of the
function, as in the other error cases.

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

// <smpl>
@@
expression E,E2;
@@

snd_card_free(E)
...
(
  E = E2
|
* E
)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:21:42 +02:00
Dan Carpenter f11947c7c5 ALSA: i2c: cleanup: change parameter to pointer
We actually pass an array of 7 chars not 5.
This silences a smatch warning.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:21:39 +02:00
Takashi Iwai 3815595e78 ALSA: hda - Add MSI blacklist for Aopen MZ915-M
The device needs MSI disablement.  Added to the quirk list.

Reported-by: Harald Dunkel <harri@afaics.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-04 12:14:03 +02:00
Janusz Krzysztofik b5442a75de ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
With recent (2.6.34) chnages in PCM handling, capture stopped working on my
OMAP1510 based Amstrad Delta videophone.

Using 2.6.34-rc2, I was able to correct the problem in 3 different ways:

1. reverting commit 7b3a177b0d,
2. enabling additional jiffies check with
	echo 4 >/proc/asound/card0/pcm0c0/xrun_debug
3. applying the patch below.

Since I wasn't able to reproduce the problem on my i686 PC, I guess the
problem is probably machine specific.

The patch reuses the method for software emulation of missing hardware
pointer, already implemented for playback on OMAP1510. It's possible that
event if a hardware pointer is available for capture on this machine, its
behaviour may be not compatible with what upper layer expects.

If you think the problem may be more general and should be solved differently,
on a higher level, I can try to work more on it if you give me a hint.

If the patch gets accepted, I suggest it goes as a fix in the current release
cycle.

Created and tested against linux-2.6.34-rc2.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-04-02 17:10:25 +01:00
Takashi Iwai c125ba3bec Merge branch 'topic/hda-alc-mute' into topic/hda 2010-04-01 16:04:28 +02:00
Takashi Iwai 7a2e38a555 Merge branch 'fix/hda' into topic/hda 2010-04-01 16:04:13 +02:00
Daniel T Chen b8e80cf386 ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
BugLink: https://launchpad.net/bugs/551606

The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_ad1981() for all models using the Thinkpad
quirk.

Reported-by: Jane Silber
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-31 11:47:14 +02:00
Takashi Iwai a68d5a5419 ALSA: hda - introduce snd_hda_codec_update_cache()
Add a new helper, snd_hda_codec_update_cache(), for reducing the unneeded
verbs.  This function checks the cached value and skips if it's identical
with the given one.  Otherwise it works like snd_hda_codec_write_cache().

The alc269 code uses this function as an example.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-30 18:03:44 +02:00
Takashi Iwai ad35879aa1 ALSA: hda - Add mute LED support for HP laptop with ALC269
Some HP laptops have a mute LED that is controlled over the unused
MIC2 VREF pin.  Implement the LED updater like patch_sigmatel.c for this
model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-30 18:03:11 +02:00
Takashi Iwai c35c9d5d3f Merge branch 'fix/hda' into topic/hda 2010-03-30 18:00:42 +02:00
Tejun Heo 5a0e3ad6af include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files.  percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.

percpu.h -> slab.h dependency is about to be removed.  Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability.  As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.

  http://userweb.kernel.org/~tj/misc/slabh-sweep.py

The script does the followings.

* Scan files for gfp and slab usages and update includes such that
  only the necessary includes are there.  ie. if only gfp is used,
  gfp.h, if slab is used, slab.h.

* When the script inserts a new include, it looks at the include
  blocks and try to put the new include such that its order conforms
  to its surrounding.  It's put in the include block which contains
  core kernel includes, in the same order that the rest are ordered -
  alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
  doesn't seem to be any matching order.

* If the script can't find a place to put a new include (mostly
  because the file doesn't have fitting include block), it prints out
  an error message indicating which .h file needs to be added to the
  file.

The conversion was done in the following steps.

1. The initial automatic conversion of all .c files updated slightly
   over 4000 files, deleting around 700 includes and adding ~480 gfp.h
   and ~3000 slab.h inclusions.  The script emitted errors for ~400
   files.

2. Each error was manually checked.  Some didn't need the inclusion,
   some needed manual addition while adding it to implementation .h or
   embedding .c file was more appropriate for others.  This step added
   inclusions to around 150 files.

3. The script was run again and the output was compared to the edits
   from #2 to make sure no file was left behind.

4. Several build tests were done and a couple of problems were fixed.
   e.g. lib/decompress_*.c used malloc/free() wrappers around slab
   APIs requiring slab.h to be added manually.

5. The script was run on all .h files but without automatically
   editing them as sprinkling gfp.h and slab.h inclusions around .h
   files could easily lead to inclusion dependency hell.  Most gfp.h
   inclusion directives were ignored as stuff from gfp.h was usually
   wildly available and often used in preprocessor macros.  Each
   slab.h inclusion directive was examined and added manually as
   necessary.

6. percpu.h was updated not to include slab.h.

7. Build test were done on the following configurations and failures
   were fixed.  CONFIG_GCOV_KERNEL was turned off for all tests (as my
   distributed build env didn't work with gcov compiles) and a few
   more options had to be turned off depending on archs to make things
   build (like ipr on powerpc/64 which failed due to missing writeq).

   * x86 and x86_64 UP and SMP allmodconfig and a custom test config.
   * powerpc and powerpc64 SMP allmodconfig
   * sparc and sparc64 SMP allmodconfig
   * ia64 SMP allmodconfig
   * s390 SMP allmodconfig
   * alpha SMP allmodconfig
   * um on x86_64 SMP allmodconfig

8. percpu.h modifications were reverted so that it could be applied as
   a separate patch and serve as bisection point.

Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.

Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-30 22:02:32 +09:00
Takashi Iwai 1f85d72d2c ALSA: hda - Add missing printk argument in previous patch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-30 07:48:05 +02:00
Mark Brown 2c9504228f Merge branch 'for-2.6.34' into for-2.6.35 2010-03-29 21:03:20 +01:00
Barry Song 9dd7b79a86 ASoC: ad193x: move codec register/unregister to bus probe/remove
The way i've factored out the bus probe and removal functions so
that there's no code in the individual I2C and SPI functions means
that the register() and unregister() functions could just be squashed
into the bus_probe() and bus_remove() functions.

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>
2010-03-29 21:02:24 +01:00
Graham Gower fb48e3c6a4 ASoC: Fix passing platform_data to ac97 bus users and fix a leak
[The issue is an attempt to write the pdata without the AC97 device
allocated when using ac97.c - also added a comment in soc-core.c for the
special case for ac97. -- broonie]

Signed-off-by: Graham Gower <graham.gower@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-29 21:00:37 +01:00
Mark Brown e6ab07ce0f Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35 2010-03-29 21:00:04 +01:00
Tejun Heo 7b7b904226 ALSA: usb - update gfp/slab.h includes
Implicit slab.h inclusion via percpu.h is about to go away.  Make sure
gfp.h or slab.h is included as necessary.

Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 21:29:03 +02:00
Tejun Heo 923a00427a ASoC: update gfp/slab.h includes
Implicit slab.h inclusion via percpu.h is about to go away.  Make sure
gfp.h or slab.h is included as necessary.

Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 21:28:43 +02:00
Takashi Iwai 6694635d3a ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALC269 codec has a few different variants, and each of them may have
different ADC and MUX widgets.  For example, one model has ADC 0x08
with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or
0x24.  The difference of ADC appears usually as the capability of
the digital mic pin (0x12), and the current driver sometimes misses
the internal mic pin due to the mismatching ADC.

This patch adds a bit more clever way to find the matching ADC instead
of the static list.  Now the driver checks all active input pins and
fills only the ADC/MUX's that contain all of them.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 17:27:31 +02:00
Stephen Rothwell 9966ddafe1 ALSA: usb pcm: use of kmalloc requires the include of slab.h
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 10:04:07 +02:00
Takashi Iwai d01e14a6b9 ASoC: Fix file permission of soc/codecs/twl6040.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 09:31:57 +02:00
Stephen Rothwell 68b40cc40a ASoC: TWL6040: use of kzalloc/kfree requires the include of slab.h
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 09:31:07 +02:00
Takashi Iwai 4671264608 ALSA: hda - Report errors when invalid values are passed to snd_hda_amp_*()
The values should be in 8 bits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 09:20:39 +02:00
Takashi Iwai 55440e4e37 Merge branch 'fix/hda' into topic/hda 2010-03-29 09:20:32 +02:00
Takashi Iwai 5dbd5ec6e1 ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
The mask and value parameters passed to snd_hda_codec_amp_stereo()
should be 8-bit values for mute and volume.  Passing AMP_IN_MUTE() is
wrong, which is found in many places in patch_realtek.c as a left-over
from the conversion to snd_hda_codec_amp_stereo().

Reported-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 09:18:49 +02:00
Takashi Iwai 85255c0e07 Merge branch 'fix/hda' into for-linus 2010-03-29 08:40:57 +02:00
Takashi Iwai f30c14b64e Merge branch 'fix/misc' into for-linus 2010-03-29 08:40:50 +02:00
Stephen Rothwell 1b132ea03e ASoC: update for removeal of slab.h from percpu.h
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 08:30:23 +02:00
Daniel T Chen 9ec8ddad59 ALSA: hda: Use LPIB for ga-ma770-ud3 board
BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669

The OR states that position_fix=1 is necessary to work around glitching
during volume adjustments using PulseAudio.

Reported-by: Carlos Laviola <claviola@debian.org>
Tested-by: Carlos Laviola <claviola@debian.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 08:26:05 +02:00
Daniel Chen 5cd165e705 ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist
BugLink: https://launchpad.net/bugs/481058

The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense'
need to be muted for sound to be audible, so just add the machine's SSID
to the ac97 jack sense blacklist.

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>
2010-03-29 08:25:20 +02:00
Stephen Rothwell 36db045658 ALSA: usb - use of kmalloc/kfree requires the include of slab.h
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-29 08:23:27 +02:00
Linus Walleij c3635c78e5 DMAENGINE: generic slave control v2
Convert the device_terminate_all() operation on the
DMA engine to a generic device_control() operation
which can now optionally support also pausing and
resuming DMA on a certain channel. Implemented for the
COH 901 318 DMAC as an example.

[dan.j.williams@intel.com: update for timberdale]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Maciej Sosnowski <maciej.sosnowski@intel.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Pavel Machek <pavel@ucw.cz>
Cc: Li Yang <leoli@freescale.com>
Cc: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Cc: Magnus Damm <damm@opensource.se>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Roland Dreier <rdreier@cisco.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
2010-03-26 16:44:01 -07:00
Takashi Iwai 5266874b09 Merge remote branch 'alsa/devel' into topic/hda 2010-03-26 15:28:41 +01:00
Jarkko Nikula 0f17014b34 ALSA: pcm_lib - fix xrun functionality
The commit 4d96eb255c broke the interrupt
time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG
is not set. This is because the xrun() is null defined without it.

Fix this by letting the function xrun() to be always defined as it was
before.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-26 15:26:38 +01:00
Kuninori Morimoto cc780d380a ASoC: fsi: Add FSI2 device support
ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2)
device which is advanced version of FSI.
This patch add simple support for it.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-26 11:17:45 +00:00
Kuninori Morimoto 4a942b457e ASoC: fsi: Add FIFO size calculate
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-26 11:16:27 +00:00
Jaroslav Kysela 079e683ebd ALSA: hda-intel - probe_only module option is int type now
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-03-26 11:16:59 +01:00
Jaroslav Kysela 10e77ddac0 ALSA: hda-intel - remove model=hwio , use probe_only=3 instead
The probe_only module parameter skips the codec initialization, too.
Remove the model=hwio code and use second bit in probe_only to
skip the HDA codec reset procedure.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-03-26 11:08:43 +01:00
Jaroslav Kysela 0bf0e5a6f3 ALSA: hda-intel - AD1984 thinkpad - add analog beep input control
For Lenovo Thinkpad T61/X61, the analog beep input is connected
to node 0x20, index 3. Move the digital beep mute/volume controls
as "Digital Beep" and create analog beep controls for mentioned node.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-03-26 10:37:43 +01:00
Jaroslav Kysela cd508fe58b ALSA: hda-intel - add special 'hwio' model to bypass initialization
Using the 'model=hwio' option, the driver bypasses any codec
initialization and the reset procedure for codecs is also
bypassed. This mode is usefull to enable direct access using
hwdep interface (using hdaverb or hda-analyzer tools) and
retain codec setup from BIOS.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-03-26 10:37:39 +01:00
Daniel T Chen e1f7f02b45 ALSA: ac97: Add IBM ThinkPad R40e to Headphone/Line Jack Sense blacklist
BugLink: https://launchpad.net/bugs/303789

This model needs both 'Headphone Jack Sense' and 'Line Jack Sense'
muted for audible audio, so just add its SSID to the blacklist and
don't enumerate the controls.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-26 08:39:54 +01:00
Sedji Gaouaou ec2755a93d ALSA: AC97: add full duplex support for atmel AT91 and AVR.
This patch add full duplex support on AT91 and AVR.
It was a bug: we needed to check first if there are some chips opened so we
could enable both reception and sending of the data.

Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-25 20:22:36 +01:00
Sedji Gaouaou 7177395fdd ALSA: AC97: add AC97 support for AT91.
This patch add AC97 support for ATMEL AT91, using the AVR32 code.
While AVR is using a DMA, the AT91 chips are using a Peripheral Data
Controller.

Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-25 20:22:15 +01:00
Takashi Iwai 05471e4c44 Merge branch 'fix/hda' into topic/hda 2010-03-25 15:06:58 +01:00
Takashi Iwai 6a4f2ccb46 ALSA: hda - Don't set invalid connection index in Realtek initialiaiton
Skip initialization of connections of DAC widgets that aren't used,
which resulted in invalid verb parameters.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-25 15:00:15 +01:00
Felix Homann fca5bca487 ALSA: usbaudio: Add basic support for M-Audio Fast Track Ultra series
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.

Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-25 12:26:44 +01:00
Dan Carpenter a8462bde78 ASoC: wm8994: playback => capture
Sparse caught that initialize "playback" two times instead of
initializing "capture".

Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-24 14:05:28 +00:00
Kuninori Morimoto 10ea76cc25 ASoC: fsi: IRQ related process had be united
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-24 11:16:47 +00:00
Kuninori Morimoto feb58cffca ASoC: fsi: ensures process inside master lock
Bit operation for fsi_master should be done inside master lock.
But soft-reset/interrupt operation were outside of it.
This patch modify this problem.
It still allow to INT_ST outside-operation on fsi_interrupt,
but it is not problem.
Because this register doesn't need the bit operation.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-24 11:16:47 +00:00
Takashi Iwai 12180024cc Merge branch 'fix/hda' into for-linus 2010-03-24 08:03:38 +01:00
Takashi Iwai b72f1343d6 Merge branch 'fix/asoc' into for-linus 2010-03-24 08:03:34 +01:00
Clemens Ladisch 1c583063a5 ALSA: cmipci: work around invalid PCM pointer
When the CMI8738 FRAME2 register is read, the chip sometimes (probably
when wrapping around) returns an invalid value that would be outside the
programmed DMA buffer. This leads to an inconsistent PCM pointer that is
likely to result in an underrun.

To work around this, read the register multiple times until we get a
valid value; the error state seems to be very short-lived.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Matija Nalis <mnalis-alsadev@voyager.hr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-24 08:02:11 +01:00
Bernhard Urban ae76148114 ALSA: aureon - Patch for suspend/resume for Terratec Aureon cards.
Add proper suspend/resume code for Terratec Aureon cards.
Based on ice1724 suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944
Tested on linux-2.6.32.9

Signed-off-by: Bernhard Urban <lewurm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-23 17:34:23 +01:00
Takashi Iwai 85ae01b2da Merge remote branch 'alsa/devel' into topic/usb 2010-03-23 14:56:33 +01:00
Kuninori Morimoto 1ad747ca9b ASoC: ak4642: Add enhanced sampling rate
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-23 10:59:11 +00:00
Kuninori Morimoto 0643ce8f42 ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-23 10:59:11 +00:00
Kuninori Morimoto 4b6316b4b1 ASoC: ak4642: Add pll select support
Current ak4642 was not able to select pll.
This patch add support it.
It still expect PLL base input pin is MCKI.
see Table 5 "setting of PLL Mode" of datasheet

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-23 10:59:11 +00:00
Mark Brown 778a76e2db ASoC: Implement WM8994 DAI tristate support
This also adds the first DAI operation for AIF3 so fill out the ID and
the ops for that too.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-23 10:57:11 +00:00
Clemens Ladisch 306ff3e473 ALSA: ua101: remove experimental status
Now that the EHCI driver copes with small iso packets without blowing
up, take the snd-ua101 driver out of the alpha-test stage.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-03-23 11:21:53 +01:00
Mark Brown 74511020dd Merge branch 'for-2.6.34' into for-2.6.35 2010-03-22 17:23:46 +00:00
Mark Brown 69266866a5 ASoC: Allow WM8903 mic detect disable and don't force bias on
Don't force enable the microphone bias on WM8903 when doing jack
detection, and don't force enable microphone bias. This allows
platforms to only enable microphone detection when a jack has been
inserted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 17:22:56 +00:00
Mark Brown f06bce9c8c ASoC: Allow disabling of WM835x jack detection
If no report is specified then disable detection. Note that we don't
disable the slow clock, though the power consumption from it should
be negligable. That should be reference counted, ideally through DAPM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 17:22:39 +00:00
Mark Brown 2f14430af5 ASoC: Move WM8350 microphone detection bias managment out of driver
Allow machines to control exactly when the bias is turned on and off.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 17:21:38 +00:00
Mark Brown 5b9e87cccc ASoC: Allow force enabled pins to be disabled
Some systems, such as those with mechanical jack detection, may wish
to force enable a pin (typically mic bias) only some of the time.
Support such systems by having disable_pin() also coveer force enabled
pins.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 17:21:23 +00:00
Mark Brown d5021ec9fc ASoC: Add a notifier for jack status changes
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.

Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 17:20:57 +00:00
Peter Ujfalusi c96907f21f ASoC: TWL4030: PM fix for output amplifiers
Gain controls on outputs affect the power consumption
when the gain is set to non 0 value.

Outputs with amps have one register to configure the
routing and the gain:
PREDL_CTL (0x25):
bit 0: Voice enable
bit 1: Audio L1 enable
bit 2: Audio L2 enable
bit 3: Audio R2 enable
bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB)

bit 0 - 3: is handled in DAPM domain (DAPM_MIXER)
bit 4 - 5: has simple volume control

If there is no audio activity (BIAS_STANDBY), and
user changes the volume, than the output amplifier will
be enabled.
If the user changes the routing (but the codec remains in
BIAS_STANDBY), than the cached gain value also be written
to the register, which enables the amplifier.

The existing workaround for this is to have virtual
PGAs associated with the outputs, and whit DAPM PMD
the gain on the output will be forced to 0 (off) by
bypassing the regcache.
This failed to disable the amplifiers in several
scenario (as mentioned above).

Also if the codec is in BIAS_ON state, and user modifies
a volume control, which path is actually not enabled, than
that amplifier will be enabled as well, but it will
be not turned off, since there is no DAPM path, which
would make mute it.

To prevent amps being enabled, when they are not
needed, introduce the following workaround:
Track the state of each of this type of output.
In twl4030_write only allow actual write, when the
given output is enabled, otherwise only update
the reg_cache.
The PGA event handlers on power up will write the cached
value to the chip (restoring gain, routing selection).
On power down 0 is written to the register (disabling
the amp, and also just in case clearing the routing).

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>
2010-03-22 16:47:12 +00:00
Takashi Iwai 7fb5622326 ALSA: hda - Fix uninitialized variable warning in alc_auto_parse_customize_define()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 17:09:47 +01:00
Randy Dunlap 6407d474e6 ALSA: usb: fix usb build error when PM is not enabled
Fix build errors when CONFIG_PM is not enabled:

sound/usb/card.c:629: error: 'usb_audio_suspend' undeclared here (not in a function)
sound/usb/card.c:630: error: 'usb_audio_resume' undeclared here (not in a function)

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 17:07:36 +01:00
Takashi Iwai 2fb20b6155 Merge branch 'topic/misc' into topic/usb 2010-03-22 17:05:48 +01:00
Daniel Mack 6da7a2aa89 ALSA: usb/caiaq: Add support for Traktor Kontrol X1
This device does not have audio controllers and backlit buttons only.
Input data is handled over a dedicated USB endpoint.

All functions are supported by the driver now.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 16:10:41 +01:00
Mark Brown 3cc4e53f86 ASoC: Remove BROKEN from i.MX audio after dependencies merged
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 11:17:41 +00:00
Mark Brown f9b44121b3 Merge commit 'v2.6.34-rc2' into for-2.6.34 2010-03-22 11:17:26 +00:00
Takashi Iwai bae84e70d6 ALSA: hda - Fix access-after-free in patch_realtek.c
alc_free_kctls() has to be called after all jobs done in alc_build_controls().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 08:30:20 +01:00
Takashi Iwai ea823c0891 ALSA: hda - Sort codec entry list of Nvidia HDMI
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 08:07:55 +01:00
Derek Kelly e933e9e523 ALSA: hda - Add support of Nvidia GT220 HDMI
This patch adds the device id for Nvidia GT220 cards to the nvhdmi
driver.  I have tested it and confirmed it to be working.

Original patch download link:
https://gist.github.com/324070/

Signed-off-by: Derek Kelly <user.vdr@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 08:06:23 +01:00
Daniel T Chen 025f206c9e ALSA: hda: Fix 0 dB offset for HP laptops using CX20551 (Waikiki)
BugLink: https://launchpad.net/bugs/420578

The OR has verified that his hardware distorts because of the 0 dB
offset not corresponding to the highest PCM level. Fix this by capping
said PCM level to 0 dB similarly to what we do for CX20549 (Venice).

Reported-by: Mike Pontillo <pontillo@gmail.com>
Tested-by: Mike Pontillo <pontillo@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-22 08:01:41 +01:00
Takashi Iwai 93929ebc81 Merge branch 'fix/hda' into topic/hda 2010-03-21 09:33:25 +01:00
Kunal Gangakhedkar e3d2530a6c ALSA: hda - Add PCI quirk for HP dv6-1110ax.
Adding this PCI quirk fixes the board config detection.
This also fixes jack sensing by using "hp_detect=1" via properly detected
board config.

Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-21 09:33:11 +01:00
Julia Lawall fc8aa7b16a sound/oss/vidc.c: change the field used with DMA_ACTIVE
The constant DMA_ACTIVE is defined with the dma_buffparams structure rather
than with the audio_operations structure.  Takashi Iwai suggested that the
dmap_out field of the audio_operations structure should be used instead.

This is not tested.

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-21 09:32:09 +01:00
Mark Brown 4ca612ebdb Merge branch 'for-2.6.34' into for-2.6.35 2010-03-19 19:39:23 +00:00
Guennadi Liakhovetski b2dfa62c52 ASoC: remove a card from the list, if instantiation failed
If instantiation of a card failed, we still have to remove it from the
card list on unregistration. This fixes an Oops on Migo-R, triggering,
when after a failed firmware load attempt the driver modules are removed
and re-inserted again.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 19:39:18 +00:00
Daniel Mack fd23b7dee5 ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.

All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.

Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 19:37:29 +00:00
Daniel Mack 8727b909bb ASoC: pxa-pcm-lib: initialize DMA channel to -1
This fixes a warning ("pxa_free_dma: trying to free channel 0 which is
already freed") when a device was opened but the hw_params() call
failed.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 15:28:04 +00:00
Mark Brown 093208f5d0 ASoC: Hook up microphone jack detection on 1133-EV1 board
Note that since all the microphones share a bias there is a single
jack exported for all three, even though there are two physical
connectors plus the soldered down silicon mic.  Note also that the SiMic
is always present by default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-19 14:09:05 +00:00
Mark Brown a655b96c24 Merge branch 'topic/jack' into for-2.6.35 2010-03-19 12:48:10 +00:00
Barry Song 698c375666 ASoC: change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver
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>
2010-03-19 12:47:34 +00:00
Mark Brown cffce322be ASoC: Unexport AD193x bus probe/remove functions
The export is not needed since the per-bus code lives in the same
module.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-19 12:22:03 +00:00
Barry Song a1533d94c6 ASoC: rename ad1938 to ad193x and add support for ad1936/7/8/9
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Yi Li <yi.li@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 12:12:16 +00:00
Misael Lopez Cruz 8ecbabd977 ASoC: TWL6040: Add twl6040 codec driver
Initial version of TWL6040 codec driver.

The TWL6040 codec uses a proprietary PDM-based digital audio interface.
Audio paths supported are:

- Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right
- Output: Headset Left/Right, Handsfree Left/Right

TWL6040 codec supports power-up/down manual and automatic sequence.
Manual sequence is done through a specific register writes sequence.
Automatic sequence is done when the codec is powered-up through the
external AUDPWRON line. The completion of the sequence is signaled
through the audio interrupt.

TWL6040 codec sysclk can be provided by: low-power or high
performance PLL:

- The low-power PLL takes a low-frequency input at 32,768 Hz and
generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz
respectively)

- The high-performance PLL generates an exact 19.2 MHz clock signal
from high-frequency input at 12/19.2/26/38.4 MHz.

Low-power playback mode is a special scenario where only headset path
(headset DAC and driver) is active.

For the particular case of headset path, PLL being used defines the
headset power mode: low-power, high-performance.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 11:29:33 +00:00
Mark Brown 6937c947d3 ASoC: Bail out of wm_hubs DC servo if calibration fails
We're keeping track of the number of times we've iterated but never
actually using this to bail out if the chip looks stuck.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-19 11:17:36 +00:00
Peter Ujfalusi fdb6b1e195 ASoC: tlv320dac33: Internal clocking changes
During validation of the internal clocking setup it has
been found that the following settings were not configured
in an optimal way:

ASRC_CTRL_A: SRCLKDIV was incorrect, instad of divide ratio 3,
             ratio of 2 has to be used (as the comment stated)
DAC_CTRL_A: Fs = Fsref is the desired configuration instead of
            Fs = Fsref / 1.5

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>
2010-03-19 11:17:24 +00:00
Peter Ujfalusi 44f497b4e0 ASoC: tlv320dac33: Fix DSP modes
To make DSP_A mode working correctly the data delay should be
configured to 0. DSP_B mode thus can not be used with DAC33,
so remove it.

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>
2010-03-19 11:17:24 +00:00
Mark Brown 27648b2f1c ASoC: Correct typoed Mic2 connections on 1133-EV1 board
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 11:15:42 +00:00
Peter Ujfalusi 299a151f53 ASoC: omap-mcbsp: Add support for Left Justified format
Basic support for Left Justified coding for OMAP McBSP.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 11:14:39 +00:00
Jorge Eduardo Candelaria 9fc71e8f58 ASoC: McPDM: Use tabs for indentation
Indentation in initial support for McPDM driver was converted to spaces.
Use tabs to comply with open source coding-style.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-19 11:14:39 +00:00
Kailang Yang c027ddcd01 ALSA: hda - Add alc_codec_rename() helper
Added alc_codec_rename() helper for renaming codec->chip_name.
Added Acer-specific codec naming for ALC269/662.

[Clean-up and refactoring by tiwai]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-19 11:40:53 +01:00
Kailang Yang da00c24493 ALSA: hda - Add parse customize define function for Realtek codecs
Added alc_auto_parse_customize_define() to parse the Realtek-specific
attributes from SKU.  Also enable beep controls only when the proper
attribute bit is set.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-19 11:38:53 +01:00
Kailang Yang 6ff86a3f33 ALSA: hda - Take internal mic as Front Mic
Add new check for MIC. Do the internal DMIC as the Front MIC.
It could solve the default record source index issue.

[Fix the check properly using the bitmask by tiwai]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-19 11:14:36 +01:00
Justin P. Mattock b7a5633ab3 fix comment typo in sound/pci/hda/hda_local.h
I think this should be automatic pin instead of ping.
(but could be wrong).

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-03-19 11:00:58 +01:00
Linus Torvalds 01da47059a 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:
  sound: sequencer: clean up remove bogus check
  ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
  ALSA: hda - Disable MSI for Nvidia controller
  ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220
  ALSA: hda - Fix secondary ADC of ALC260 basic model
  ALSA: hda - Add an error message for invalid mapping NID
  ALSA: hda - New Intel HDA controller
2010-03-18 16:48:19 -07:00
Guennadi Liakhovetski da3b062e30 ASoC: SIU driver shall select FW_LOADER
The SIU ASoC driver must load firmware to program the DSP, therefore it
has to select FW_LOADER in its Kconfig entry.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-18 11:31:13 +00:00
Barry Song f4bee1bb00 ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode
The registers for AD193X are defined as 0x800-0x810 for spi which uses
16_8 mode, for i2c to support AD1937, we will use 8_8 mode, only the low
byte of 0x800-0x810 is valid.  The patch will not destory other codecs,
but make soc cache interface more useful.

Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-18 11:23:23 +00:00
Cliff Cai 85dfcdffc2 ASoC: soc-cache: add i2c read entry for 8_8 mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-18 11:23:15 +00:00
Mark Brown ebb812cb8d ALSA: Add support for key reporting via the jack interface
Some devices provide support for detection of a small number of
buttons on their jacks. One common implementation provides a single
button, implemented by shorting the microphone to ground and detected
along with microphone presence detection by detecting varying current
draws on the microphone bias signal.

Provide support for up to three buttons via the jack interface. These
default to reporting BTN_n but an API is provided to allow these to
be remapped to other keys by the machine driver where it knows what
the keys are. More keys can be added with ease if required.

This is only intended to support simple accessory button designs. If
the interface is limiting then either creating a child device for the
accessory or accessing the input device in the jack directly is
recommended.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-17 18:10:46 +00:00
Mark Brown 1c6e555c3a ALSA: Rename jack switch table in preparation for button support
Avoids confusion when we have button support.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-17 18:10:44 +00:00
Mark Brown dd76769dd5 ASoC: Refresh WM8750 bias management
The WM8750 is using some delayed work to manage the ramping of the bias
at startup and resume out of line from the normal flow.  This predates
the support within ASoC core for moving the resume out of line from the
main system resume which provides equivalent functionality with better
interaction with applications.  Change to doing the ramp in line to make
use of the core functionality.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-17 14:09:02 +00:00
Mark Brown a6c65736bc ASoC: Remove current PGA control handling
A code audit reveals that there are currently no users of the widget
controls on PGAs. This is likely to continue to be the case since
while there are useful things that can be done with integrating the
PGA gain and mute controls with the power sequencing userspace
generally wants stereo controls for output stages which this doesn't
map onto well.

In preparation for implementing something more useful strip out the
existing code, leaving the parameters there for use by the new code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:08:31 +00:00
Mark Brown 2a0761a35b ASoC: Implement WM835x microphone jack detection support
The WM8350 provides microphone presence and short circuit detection.
Integrate this with the ASoC jack reporting API.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 09:27:19 +00:00
Takashi Iwai e04dd2d21b Merge branch 'fix/hda' into for-linus 2010-03-17 09:01:38 +01:00