Commit Graph

5115 Commits

Author SHA1 Message Date
Takashi Iwai f1aa298679 Merge branch 'fix/opl3sa2-suspend' into for-linus 2009-03-18 08:04:36 +01:00
Takashi Iwai a232ee66e0 Merge branch 'fix/hda' into for-linus 2009-03-18 08:04:16 +01:00
Takashi Iwai 6af845e4eb ALSA: Fix vunmap and free order in snd_free_sgbuf_pages()
In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG
pages, and it causes errors on Xen as Xen manages the pages
differently.  Although no significant errors have been reported on
the actual hardware, this order should be fixed other way round,
first vunmap() then free pages.

Cc: Jan Beulich <jbeulich@novell.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:04:01 +01:00
Jiri Slaby 82f5d57163 ALSA: mixart, fix lock imbalance
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:03:49 +01:00
Jiri Slaby 91054598f7 ALSA: pcm_oss, fix locking typo
s/mutex_lock/mutex_unlock/ on 2 fail paths in snd_pcm_oss_proc_write.
Probably a typo, lock should be unlocked when leaving the function.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:03:33 +01:00
Viral Mehta 36c7b833e5 ALSA: oss-mixer - Fixes recording gain control
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control for audio recording devices.

Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.

Have tested two usb audio mic devices.

Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:52:28 +01:00
Takashi Iwai 4a10079345 Merge branch 'fix/hda' into topic/hda 2009-03-18 07:50:56 +01:00
Jaroslav Kysela ee5047102c ALSA: snd-hda-intel - add checks for invalid values to *query_supported_pcm()
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.

Also, clean a bit the code.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:50:44 +01:00
Takashi Iwai c673ba1c23 ALSA: hda - Workaround for buggy DMA position on ATI controllers
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:46:21 +01:00
Takashi Iwai 09240cf429 ALSA: hda - Fix DMA mask for ATI controllers
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA.  As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:45:41 +01:00
Mark Brown da88b48b84 Merge branch 'pxa-ssp' into for-2.6.30 2009-03-17 19:07:26 +00:00
Dmitry Artamonow 323a59613e ALSA: drop outdated and broken sa11xx-uda1341 driver
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).

Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-17 17:58:13 +01:00
Takashi Iwai dbe36c9dd5 Merge branch 'topic/snd_card_new-err' into topic/drop-l3 2009-03-17 17:57:37 +01:00
Atsushi Nemoto d2314e0e27 ASoC: Only deregister AC97 dev if it's name was not "AC97"
The commit 14fa43f53f ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister().  This patch adds same condition for
soc_ac97_dev_unregister().  Without this fix, kernel crashes when
unloading an asoc driver.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-17 13:59:47 +00:00
Takashi Iwai 37ba1b6283 Merge branch 'fix/opl3sa2-suspend' into topic/isa-misc 2009-03-17 09:28:13 +01:00
Krzysztof Helt dde332b660 ALSA: opl3sa2 - Fix NULL dereference when suspending snd_opl3sa2
Fix the OOPS during a opl3sa2 card suspend
and resume if the driver is loaded but the card
is not found.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-17 09:27:47 +01:00
Paul Mundt 40f49e7ed7 sh: dma: Make G2 DMA configurable.
Follow the PVR2 DMAC change for G2 DMA.

Signed-off-by: Paul Mundt <lethal@linux-sh.org>
2009-03-17 12:47:56 +09:00
Jonathan Corbet 60aa49243d Rationalize fasync return values
Most fasync implementations do something like:

     return fasync_helper(...);

But fasync_helper() will return a positive value at times - a feature used
in at least one place.  Thus, a number of other drivers do:

     err = fasync_helper(...);
     if (err < 0)
             return err;
     return 0;

In the interests of consistency and more concise code, it makes sense to
map positive return values onto zero where ->fasync() is called.

Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
2009-03-16 08:34:35 -06:00
Jonathan Corbet db1dd4d376 Use f_lock to protect f_flags
Traditionally, changes to struct file->f_flags have been done under BKL
protection, or with no protection at all.  This patch causes all f_flags
changes after file open/creation time to be done under protection of
f_lock.  This allows the removal of some BKL usage and fixes a number of
longstanding (if microscopic) races.

Reviewed-by: Christoph Hellwig <hch@lst.de>
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
2009-03-16 08:32:27 -06:00
Takashi Iwai b9591448e5 ALSA: hda - Fix ALC662 beep again
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 15:26:01 +01:00
Jaroslav Kysela b8dbed0f09 ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 15:23:33 +01:00
Mark Brown 852fd9e50f ASoC: Each PXA AC97 DAI needs a separate ops
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:57 +00:00
Mark Brown f2a5d6a2ea ASoC: Fix some missing dai_ops conversions
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:57 +00:00
Joonyoung Shim 10d9e3d99e ASoC: twl4030 - Fix build error
CC      sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:56 +00:00
Giuliano Pochini 4c55bb0149 ALSA: echoaudio: remove line-out volume from vmixer cards
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 08:38:00 +01:00
Giuliano Pochini 9f5d790d1b ALSA: echoaudio: remove line-out volume from vmixer cards
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.

This patch removes the line-out volume control from vmixer-equipped cards.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 08:37:29 +01:00
Robert Jarzmik 26ade896b6 ASoC: Allow choice of ac97 gpio reset line
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.

This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.

Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-15 20:20:37 +00:00
Mark Brown 85fab7802a ASoC: Fix Zylonite for non-networked SSP mode
This also simplifies the code a bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-14 11:38:16 +00:00
Mark Brown 0ce36c5f7f ASoC: Fix non-networked I2S mode for PXA SSP
Two issues are fixed here:

 - I2S transmits the left frame with the clock low but I don't seem to
   get LRCLK out without SFRMDLY being set so invert SFRMP and set a
   delay.
 - I2S has a clock cycle prior to the first data byte in each channel
   so we need to delay the data by one cycle.

Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-14 11:37:46 +00:00
Russell King 97fb44eb6b Merge branch 'for-rmk' of git://git.pengutronix.de/git/imx/linux-2.6 into devel
Conflicts:

	arch/arm/mach-at91/gpio.c
2009-03-13 21:44:51 +00:00
Takashi Iwai 58d8395b74 ALSA: hda - Add another HP model with IDT92HD71bx codec
HP laptops require GPIO0 on as EAPD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-13 17:04:34 +01:00
Daniel Mack 72d7466468 ASoC: switch PXA SSP driver from network mode to PSP
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.

Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-13 13:23:34 +00:00
Lopez Cruz, Misael 77dd7e17b8 ASoC: Move headset jack registration to device initialization for SDP3430
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-13 12:08:53 +00:00
Takashi Iwai bb6ac72fb1 ALSA: hda - power up before codec initialization
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.

Also, keep power-up during hwdep reconfiguration.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-13 09:06:31 +01:00
Takashi Iwai 307282c899 ALSA: hda - Add model=vaio for STAC9872
Add the default pin config for model=vaio (in case of broken BIOS).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 18:17:58 +01:00
Takashi Iwai 9421f9543b ALSA: hda - Print multiple out-amp values of pin widgets on Conext codecs
Add a flag to work around the non-standard amp-value handling on
Conexant codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 17:06:07 +01:00
Takashi Iwai 3b7523fc82 ALSA: hda - Add comments for the previous fix for conexant codecs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 16:45:01 +01:00
Philipp Zabel eb5f6d753e ASoC: Replace remaining uses of snd_soc_cnew with snd_soc_add_controls.
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-12 15:43:30 +00:00
Mark Brown 6f7cb44ba1 ASoC: Move WM8580 to normal I2C device probe
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-12 15:43:24 +00:00
Gregorio Guidi 5d75bc5578 ALSA: hda - fix headphone settings and master volume (Conexant CX20551)
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.

Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 16:41:51 +01:00
Mark Brown 65ec1cd1e2 ASoC: Merge dai_ops factor out
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line.  Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 16:51:31 +00:00
Mark Brown 5314adc361 ASoC: Fix formats for s3c24xx-i2s register prints
The register values are all u32 so don't need the long format.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 16:28:29 +00:00
Mark Brown 02b7cbc399 ASoC: Remove version display from WM8580 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 14:40:41 +00:00
Mark Brown aaf1e176fa ASoC: Add initial driver for the WM8400 CODEC
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application.  This driver supports the
primary audio CODEC features, including:

 - 1W speaker driver
 - Fully differential headphone output
 - Up to 4 differential microphone inputs

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 13:49:46 +00:00
David Brownell 5706d50132 ASoC: buildfix for OSK
Buildfix:

  CC      sound/soc/omap/osk5912.o
  sound/soc/omap/osk5912.c: In function 'osk_soc_init':
  sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
  make[3]: *** [sound/soc/omap/osk5912.o] Error 1

There's no such (standard) clock interface.

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 12:49:28 +00:00
Daniel Mack cbf1146d5e ASoC: don't touch pxa-ssp registers when stream is running
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 19:44:04 +00:00
Hugo Villeneuve 090cec81ae ALSA: ASoC: Davinci: Updated sffsdr_hw_params() function to new format
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 15:42:48 +00:00
Hugo Villeneuve 14cbba89ae ALSA: ASoC: Davinci: Replaced DAI format RIGHT_J by DSP_B for SFFSDR
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 15:42:48 +00:00
Mark Brown b3d7e3c99d Merge commit 'takashi/topic/asoc' into for-2.6.30 2009-03-10 15:42:03 +00:00
Takashi Iwai df481e41b9 ALSA: hda - Clean up Cxt5047 parser
Clean up Conexant 5047 pareser code:
 - Split mixer elements to separate arrays to reduce the duplicated
   entires
 - Fix mixer element names to the standard ones
 - Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
   handler works fine.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 15:35:35 +01:00
Takashi Iwai 5b3a7440cb ALSA: hda - Fix / clean up init verbs for Cxt5047 codec
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.

Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 15:13:25 +01:00
Takashi Iwai 3b628867f3 ALSA: hda - Remove superfluous verbs for Cxt5047 laptop-eapd model
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 15:13:24 +01:00
Takashi Iwai b880c74adf ALSA: hda - Create "Capture Source" control dynamically in patch_conexant.c
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 15:13:23 +01:00
Takashi Iwai dd5746a85c ALSA: hda - Create vmaster for conexant codecs
Instead of binding volumes, create a virtual master volume for Conexant
codecs.  This allows separate HP and speaker volume controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 15:13:17 +01:00
Takashi Iwai 6fce61aeaf ALSA: hda - Fix coding style issues in last two patches
Also re-ordered the quirk entries per SSID.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 07:49:50 +01:00
Christoph Plattner 443e26d014 ALSA: hda - Rework on patch_sigmatel.c for HP HDX16/HDX18
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).

Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 07:36:19 +01:00
Christoph Plattner ae6241fbf5 ALSA: hda - Added HP HDX16/HDX18 notebook support for HDA codecs (82HD71)
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.

Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-10 07:35:20 +01:00
Mark Brown 6b849bcff0 ASoC: Convert PXA AC97 driver to probe with the platform device
This will break any boards that don't register the AC97 controller
device due to using ASoC.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-09 18:19:01 +00:00
Takashi Iwai 9a1b64caac ALSA: rawmidi - Refactor rawmidi open/close codes
Refactor rawmidi open/close code messes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 15:17:23 +01:00
Takashi Iwai f9d202833d ALSA: rawmidi - Fix possible race in open
The module refcount should be handled in the register_mutex to avoid
possible races with module unloading.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 15:17:21 +01:00
Takashi Iwai 118dd6bfe7 ALSA: Clean up snd_monitor_file management
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 15:16:11 +01:00
Takashi Iwai 79c7cdd544 ALSA: Add kernel-doc comments to vmaster stuff
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 15:10:01 +01:00
Roel Kluin 3966175863 ALSA: snd-powermac: timeout reaches -1
If unsuccessful, timeout reaches -1 after the loop.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:37 +01:00
Takashi Iwai 6da6711385 ALSA: powermac - Add missing KERN_* prefix to printk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:31 +01:00
Risto Suominen dca7c74172 ALSA: Add vmaster controls for Pmac 5500, iMac G3 SL, and PBook G3 Lombard
Add virtual master controls for PowerMac 5500 (AWACS) and iMac G3 Slot-loading
and PowerBook G3 Lombard (Screamer).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:26 +01:00
Risto Suominen ed336d3404 ALSA: powermac - Allow input from mic in iBook G3 Dual-USB
Allow input from microphone on iBook G3 Dual-USB (Tumbler).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:19 +01:00
Risto Suominen 4d9e93b1ad ALSA: powermac - Correct volume controls and HP detection for PMac 8500/9500
Correct volume controls and headphone detection for PowerMac 8500/9500 (AWACS).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:13 +01:00
Risto Suominen 573934bc03 ALSA: powermac - Correct volume controls for PowerBook G3 Lombard
Correct volume controls for PowerBook G3 Lombard (Screamer).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:07 +01:00
Risto Suominen b0a8a8fd1b ALSA: powermac - Correct HP detection and input selectors for PMac 5500
Correct headphone detection and input selectors for PowerMac 5500 (AWACS).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:58:01 +01:00
Takashi Iwai f5b1db6342 ALSA: add snd_ctl_add_slave_uncached()
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls.  The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks.  OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.

The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:56:19 +01:00
Eric Miao 5742964e91 [ARM] pxa: remove unnecessary #include of pxa-regs.h and hardware.h
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
2009-03-09 21:22:38 +08:00
Eric Miao 7ebc8d56f4 [ARM] pxa: move DMA registers definitions into <mach/dma.h>
1. Driver code where pxa_request_dma() is called will most likely
   reference DMA registers as well,  and it is really unnecessary
   to include pxa-regs.h just for this. Move the definitions into
   <mach/dma.h> and make relevant drivers include it instead of
   <mach/pxa-regs.h>.

2. Introduce DMAC_REGS_VIRT as the virtual address base for these
   DMA registers. This allows later processors to re-use the same
   IP while registers may start at different I/O address.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
2009-03-09 21:22:36 +08:00
Takashi Iwai 85122ea40c ALSA: Remove unneeded snd_pcm_substream.timer_lock
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 14:02:00 +01:00
Takashi Iwai ed3da3d9a0 ALSA: Rewrite hw_ptr updaters
Clean up and improve snd_pcm_update_hw_ptr*() functions.

snd_pcm_update_hw_ptr() tries to detect the unexpected hwptr jumps
more strictly to avoid the position mess-up, which often results in
the bad quality I/O with pulseaudio.

The hw-ptr skip error messages are printed when xrun proc is set to
non-zero.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 12:56:49 +01:00
Takashi Iwai 0a4e1c9069 Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-03-09 12:05:21 +01:00
Daniel Mack a381934e5f ASoC: Add a driver for AK4104 S/PDIF transmitter
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-09 10:46:17 +00:00
Clemens Ladisch 873591db59 sound: oxygen: enable headphone output on Claro cards
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 09:45:11 +01:00
Takashi Iwai f271fa28fb ASoC: Fix Kconfig dependency of CONFIG_SND_S3C24XX_SOC_JIVE_WM8750
Remove a non-existing Kconfig CONFIG_SND_SOC_WM8750_SPI.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-09 00:52:17 +01:00
Mark Brown 055a49b0c9 ASoC: Remove unneeded forward reference to WM8753 SPI implementation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-08 20:43:33 +00:00
Daniel Mack b191f63c4f ASoC: bring cs4270 feature/limitations list in sync
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-08 18:27:36 +00:00
Linus Torvalds d3dea1e2d5 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix headphone-detect regression with multiple HP jacks
  ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
2009-03-08 10:03:31 -07:00
Timur Tabi 3a638ff272 ASoC: Improve pause/unpause performance in Freescale 8610 drivers
Add support for true pause and unpause.  Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.

Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.

Optimize the delay after starting capture.  Instead of delaying 1ms, the driver
now polls the hardware.  The new delay is shorter by over 90% yet still
effective.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-07 11:01:49 +00:00
Hugo Villeneuve 96deff6baf ASoC: Davinci: Fix incorrect machine type for SFFSDR board
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-07 11:01:40 +00:00
Mark Brown b52a5195ef ASoC: Fix logging severity for some S3C error messages
Upgrade the severity of some failure messages from debug level so
they're displayed by default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 18:13:43 +00:00
Mark Brown ee7d476714 ASoC: Re-remove hand-rolled pr_debug() macros
The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 18:04:34 +00:00
Mark Brown 26bd7b496c ASoC: Staticise workqueue function for GPIO jack detection
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:19 +00:00
Mike Frysinger 67a9c573b5 ASoC: Blackfin: fix typo in MUTE definition
Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:18 +00:00
Mike Frysinger 3465d93a12 ASoC: Blackfin: move gpio_err behind the define that is only user of it
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:17 +00:00
Lopez Cruz, Misael de0b988828 ASoC: Add headset jack detection for SDP3430 machine driver
Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:17 +00:00
Timur Tabi a454dad19e ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers
Add a new device tree property for the SSI node: "fsl,ssi-asynchronous".  If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode.  In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS.  Asynchronous mode allows playback and capture to
use different sample sizes.  It also technically allows different sample rates,
but the driver does not support that.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:16 +00:00
Mark Brown 499d8f4a52 ASoC: Update Kconfig for Samsung CPUs to reflect S3C64xx support
We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:15 +00:00
Mark Brown 07495f3e5a ASoC: Fix memory allocation for snd_soc_dapm_switch names
snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.

Also fix the coding style for the switch below while we're here.

Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:14 +00:00
Mark Brown 42aa3418eb ASoC: Factor out DAPM widget power check into separate function
Essentially simple code motion to facilitate refactoring of the power
decisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:13 +00:00
Daniel Mack 20a41eac4f ASoC: Fix name of register bit in pxa-ssp
A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:12 +00:00
Peter Ujfalusi 89492be886 ASoC: TWL4030: Make the HS ramp delay configurable
Enum type for selecting the desired ramp delay for the headset output.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:11 +00:00
Mark Brown a1b3eaeb14 ASoC: Refresh JIVE driver
Remove uneeded startup callback and use snd_soc_dapm_nc_pin()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:10 +00:00
Ben Dooks c36623a754 ASoC: Select DMA if I2S is configured
Select the relevant DMA implementation when the
sound driver is selected.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:09 +00:00
Ben Dooks f8cf8176c7 ASoC: Add s3c64xx-i2s support
Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:08 +00:00
Ben Dooks dc85447b19 ASoC: Split s3c2412-i2s.c into core and SoC specific parts
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.

As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:08 +00:00
Ben Dooks 3093e48c48 ASoC: Add JIVE audio support
Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:07 +00:00
Lopez Cruz, Misael 979c036e09 ASoC: Add DAPM machine widgets to SDP3430 driver
Add DAPM machine domain widgets to SDP3430 machine driver.
Interconnection:
* Ext Mic: MAINMIC, SUBMIC
* Ext Spk: HFL, HFR
* Headset Jack: HSMIC, HSOL, HSOR

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:06 +00:00
Mark Brown 4f5b31c3f2 Merge commit 's3c-iis-header' into HEAD 2009-03-06 13:36:44 +00:00
Takashi Iwai 90f349d96e ALSA: ac97 - Add patch entry for Conexant CX20468-31 chip
Added the patch entry for Conexant CX20468-31 chip (4358:5430).

Reference: Novell bnc#471265
	https://bugzilla.novell.com/show_bug.cgi?id=471265

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 14:30:08 +01:00
Takashi Iwai 139e071b0f ALSA: hda - Assign HP and speaker DACs before mic/line-in
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 12:10:41 +01:00
Takashi Iwai ee58a7ca21 ALSA: hda - Connect to primary DAC if no individual DAC is available
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin.  This ensures that the pin
works somehow at least.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 12:00:24 +01:00
Takashi Iwai 668b9652be ALSA: hda - Create multiple HP / speaker controls with index
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 10:13:24 +01:00
Takashi Iwai 7a411ee01b ALSA: hda - Allow slave controls with non-zero indices
Fix snd_hda_add_vmaster() to check the non-zero indices of slave controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 10:08:14 +01:00
Takashi Iwai dc04d1b4d2 ALSA: hda - Create output controls according to pin types for IDT/STAC
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.

Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 10:04:24 +01:00
Takashi Iwai b3225190c1 Merge branch 'fix/hda' into topic/hda 2009-03-06 09:52:36 +01:00
Takashi Iwai c50ff7c042 ALSA: hda - Fix headphone-detect regression with multiple HP jacks
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output.  Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.

Reference: Novell bnc#482052
	https://bugzilla.novell.com/show_bug.cgi?id=482052

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 09:47:22 +01:00
Takashi Iwai 14b97595e0 ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-06 09:42:07 +01:00
Takashi Iwai f03d3115a6 ALSA: Fix sample rate of Lenovo Ideapad to 44.1kHz
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-05 14:18:26 +01:00
Ben Dooks 899e6cf5e6 S3C: Move <mach/audio.h> to <plat/audio.h>
The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-05 12:01:00 +00:00
Ben Dooks 8150bc886b S3C24XX: Move and update IIS headers
Move the IIS headers to their correct place.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-05 12:00:59 +00:00
Takashi Iwai 37db623ae2 ALSA: hda - Fix check of ALC888S-VC in alc888_coef_init()
Fixed the wrong bits check to identify ALC888S-VC model in
alc888_coef_init().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-05 09:40:16 +01:00
Takashi Iwai c2503cd3be ALSA: hdsp - Ignore MIDI and PCM events in interrupts until initialized
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly.  Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-05 09:37:40 +01:00
Eric Miao 6335d05548 ASoC: make ops a pointer in 'struct snd_soc_dai'
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.

The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-04 22:29:47 +00:00
Herton Ronaldo Krzesinski 7ec30f0e77 ALSA: hda - Map 3stack-hp model (ALC888) for HP Educ.ar
Added model=3stack-hp for HP Educ.ar desktop machine (103c:2a72).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-04 21:05:40 +01:00
Herton Ronaldo Krzesinski 8718b700cc ALSA: hda - Add headphone automute support for 3stack-hp model (ALC888)
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-04 21:05:25 +01:00
Herton Ronaldo Krzesinski 3ea0d7cf47 ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888)
Add additional 4 channel mode for 3stack-hp models.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-04 21:05:11 +01:00
Jonas Andersson 86027ae78c ASoC: wm8510 pll settings
When setting WM8510_MCLKDIV the pll was turned off.

When setting pll frequency you got twice the expected freq, because
the  code calculated  with postscaler of 8,  but  the hardware divide by 4.

Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-04 14:47:39 +00:00
Lopez Cruz, Misael ec67624d33 ASoC: Add GPIO support for jack reporting interface
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.

Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.

All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-04 14:47:38 +00:00
Takashi Iwai bd6afe3f34 ALSA: hda - Fix conflict of mixer controls on Sony VAIO VGN-AR71S
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx.  But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec.  For this device, the model=auto must be chosen
to work properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-04 11:30:25 +01:00
Takashi Iwai 79d7d5333b ALSA: hda - Fix HP dv6736 mic input
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.

Reference: Novell bnc#480753
	https://bugzilla.novell.com/show_bug.cgi?id=480753

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-04 09:03:50 +01:00
Ingo Molnar 8b0e5860cb Merge branches 'x86/apic', 'x86/cpu', 'x86/fixmap', 'x86/mm', 'x86/sched', 'x86/setup-lzma', 'x86/signal' and 'x86/urgent' into x86/core 2009-03-04 02:22:31 +01:00
Philipp Zabel 5f2a9384a9 ASoC: UDA1380: DATAI is slave only
Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:58:51 +00:00
Philipp Zabel aa4ef01de5 ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:54:11 +00:00
Philipp Zabel ef9e5e5c31 ASoC: UDA1380: change decimator/interpolator register handling
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).

* Queue work in the alsa PCM_START .trigger to flush registers
  as soon as the link is running. This replaces the .prepare
  and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
  its alsa control to avoid confusion.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:54:10 +00:00
Philipp Zabel a3c7729e6c ASoC: Remove version display from the UDA1380 driver
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:54:10 +00:00
Takashi Iwai 82ad39f939 ALSA: hda - Fix gcc compile warning
It's false positive, but annoying.
  sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
  sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-03 15:00:35 +01:00
Linus Torvalds bd5e89c813 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
  ALSA: hda - Add quirk for new HP xw series
  ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
2009-03-02 15:47:19 -08:00
Takashi Iwai 6565e4faca ALSA: hda - Add more hint options for IDT/Sigmatel codecs
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.

For example, to disable hp_detect on the fly,
	# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 17:30:00 +01:00
Takashi Iwai d78d7a90ad ALSA: hda - Create "Analog Loopback" controls optionally
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 17:30:00 +01:00
Takashi Iwai ab1726f920 ALSA: hda - Add show for init_verbs and hints sysfs entries
Added the show method for init_verbs and hints hwdep sysfs entries.
They show the current values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 17:29:53 +01:00
Takashi Iwai 43b62713f6 ALSA: hda - Add hint string helper functions
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.

Internally, the hint is stored in a pair of two strings, key and val.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 17:28:54 +01:00
Daniel Mack ff09d49ad0 ASoC: fix typo and removed unneeded switch case for cs4270
This removes a misspelled comment and got rid of superfluous switch
case.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-02 14:39:23 +00:00
Clemens Ladisch b1c86bb807 sound: usb-audio: fix queue length check for high speed devices
When checking for the maximum queue length, we have to take into account
that MAX_QUEUE is measured in milliseconds (i.e., frames) while the unit
of urb_packs is whatever data packet interval the device uses (possibly
less than one frame when using high speed devices).

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 14:50:01 +01:00
Clemens Ladisch eab2b553c3 sound: usb-audio: fix rules check for 32-channel devices
When storing the channel numbers used by a format, and if the device
happens to support 32 channels, the code would try to store 1<<32 in
a 32-bit value.

Since no valid format can have zero channels, we can use 1<<(channels-1)
instead of 1<<channels so that all the channel numbers that we test for
fit into 32 bits.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 14:49:58 +01:00
Krzysztof Helt 1713c0d508 ALSA: opl3sa2 fix irq releasing and short name of card
Two simple fixes:

1. Use the same pointer for the free_irq() and
   the request_irq() calls.

2. A short name of card is appended with '2' or '3'
   character depending on a detected chip. Remove
   the '2' character from the short name.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 12:21:01 +01:00
Takashi Iwai 6e655bf216 ALSA: hda - Don't return a fatal error at PCM-creation errors
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 10:46:30 +01:00
Takashi Iwai f93d461bcd ALSA: hda - Revert the codec probe at control-creation errors
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 10:44:15 +01:00
Takashi Iwai d1f1af2dbf ALSA: hda - Intialize more codec fields in snd_hda_codec_reset()
Initiailize forgotten fields in snd_hda_codec_reset().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 10:35:29 +01:00
Takashi Iwai 4c4531d64d ALSA: hda - Remove Toshiba probe_mask quirk
Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 08:06:11 +01:00
Takashi Iwai 892981ffbe ALSA: hda - Don't create a beep control for digital-only ALC268
When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 08:04:35 +01:00
Takashi Iwai b31b43e9fb Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/hda_intel.c
2009-03-02 08:04:10 +01:00
Takashi Iwai 38f1df27e3 ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts
with the primary codec.  The codec#3 is for the digital I/O, and should
be fixed by the driver, but it'd need a bunch of changes.

So, let's fix the probe problem temporarily by setting the default
probe_mask value.

Reference: kernel bugzilla #12735
	http://bugzilla.kernel.org/show_bug.cgi?id=12735

Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 07:55:52 +01:00
Ingo Molnar 55f2b78995 Merge branch 'x86/urgent' into x86/pat 2009-03-01 12:47:58 +01:00
Mark Brown 8b37dbd2a1 ASoC: Add SND_SOC_DAPM_PIN_SWITCH controls for exposing DAPM pins
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-28 21:31:21 +00:00
Daniel Mack 4eae080dda ASoC: Add cs4270 support for slave mode configurations
Added support for scenarios where the Cirrus CS4270 audio codec is slave
to the bitclk and lrclk. Mixed setups are unsupported.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-28 17:18:25 +00:00
Ben Dooks c8efef1745 ASoC: Fix copyright statements on Simtec files
Fix the copyright statements in two of the S3C24XX ASoC files
that have (c) when we require the full word.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-28 17:18:24 +00:00
Takashi Iwai c82c8abdee ALSA: hda - Fix an "unused variable" compile warning
Forgot to remove an unused variable.
  sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’:
  sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 17:52:22 +01:00
Takashi Iwai 53eff7e1e0 ALSA: hda - Match all 103c:17xx devices for HP BPC model
Use SND_PCI_QUIRK_MASK() to match all devices with 103c:17xx for
HP BPC model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 17:49:44 +01:00
Takashi Iwai f897497673 Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_realtek.c
2009-02-27 17:47:31 +01:00
Takashi Iwai bb543c9694 ALSA: hda - Add quirk for new HP xw series
Added model=hp-bpc for new HP xw series (103c:170b).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 17:44:07 +01:00
Takashi Iwai ea18aa4644 ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models
of IDT 92HD71bxx codec, which was wrongly set to zero.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 17:36:33 +01:00
Clemens Ladisch 82af308f65 sound: oxygen: zero-initialize model data
Model drivers assume that model_data is zeroed, so we better use
kzalloc() (like we did before when it was allocated together with the
card structure).

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 17:32:21 +01:00
peerchen bedfcebb4f ALSA: hda - Add the Device IDs for MCP89 and remove IDs of MCP7B
Added the Device IDs for MCP89 HD audio controller.
Removed the IDs of MCP7B cause this chipset had been cancelled.

Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-27 10:22:55 +01:00
Takashi Iwai 1607b8ea0a ALSA: hda - Add model=auto for STAC/IDT codecs
Added the model=auto to STAC/IDT codecs to use the BIOS default setup
explicitly.  It can be used to disable the device-specific model quirk
in the driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 16:50:43 +01:00
Takashi Iwai 23f0c048ba ALSA: hda - Clean up the input pin setup in automatic mode
Clean up the input-pin setup in automatic mode in patch_realtek.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 13:03:58 +01:00
Takashi Iwai 6d5643455c ASoC: wm8753 - Fix build error
sound/soc/codecs/wm8753.c: In function 'wm8753_probe':
sound/soc/codecs/wm8753.c:1577: error: implicit declaration of function 'wm8753_add_controls'

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 11:29:58 +01:00
Hannes Eder 5d9b6c0783 ALSA: sound/pci/hda: fix sparse warning: different signedness
Fix this sparse warning:
  sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness)
  sound/pci/hda/hda_codec.c:1544:19:    expected unsigned long *vals
  sound/pci/hda/hda_codec.c:1544:19:    got long *<noident>

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:59:04 +01:00
Hannes Eder 730d45f913 ALSA: sound/pci/emu10k1: fix sparse warning: different signedness
Fix this sparse warnings:
  sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness)

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:57:53 +01:00
Hannes Eder d73d341d39 ALSA: sound/drivers/vx: fix sparse warning: different signedness
Fix this sparse warning:
  sound/drivers/vx/vx_uer.c:301:42: warning: incorrect type in argument 2 (different signedness)

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:57:21 +01:00
Hannes Eder 3a755ec2e8 ALSA: sound/usb/usx2y: fix sparse warning: do-while statement is not a compound ...
Fix this sparse warning:
  sound/usb/usx2y/usbusx2y.c:231:33: warning: do-while statement is not a compound statement

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:57:19 +01:00
Hannes Eder 619389882b ALSA: sound/usb/usx2y: fix sparse warning: Should it be static?
Impact: Move declaration to header file.

Fix this sparse warning:
  sound/usb/usx2y/usx2yhwdeppcm.c:739:5: warning: symbol 'usX2Y_hwdep_pcm_new' was not declared. Should it be static?

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:57:17 +01:00
Hannes Eder e5bf484373 sound/oss: fix sparse warning: symbol shadows an earlier one
Impact: Move variable to a more inner scope.

Fix this sparse warning:
  sound/oss/sequencer.c:235:29: warning: symbol 'err' shadows an earlier one
  sound/oss/sequencer.c:215:13: originally declared here

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:55:30 +01:00
Hannes Eder 5d44aa4c73 sound/oss: fix sparse warnings: different signedness
Impact: Change signature of 'set_volume_stereo' and 'set_volume_mono'.

Fix this sparse warnings:
  sound/oss/pss.c:545:42: warning: incorrect type in argument 2 (different signedness)
  sound/oss/pss.c:546:42: warning: incorrect type in argument 3 (different signedness)
  sound/oss/pss.c:554:59: warning: incorrect type in argument 2 (different signedness)
  sound/oss/pss.c:560:59: warning: incorrect type in argument 2 (different signedness)
  sound/oss/pss.c:566:59: warning: incorrect type in argument 2 (different signedness)

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:55:17 +01:00
Clemens Ladisch 930738de60 sound: virtuoso: add Xonar Essence STX support
Add support for the Asus Xonar Essence STX sound card.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 09:39:08 +01:00
Takashi Iwai f872a9194c ALSA: hda - Clean up / fix quirk for Sony laptops with ALC262
Clean up / fix quirk entries for Sony laptops with ALC262 codec
using NSD_PCI_QUIRK_MASK().

This also fixes the kernel bug #12780
	http://bugme.linux-foundation.org/show_bug.cgi?id=12780

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-26 00:57:01 +01:00
Takashi Iwai 873dc78a86 ALSA: hda - Clean up / fix quirks for HP laptops with AD1984A
Use SND_PCI_QUIRK_MASK() to clean up / support better HP laptops with
AD1984A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-25 18:12:13 +01:00
Takashi Iwai 1440566f2d Merge branch 'fix/misc' into for-linus 2009-02-25 09:52:42 +01:00
Takashi Iwai 308b892cb4 Merge branch 'fix/hda' into for-linus 2009-02-25 09:52:38 +01:00
Mark Brown e611bd8244 ASoC: Only write back non-default registers when resuming WM8753
This will reduce the number of writes done on resume, allowing that to
complete faster (especially on systems with very slow I2C like the
current Samsung driver).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-24 23:49:56 +00:00
Mark Brown c2bac1606a ASoC: Convert WM8753 to register via normal device probe
The base support for the only in-tree user, the GTA01, is out of tree
and will be updated separately.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-24 23:49:53 +00:00
Mark Brown 69e169da5a ASoC: Shuffle WM8753 device registration code
This patch should be pure code motion, separating that out from the
functional changes to move to new style device registration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-24 23:48:14 +00:00
Mark Brown d3b8942184 ASoC: Fix Zylonite voice interface stereo configurations
We always run in the first timeslot of one.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-24 23:48:12 +00:00
Mark Brown 8056d9bbb5 ASoC: Improve WM9713 voice DAC shutdown procedure
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-24 23:48:11 +00:00
Takashi Iwai 1f9da55440 ALSA: emu10k1 - Fix digital/analog switch on audigy2 ZS
Fix the inverted logic of shared spdif switch.

Reference: Novell bnc#478496
	https://bugzilla.novell.com/show_bug.cgi?id=478496

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-24 15:31:02 +01:00
Takashi Iwai a65d629ceb ALSA: hda - Add pseudo device-locking for clear/reconfig
Added the pseudo device-locking using card->shutdown flag to avoid
the crash via clear/reconfig during operations.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 16:57:04 +01:00
Takashi Iwai 209b140336 Merge branch 'test/hda-pincfg' into topic/hda 2009-02-23 14:15:47 +01:00
Takashi Iwai 13c989beba ALSA: hda - Don't give over 0dB volume for AD1984A HP laptops
Set the upper limit 0dB to the volume of mixer amp 0x20 for
AD1984A HP laptops.  The overloaded volume may damage the internal
speaker.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 11:33:34 +01:00
Takashi Iwai 5e7b8e0d87 ALSA: hda - Make user_pin overriding the driver setup
Make user_pin overriding even the driver pincfg, e.g. the static / fixed
pin config table in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 09:45:59 +01:00
Takashi Iwai 346ff70fdb ALSA: hda - Rename {override,cur}_pin with {user,driver}_pin
Rename from override_pin and cur_pin with user_pin and driver_pin,
respectively, to be a bit more intuitive.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 09:42:57 +01:00
Takashi Iwai c17a1abae2 ALSA: hda - Use snd_hda_codec_get_pincfg() in the rest places
Replace with snd_hda_codec_get_pincfg() in the places where available.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 09:28:12 +01:00
Tim Blechmann f9ffc5d6f0 ALSA: hdsp - whitespace cleanup
Impact: remove trailing spaces

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 08:28:02 +01:00
Tim Blechmann e588ed8304 ALSA: hdsp - poll for iobox
sleeping for 2 seconds before checking for the iobox is not enough
on some systems.
this patch increases the timeout, but polls the card during that
time. it thus speeds up the module loading when the card has already
been initialized, while being more robust on systems, which require
a higher timeout than the predefined 2 seconds.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 08:20:01 +01:00
Takashi Iwai 66a101dda6 Merge branch 'topic/hwdep-cleanup' into topic/hdsp 2009-02-23 08:17:28 +01:00
Takashi Iwai 1618a3281b Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_realtek.c
2009-02-23 08:14:41 +01:00
Juan Jesus Garcia de Soria cc374c477c ALSA: hda - Quirk for Acer Aspire 6530G
The Acer Aspire 6530G needs the 4930G "model" for the front mic to
work properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 08:13:53 +01:00
Luke Yelavich 2d46638160 ALSA: hda - add another MacBook Pro 3,1 SSID
Reference: Ubuntu bug #33245
    https://bugs.launchpad.net/bugs/332456

Signed-off-by: Luke Yelavich <themuso@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 07:54:26 +01:00
Steve Chen 5370d96f85 ALSA: fix excessive background noise introduced by OSS emulation rate shrink
Incorrect variable was used to get the next sample which caused S2
to be stuck with the same value resulting in loud background noise.

Signed-off-by: Steve Chen <schen at mvista.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 07:49:04 +01:00
Andreas Mohr ce71bfd1aa ALSA: ALS4000, slight mixer improvements
- add 8kHz / 20 kHz low-pass filter switch control
- add ALS4000 Mono capture route control
- add annotations to specs pages
- improve ALS4000 PM saved regs selection (remove SB dummy register,
  add missing ones)
- add some missing ALS4000 register defines
- constify two variables

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 07:47:52 +01:00
Anssi Hannula e8bf069c41 ALSA: aw2: do not grab every saa7146 based device
Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not
check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver
grabs them as well.

According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the
subsystem ids set to 0, the saa7146 default.

Fix conflicts with DVB devices by checking for subsystem ids = 0
specifically.

Signed-off-by: Anssi Hannula <anssi.hannula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 07:41:25 +01:00
Michael Schwingen cc95948972 ALSA: hda - add support for "Maxdata Favorit 100XS" (Intel HDA/ALC260)
Signed-off-by: Michael Schwingen <michael@schwingen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-23 07:39:34 +01:00
Ingo Molnar fc6fc7f1b1 Merge branch 'linus' into x86/apic
Conflicts:
	arch/x86/mach-default/setup.c

Semantic conflict resolution:
	arch/x86/kernel/setup.c

Signed-off-by: Ingo Molnar <mingo@elte.hu>
2009-02-22 20:05:19 +01:00
Geert Uytterhoeven 3d92e8f3ae m68k: atari - Rename "mfp" to "st_mfp"
http://kisskb.ellerman.id.au/kisskb/buildresult/72115/:
| net/mac80211/ieee80211_i.h:327: error: syntax error before 'volatile'
| net/mac80211/ieee80211_i.h:350: error: syntax error before '}' token
| net/mac80211/ieee80211_i.h:455: error: field 'sta' has incomplete type
| distcc[19430] ERROR: compile net/mac80211/main.c on sprygo/32 failed

This is caused by

| # define mfp ((*(volatile struct MFP*)MFP_BAS))

in arch/m68k/include/asm/atarihw.h, which conflicts with the new "mfp" enum in
net/mac80211/ieee80211_i.h.

Rename "mfp" to "st_mfp", as it's a way too generic name for a global #define.

Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2009-02-22 09:23:02 -08:00
Mark Brown 93e051d277 ASoC: Only unregister drivers we registered for WM8753
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-22 14:24:00 +00:00
Mark Brown eeb1080b29 ASoC: Report I/O errors from WM8753 reset
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-22 14:19:23 +00:00
Ingo Molnar 3b6f7b9beb Merge branch 'x86/urgent' into x86/core 2009-02-20 17:40:43 +01:00
Takashi Iwai 2f334f92cf ALSA: hda - Remove codec-specific pin save/restore functions
Replace the accessor to pin defaults with the common code for caching.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 16:48:40 +01:00
Takashi Iwai 330ee99579 ALSA: hda - Remove IDT codec-specific pin save/restore functions
Removed its own save/restore functions and replaced with the common code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 16:48:39 +01:00
Takashi Iwai 0e8a21b59d ALSA: hda - Remove realtek codec-specific pin save/restore functions
Now it's done in the common code.
Also use the common access functions for pin defaults.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 16:48:38 +01:00
Takashi Iwai 3be141494a ALSA: hda - Add generic pincfg initialization
Added the generic pincfg cache and save/restore functions.
Also introduced the pin-overriding via hwdep sysfs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 16:48:26 +01:00
Takashi Iwai d2f57cd54a Merge branch 'fix/hda' into topic/hda 2009-02-20 16:06:47 +01:00
Takashi Iwai 55290e1932 ALSA: hda - Fix parse of init_verbs sysfs entry
Fixed the parse of init_verbs hwdep sysfs entry.
Simplieied using sscanf.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 16:06:30 +01:00
Clemens Ladisch f3990e610a sound: usb-audio: remove MIN_PACKS_URB
Remove the MIN_PACKS_URB symbol because other limits can force the
number of packets down to one, regardless of the value of this symbol,
and nobody has ever changed it anyway.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 11:14:55 +01:00
Clemens Ladisch eacbb9dba6 sound: virtuoso: increase minimum volume to -60 dB
Use -60 dB as the minimum value of the master volume mixer control.
While the DACs would support ranges down to about -120 dB, such
attenuations are not useful in practice.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 11:14:25 +01:00
Clemens Ladisch d91b424d6d sound: oxygen: handle AK5385 ADC on Claro halo cards
The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we
should handle the ADC parameters as we do with the X-Meridian.

Using the code for the wrong ADC does not seem to have any audible
effects, and the Windows driver does it, but it is nonetheless a good
idea to run the AK5385 with an oversampling ratio that is not outside
the documented limits.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 11:14:09 +01:00
Harvey Harrison e32740d978 ALSA: pcxhr.h replace signed one-bit bitfields
The usage and comments make it clear values of 1/0 were intended
rather than -1/0

Noticed by sparse:
sound/pci/pcxhr/pcxhr.h💯20: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-20 08:02:37 +01:00
Mark Brown ce3bdaa871 ASoC: Disable WM8731 line bypass by default
This avoids temporarily enabling the ouput stages during startup which
can cause audible effets in the output stages.

Reported-by: Fredrik Redgård <rik@svep.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-19 14:30:57 +00:00
Takashi Iwai e432472db2 Merge branch 'fix/usb-audio' into for-linus 2009-02-19 13:58:05 +01:00
Takashi Iwai e6845d9101 Merge branch 'fix/misc' into for-linus 2009-02-19 13:58:01 +01:00
Takashi Iwai 379752fdf8 Merge branch 'fix/hda' into for-linus 2009-02-19 13:57:52 +01:00
Clemens Ladisch 1275d6f608 sound: oxygen: automatically restore overwritten EEPROM
If the EEPROM was partially overwritten (which seems to happen before the OS is
booted), restore its entire contents by deducing it from the remaining
information.

This does not have any effect on the Linux driver, which works even with
incomplete information in the EEPROM, but it makes other drivers work again.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:22:26 +01:00
Clemens Ladisch 30459d7b18 sound: oxygen: handle cards with broken EEPROM
Under as yet unknown circumstances, the first word of the sound card's
EEPROM gets overwritten.  When this has happened, we cannot rely on the
subsystem IDs that the kernel reads from the PCI configuration
registers.  Instead, we read the IDs directly from the EEPROM and do the
ID matching manually.

Because the model-specific driver cannot determine the model before
calling oxygen_pci_probe(), that function now gets a get_model()
callback as parameter.  The customizing of the model structure, which
was formerly done by the probe() callback, also has moved into
get_model().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:22:25 +01:00
Clemens Ladisch a69bb3c3fe sound: oxygen: use static driver name
When allocating resources, use a fixed name instead of reading it from
the model structure.  This allows us to allocate the resources before
the actual model is known.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:22:24 +01:00
Clemens Ladisch 6ed9115709 sound: oxygen: allocate model_data dynamically
Allocate the model-specific data dynamically instead of including it in
the memory block of the card structure.  This will allow us to determine
the actual model after the card creation.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:22:23 +01:00
Clemens Ladisch bb71858853 sound: oxygen: make the owner module a parameter of the probe function
Move the owner field out of the oxygen_model structure and make it
a parameter of oxygen_pci_probe(), because the actual owner module does
not depend on the card model.  Furthermore, moving it out of the model
structure allows us to create the card structure before the actual model
is known.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:22:22 +01:00
Takashi Iwai a5e0e970c0 Merge branch 'topic/snd_card_new-err' into topic/oxygen 2009-02-19 10:22:14 +01:00
Clemens Ladisch 6ce6c473a7 sound: virtuoso: revert "do not overwrite EEPROM on Xonar D2/D2X"
This reverts commit 7e86c0e685 ("do not
overwrite EEPROM on Xonar D2/D2X") because it did not actually help with
the problem.

More user reports show that the overwriting of the EEPROM is not
triggered by using this driver but by installing Linux, and that the
installation of any other operating system (even one without any CMI8788
driver) has the same effect.  In other words, the presence of this
driver does not have any effect on the occurrence of the error.  (So
far, the available evidence seems to point to a BIOS bug.)

Furthermore, it turns out that the EEPROM chip is protected against
stray write commands by the command format and by requiring a separate
write-enable command, so the error scenario in the previous commit (that
SPI writes can be misinterpreted as an EEPROM write command) is not even
theoretically possible.

The mixer control that was removed as a consequence of the previous
commit can only be partially emulated in userspace, which also means it
cannot be seen be the in-kernel OSS API emulation, so it is better to
revert that change.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 10:15:39 +01:00
Takashi Iwai 7e0e44d430 ALSA: hda - Add digital-only mode for ALC268
ALC268 can be configured as digital-only, e.g. for HDMI, on some
machines.  Allow the parser to set up the digital-only mode.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 08:15:49 +01:00
Takashi Iwai ab9fec099b ALSA: hda - Avoid doubly beep attachment in patch_alc268()
Remove the doubly attachment in patch_alc268().
The input beep is attached conditionally only when needed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 08:13:26 +01:00
Takashi Iwai 07eba61dd6 ALSA: hda - Don't enable beep for digital-only ALC262
When ALC262 codec is configured as digital-only, it's meaningless to
add the digital beep input.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-19 08:06:35 +01:00
Mark Brown c6f2981170 ASoC: Add device init/exit annotations to new-style Wolfson CODEC drivers
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 21:26:58 +00:00
Mark Brown 519cf2df5f ASoC: Check for errors when writing WM8731 reset register
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 21:06:01 +00:00
Peter Ujfalusi 6bab83fd88 ASoC: TWL4030: Add digital loopback support
This patch adds the digital loopback/bypass support for twl4030 codec.

The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.

Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 17:14:23 +00:00
Takashi Iwai 2678f60d2b ALSA: jack - Use card->shortname for input name
Currently the jack layer refers to card->longname as a part of
its input device name string.  However, longname is often really long
and way too ugly as an identifier, such as,
"HDA Intel at 0xf8400000 irq 21".

This patch changes the code to use card->shortname instead.
The shortname string contains usually the h/w vendor and product
names but without messy I/O port or IRQ numbers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-18 16:46:27 +01:00
Mark Brown 93b760b707 ASoC: Implement SPI device unregistration for WM8731
Completely untested.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 14:47:20 +00:00
Mark Brown fc99675768 ASoC: Fix build for corgi and poodle
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 14:46:58 +00:00
Takashi Iwai b3bdb30b6d ALSA: hda - Add quirk for Acer X3200
Acer X3200 needs model=auto, otherwise model=acer is pre-selected.

Reference: Novell bnc#476268
	https://bugzilla.novell.com/show_bug.cgi?id=476268

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-18 13:16:26 +01:00
Mark Brown 59544d33ff ASoC: Remove version display from the WM8753 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:55:26 +00:00
Mark Brown 5998102b90 ASoC: Refactor WM8731 device registration
Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.

As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:55:22 +00:00
Mark Brown a8035c8f04 ASoC: Shuffle WM8731 SPI and I2C device registration
This is a pure code motion patch intended to improve reviewability of a
following patch moving WM8731 to use more standard device registration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:25:48 +00:00
Mark Brown 7ee7538041 ASoC: Rename AT91SAMG20-EK for applications
This is a bit more idiomatic and makes identifying a configuration
based on the board type work better.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:25:23 +00:00
Mark Brown 5de7f9b200 ASoC: Actively manage MCLK for AT91SAM9G20-EK
We have software control of the MCLK for the WM8731 so save a bit of
power by actively managing it within the machine driver, enabling it
only while the codec is active.

Once ASoC supports multiple boards and doesn't require the soc-audio
device the initial clock setup should be pushed down into the arch/arm
code but for now this reduces merge issues.

Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:25:23 +00:00
Mark Brown 40135ea071 ASoC: Check machine type before loading on AT91SAM9G20-EK
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:25:22 +00:00
Mark Brown d694354115 ASoC: Improve diagnostics for AT91SAM9G20-EK probe
We should display an error by default if we fail to register.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 11:25:21 +00:00
Roel Kluin c16159123d sound: OSS: missing parentheses in pas2_card.c
Add missing parentheses in pas2_card.c.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-18 11:37:51 +01:00
Mark Brown 22d22ee514 ASoC: Clean up WM8731 bias level configuration
The WM8731 bias level configuration function was written slightly
obscurely - streamline the code a little and refresh the comments.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-17 14:29:54 +00:00
Mark Brown 7b317b692a ASoC: Remove version display from the WM8731 driver
It makes boot a bit more noisy and I never remember to update it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-17 14:29:54 +00:00
Paul Fertser 31b59cf9ce ASoC: Fix WM8753 DAIs unregistering
WM8753 uses a tricky way to switch DAIs "on the fly", for that it
registers 2 dummy DAIs and substitutes them depending on mixer control.

List element of registered dummy DAIs should be preserved to allow
unregistering of DAIs on module unload.

Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-17 14:29:54 +00:00
Takashi Iwai cda9043d56 ALSA: cs4236 - Merge snd-cs4236-lib module into snd-cs4236
Since cs4232 and cs4236 drivers are merged, there is no reason to keep
snd-cs4236-lib module separately.  Let's merge it into the main driver
as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-17 08:10:54 +01:00
Takashi Iwai b22f5d94c4 sound: OSS: ad1848 - Fix another typo
Fix another typo of || and &&.

Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-17 08:02:16 +01:00
Takashi Iwai 8380740079 ALSA: au88x0 - Fix &&|| typo
Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.

Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-17 08:00:44 +01:00
Krzysztof Helt c2b73d1458 ALSA: cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection
cs4232 and cs4236 driver merge to solve PnP BIOS detection.

Also, the patch adds recognition if the chip is cs4236b+
or earlier part. This unifies drivers for both cs4232
and cs4236+ chips. It allows to use the PnP BIOS
detection for the cs4236+ chips. Previously, only
the snd-cs4232 could be detected by the PnP BIOS.

The cs4232+ cards reports two separate PnP BIOS ids.

The patch adds search for the second id to find out
resources assigned to a control port.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 23:05:25 +01:00
Takashi Iwai 96cf45cf55 Merge branch 'topic/snd_card_new-err' into topic/cs423x-merge 2009-02-16 23:03:57 +01:00
Joris van Rantwijk 3b03cc5b86 ALSA: usb-audio - Workaround for misdetected sample rate with CM6207
The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz
in its USB device descriptor. This patch extends an existing workaround
in usbaudio.c to also cover the CM6207.

This resolves issue 0004249 in the ALSA bug tracker.

Signed-off-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 22:58:23 +01:00
Takashi Iwai 0412558c87 ALSA: usb-audio - Fix non-continuous rate detection
The detection of non-continuous rates (given via rate tables) isn't
processed properly (e.g. for type II).

This patch fixes and simplifies the detection code.

Tested-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 22:48:12 +01:00
Clemens Ladisch e156ac4c57 sound: usb-audio: fix uninitialized variable with M-Audio MIDI interfaces
Fix the snd_usbmidi_create_endpoints_midiman() function, which forgot to
set the out_interval member of the endpoint info structure for Midiman/
M-Audio devices.  Since kernel 2.6.24, any non-zero value makes the
driver use interrupt transfers instead of bulk transfers.  With EHCI
controllers, these random interval values result in unbearably large
latencies for output MIDI transfers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: David <devurandom@foobox.com>
Tested-by: David <devurandom@foobox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 15:31:48 +01:00
Takashi Iwai c23127566c ALSA: hda - Clean up quirks for HP laptops with AD1984A
Clean up quirks for HP laptops with AD1984A using SND_PCI_QUIRK_MASK()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 15:20:41 +01:00
Takashi Iwai 2ae466f8cc ALSA: hda - Cleanup IDT92HD7x HP quirks
Clean up IDT92HD7x quirks for HP laptops with SND_PCI_QUIRK_MASK().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 14:16:36 +01:00
Roel Kluin a259cb8eb7 sound: OSS: &&/|| typo in ad1848.c
&&/|| typo

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 10:25:11 +01:00
Takashi Iwai e23573d7e3 Merge branch 'fix/hda' into topic/hda 2009-02-16 10:23:35 +01:00
Herton Ronaldo Krzesinski e2ea57a8df ALSA: hda - Fix speaker output on HP DV4 1155-SE
Force speaker pin config with model=hp-dv5 model for cases when bios
doesn't set it up properly. All reported hp laptops using model=hp-dv5
model have speaker at pin 0x0d with same config, so it's safe to add
this within hp-dv5 model.

Reference: alsa-devel mailing list thread on
    http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 10:23:00 +01:00
Takashi Iwai d14a7e0bfc Revert "Sound: hda - Restore PCI configuration space with interrupts off"
This reverts commit 32e176c14d.

That commit caused a regression with suspend on Thinkpad SL300.

Reference: kernel bug#12711
	http://bugzilla.kernel.org/show_bug.cgi?id=12711

Tested-by:  Alexandre Rostovtsev <tetromino@gmail.com>
Acked-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-16 10:16:27 +01:00
Ingo Molnar b69bc39674 Merge commit 'v2.6.29-rc5' into x86/apic 2009-02-15 09:00:18 +01:00
Kevin Hilman bf3dbe5c8c ASoC: Fix DaVinci module unload error
Fix for the error when the audio module is unloaded.  On unregistering
the platform_device, platform_device_release will free the platform
data.If platform data is static the kernel panics when it is freed.
Instead use the platform device helper function to add data.

This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-13 20:21:30 +00:00
Linus Torvalds b51ebdc40c 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:
  ASoC: Only register AC97 bus if it's not done already
  ALSA: hda - Add snd_hda_multi_out_dig_cleanup()
  ALSA: hda - Add missing terminator in slave dig-out array
  ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model
  ALSA: hda - Register (new) devices at reconfig
  ALSA: mtpav - Fix initial value for input hwport
  ALSA: hda - add id for Intel IbexPeak integrated HDMI codec
  ALSA: hda - compute checksum in HDMI audio infoframe
  ALSA: hda - enable HDMI audio pin out at module loading time
  ALSA: hda - allow multi-channel HDMI audio playback when ELD is not present
  ASoC: Update SDP3430 machine driver for snd_soc_card
  ALSA: hda - Add quirk for Asus z37e (1043:8284)
  sound: Remove OSSlib stuff from linux/soundcard.h
  ASoC: WM8990: Fix kcontrol's private value use in put callback
  ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
2009-02-13 08:19:11 -08:00
Takashi Iwai f1464ede55 Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-02-13 15:10:35 +01:00
Takashi Iwai 99cbb86180 Merge branch 'fix/asoc' into for-linus 2009-02-13 15:06:04 +01:00
Takashi Iwai 7c56c29a3b Merge branch 'fix/hda' into for-linus 2009-02-13 15:05:59 +01:00
Mark Brown c85e5a4161 Merge branch 'for-2.6.29' into for-2.6.30 2009-02-13 14:02:08 +00:00
Mark Brown 14fa43f53f ASoC: Only register AC97 bus if it's not done already
ASoC supports both explicit codec drivers for AC97 devices and a simple
driver which uses the standard ALSA AC97 framework for codec support.
When used with the generic AC97 codec support that will provide the
ad hoc AC97 device for drivers like touchscreens to attach to so the
core shouldn't do so.

Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-13 13:50:22 +00:00
Timur Tabi d5e9ba1d58 ASoC: add additional controls to the CS4270 codec driver
Update the CS4270 codec driver to allow applications to use the mixer to
control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute.
Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first
initializes the hardware, but these features either don't work or interfere
with normal ALSA behavior.  However, they can now be re-enabled by an
application if desired.

Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute
bits.  The driver previously and erroneously assumed that these bits
control only external muting circuitry, but they also control internal
muting circuitry, so they should always be used.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-13 13:32:51 +00:00
Takashi Iwai 6a05ac4afa ALSA: hda - Support multiple digital outs with auto-probing
Added the support of multiple digital outputs via auto-probing for
Realtek ALC88x codecs.  The multiple outputs are handled as slave
streams, so only one PCM stream (and the corresponding IEC958*
elements) is provided to control both digital outputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 11:59:33 +01:00
Takashi Iwai 9b5f12e5a4 ALSA: hda - Add proper cleanup for multiout-dig for ALC codecs
The recent patch_realtek.c contains the slave digital-out support
as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 11:59:31 +01:00
Takashi Iwai c8a1a8985d Merge branch 'fix/hda' into topic/hda 2009-02-13 11:59:26 +01:00
Takashi Iwai 9411e21cd0 ALSA: hda - Add snd_hda_multi_out_dig_cleanup()
Added the helper function snd_hda_multi_out_dig_cleanup() to clean up
the digital outputs with multi setup.  This call is needed in cases
the codec supports multiple digital outputs as slaves.  Otherwise the
slave widgets aren't properly cleaned up.

For a single digital output (e.g. in patch_conexant.c), this call isn't
needed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 11:59:13 +01:00
Takashi Iwai 3a08e30de2 ALSA: hda - Add missing terminator in slave dig-out array
Added the missing terminator for ad1989b_slave_dig_outs[].

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 11:53:27 +01:00
Ingo Molnar f8a6b2b9ce Merge branch 'linus' into x86/apic
Conflicts:
	arch/x86/kernel/acpi/boot.c
	arch/x86/mm/fault.c
2009-02-13 09:44:22 +01:00
Takashi Iwai 946835074e ALSA: hda - Add quirk for Acer AX1700-U3700A
Force model=auto for Acer AX1700-U3700A with ALC888 codec.
Since Acer devices are handlded as model=acer as default, the auto
parsing has to be specified explicitly.
(Maybe it's better rather to remove this default model=acer handling,
 though.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 09:31:20 +01:00
Takashi Iwai 344384494e Merge branch 'fix/hda' into topic/hda 2009-02-13 08:41:44 +01:00
Herton Ronaldo Krzesinski 92258a3ed2 ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model
Change HP dv7 quirk: although reported to work with hp-m4 model
(https://bugzilla.novell.com/show_bug.cgi?id=445321), the original
report doesn't contain info about testing of internal microphone.

Recently I received a report about internal mic not working
(https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be
related with the forced line in on pin 0x0e done with hp-m4 model. Thus
change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported
to be fixed on a provided kernel with this change
(https://qa.mandriva.com/show_bug.cgi?id=44855#c196).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 08:41:00 +01:00
Aristeu Sergio Rozanski Filho 27e089888f ALSA: hda: add quirk for Lenovo X200 laptop dock
Currently the HP connector on X200 dock doesn't detect when a HP is connected
nor allows sound to be played using it. This patch fixes the problem by adding
a quirk for this specific model. It's possible that others have the same NID
(0x19) to report when dock HP is connected, but I don't have access to any.
Please Cc me in the reply since I'm not subscribed to alsa-devel@.

Signed-off-by: Aristeu Rozanski <aris@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 08:40:20 +01:00
Matthew Ranostay 8bb0ac5573 ALSA: hda: Add STAC_DELL_S14 quirk
Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins
in pin_nids for stac92hd83xxx. Also reorganized connection selection
code for the respective ports per quirk define.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 08:35:08 +01:00
Takashi Iwai 20db7cb0ac ALSA: hda - Add forced codec-slots for ASUS W5F
ASUS W5F needs the fixed codec-slots to probe to override the BIOS
problem.

Tested-by: Giovanni Moser Frainer <giovanni@redix.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 08:19:30 +01:00
Takashi Iwai f1eaaeec11 ALSA: hda - Allow fixed codec-probe mask
Some devices have broken BIOS and they don't set the codec probe-bit
properly after cleared by the driver.  This makes the driver skipping
the necessary codec slots.

Since BIOS update isn't always easy, now the semantics of probe_mask
option is changed a bit.  When it contains the bit 8 (0x100), the
lower bits are used to probe that slots regardless of codec-probe bits
returned by the hardware.

For example, probe_mask=0x103 will force to probe the codec slot #0
and #1.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-13 08:16:55 +01:00
Harvey Harrison e930e99500 ALSA: echoaudio - replace uses of __constant_{endian}
The base versions handle constant folding now.

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-12 00:24:11 +01:00
Herton Ronaldo Krzesinski c98041f7d7 ALSA: hda - Cleanup setting of pin_configs in patch_stac927x
After commit "ALSA: hda - Fix restore of pin configs at resume for
STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks
already for pins == NULL case, saving then default pin configs from
machine with stac92xx_save_bios_config_regs. So we can remove the
extra checks when stac927x_brd_tbl[spec->board_config] == NULL.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-12 00:17:31 +01:00
Takashi Iwai a1152e3570 Merge branch 'fix/hda' into topic/hda 2009-02-12 00:14:34 +01:00
Takashi Iwai 26a74f1f61 ALSA: hda - Register (new) devices at reconfig
The devices that have been newly added during reconfig must be
registered.  Otherwise they won't be visible to user-space.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-12 00:13:19 +01:00
Takashi Iwai 32cf9a16f4 ALSA: mtpav - Fix initial value for input hwport
Fix the initial value for input hwport.  The old value (-1) may cause
Oops when an realtime MIDI byte is received before the input port is
explicitly given.
Instead, now it's set to the broadcasting as default.

Tested-by: Holger Dehnhardt <dehnhardt@ahdehnhardt.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-12 00:06:42 +01:00
Takashi Iwai 0852d7a654 ALSA: hda - Detect multiple digital-out pins
Detect multiple digital-out pins in snd_hda_parse_pin_defconfig().
The dig_out_pin and dig_out_type fields become arrays.

The codec parser still doesn't use this multiple pins detection, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-12 00:04:19 +01:00
Roel Kluin 1afa6e2e1d sound: OSS: dmabuf: too many loops
loop adev->dmap_out->nbufs times

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 14:12:04 +01:00
Takashi Iwai 32d2c7fa13 ALSA: hda - Fix a wrong pin check in snd_hda_parse_pin_def_config()
Fixed a wrong pin check (a typo) for debug print of digital input pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 11:33:13 +01:00
Takashi Iwai df22313637 Merge branch 'fix/hda' into topic/hda 2009-02-11 09:09:29 +01:00
Wu Fengguang a57c0eb655 ALSA: hda - add id for Intel IbexPeak integrated HDMI codec
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 09:09:15 +01:00
Wu Fengguang 9a957a24e3 ALSA: hda - compute checksum in HDMI audio infoframe
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 09:09:07 +01:00
Wu Fengguang 606c0cee69 ALSA: hda - enable HDMI audio pin out at module loading time
We found that enabling/disabling HDMI audio pin out at stream start/stop
time will kill the leading 500ms or so sound samples. Avoid this by enabling
pin out once and for ever at module loading time.

The leading ~500ms audio samples will still be lost when switching from
X-channel playback to Y-channel playback where X != Y. However there's no
much we can do about it: the audio infoframe has to change and it looks like
either G45 or YAMAHA requires some time to switch the configuration.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 09:08:59 +01:00
Wu Fengguang a1667e4eea ALSA: hda - allow multi-channel HDMI audio playback when ELD is not present
The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel
mapping to play >2 channel HDMI audio. In theory that mapping should be
derived from its speaker configurations contained in its ELD. However we
currently cannot get ELD in console before the KMS functionalities are ready.
This is a more or less general issue at least in the near future. As a
workaround, we propose to allow playback of mult-channel audio when ELD
is not available.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 09:08:47 +01:00
Takashi Iwai 9e30d7718b ASoC: Fix forgotten replacements of socdev->codec
The snd_soc_codec was moved into socdev->card, but this change wasn't
applied in some places.  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-11 08:28:04 +01:00
Mark Brown 9e32ebdb3a Merge branch 'for-2.6.29' into for-2.6.30 2009-02-10 21:37:01 +00:00
Lopez Cruz, Misael 272edb0049 ASoC: Update SDP3430 machine driver for snd_soc_card
This patch replaces "snd_soc_machine" structure by "snd_soc_card" in
SP3430 driver. This change is needed in SDP3430 driver to reflect
changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch
(875065491f).

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-10 21:36:27 +00:00
Takashi Iwai 501ca25629 Merge branch 'fix/hda' into topic/hda 2009-02-10 17:17:17 +01:00
Mackenzie Morgan 44a678d04b ALSA: hda - Add quirk for Asus z37e (1043:8284)
Added a quirk for Asus Z37E for fixing suspend/hibernation problem.

Reference:
	https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/25896
	http://launchpadlibrarian.net/17053575/0001-Add-quirk-for-ASUS-Z37E-to-make-sound-audible-afte.patch
	https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=4282

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-10 17:13:43 +01:00
Takashi Iwai 22971e3a77 ALSA: hda - add digital beep support for ALC268
Added the digital beep support for ALC268.  It was missing in the
last patches.

However, ALC268 has a strange pin use for widget 0x1d, which could be
used as another purpose.  So, the patch adds a check of the beep control
before creating the hook for input layer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-10 11:56:44 +01:00
Roel Kluin f6f35bbe7c [ARM] AACI: timeout will reach -1
With a postfix decrement the timeout will reach -1 rather than 0,
so the warning will not be issued.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
2009-02-10 09:59:20 +00:00
Jarkko Nikula 7565fc38cc ASoC: TLV320AIC3X: Add TLV information for volume controls
TLV320AIC3X volume controls are logarithmic. Export their dB ranges.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-09 18:27:59 +00:00
Jarkko Nikula b93f74f604 ASoC: TLV320AIC3X: Fix volume ranges
This is a minor fix but helps to define dB ranges for volume controls.

Only DAC digital volume has full register value range from 0 to 127 but
ADC PGA gain and output stage volume controls don't.

For ADC PGA, maximum value is 119 and then it saturates to the same
gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB
and is muted for values 118 and above.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-09 18:27:59 +00:00
Takashi Iwai a85165c66c ALSA: via82xx - Clean up quirk list
Use SND_PCI_QUIRK_VENDOR() macro.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-02-09 17:20:19 +01:00