Commit Graph

5564 Commits

Author SHA1 Message Date
Takashi Iwai
f03ecf5053 ALSA: hda - Fix the previous tagra-8ch patch
- Fix a typo in the patch
- Adapted to follow the recent change for unsol event handling

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 16:38:17 +02:00
David Heidelberger
64a8be7435 ALSA: hda - Add 7.1 support for MSI GX620
Added 7.1 support for MSI GX620 and jack quirk.

Reference: kernel bug#13430
	http://bugzilla.kernel.org/show_bug.cgi?id=13430

Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 16:18:31 +02:00
Takashi Iwai
c00701101b ALSA: pcm - A helper function to compose PCM stream name for debug prints
Use a common helper function for the PCM stream name displayed in
XRUN and buffer-pointer debug prints.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 16:01:47 +02:00
Takashi Iwai
806d31d738 ALSA: emu10k1 - Fix minimum periods for efx playback
EFX playback stream should have periods_min = 2 to avoid the buffer
position overflow (due to restrictions of the pcm-indirect helper).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 15:50:42 +02:00
Takashi Iwai
d362af62ed ALSA: ctxfi - Fix / clean up hw20k2 chip code
- Clean up Hungarian coding style
- Don't use static variables for I2C information; this unables to use
  multiple instances.  Now they are stored in struct hw20k2 fields.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 15:31:22 +02:00
Takashi Iwai
af8500bbbd ALSA: ctxfi - Fix possible buffer pointer overrun
Fix possible buffer pointer overruns.  Back to zero when it's equal
or over the buffer size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 15:07:46 +02:00
Takashi Iwai
514eef9c2a ALSA: ctxfi - Remove useless initializations and cast
Remove useless variable initializations and cast at the beginning of
functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 14:57:57 +02:00
Takashi Iwai
4836ac6554 ALSA: ctxfi - Fix DMA mask for emu20k2 chip
Allow 64bit DMA mask for emu20k2 chip, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 14:51:24 +02:00
Takashi Iwai
d436dd063b ALSA: ctxfi - Make volume controls more intuitive
Change the volume control to dB scale (as the raw data seems so).
Also added the TLV dB-scale information.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 13:50:18 +02:00
Takashi Iwai
c4865679df ALSA: ca0106 - Fix master volume scale
The master volume dB scale was wrongly defined as 0.50dB setp while
it must be 0.25dB step.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 12:57:17 +02:00
Takashi Iwai
54de6bc8b2 ALSA: ctxfi - Optimize the native timer handling using wc counter
Optimize the timer update routine to look up wall clock once instead of
checking the position of each stream at each timer update.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08 12:38:54 +02:00
Takashi Iwai
ab1863fc9b ALSA: pcm - Fix update of runtime->hw_ptr_interrupt
The commit 13f040f9e5 made another
regression, the missing update of runtime->hw_ptr_interrupt.
Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt(),
not in snd_pcm_update_hw_ptr(), it must be updated before the hw_ptr
change check.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-07 12:19:33 +02:00
Figo.zhang
ad0b0822f9 ALSA: sgio2audio.c: clean up checking
vfree() does it's own 'NULL' check,so no need for check before
calling it.

Signed-off-by: Figo.zhang <figo1802@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-07 09:08:43 +02:00
Jaroslav Kysela
d86bf92313 ALSA: pcm - Fix a typo in hw_ptr update check
Fix a typo in the commit 13f040f9e5
  ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
which causes obvious problems with PA.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-06 18:32:06 +02:00
Mark Brown
74b8f955a7 ASoC: Apostrophe patrol
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-06 11:26:15 +01:00
Troy Kisky
ccff4b15e0 ASoC: codec tlv320aic23 fix bogus divide by 0 message
Some code analyzer software mistakenly gives
divide by 0 error messages for these lines.
This patch will end its confusion.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-06 09:24:48 +01:00
Takashi Iwai
28cd4aa43d ALSA: ctxfi - Add missing inclusion of linux/math64.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 18:07:12 +02:00
Takashi Iwai
3f7440a6b7 ALSA: Clean up 64bit division functions
Replace the house-made div64_32() with the standard div_u64*() functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 17:45:17 +02:00
Takashi Iwai
032abb519c ALSA: ctxfi - Set device 0 for mixer control elements
Mixer control elements are usually assigned to device 0.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 16:44:25 +02:00
Takashi Iwai
2a36f67f8c ALSA: ctxfi - Clean up / optimize
- Use static tables instead of assigining each funciton pointer
- Add __devinit* to appropriate places; pcm, mixer and timer cannot be
  marked because they are kept in the function table that lives long
- Move create_alsa_devs function out of struct ct_atc to mark it
  __devinit

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 16:44:24 +02:00
Takashi Iwai
775ffa1d3e ALSA: ctxfi - Set periods_min to 2
Set 2 to minimal periods of playback pcm setups, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 16:44:22 +02:00
Takashi Iwai
b7bbf87608 ALSA: ctxfi - Use native timer interrupt on emu20k1
emu20k1 has a native timer interrupt based on the audio clock, which
is more accurate than the system timer (from the synchronization POV).
This patch adds the code to handle this with multiple streams.

The system timer is still used on emu20k2, and can be used also for
emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 16:44:13 +02:00
Takashi Iwai
6bc5874a1d ALSA: ctxfi - Fix previous fix for 64bit DMA
Remove unneeded substitution to 32bit int to make it really working.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 12:18:37 +02:00
Guido Günther
3e1647c5b5 ALSA: support Sony Vaio TT
with BIOS probing only we offer a non functional headphone swith and
volume slider.

Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 12:12:26 +02:00
Takashi Iwai
42a0b31827 ALSA: ctxfi - Fix endian-dependent codes
The UAA-mode check in hwct20k1.c is implemented with the endian-dependent
codes.  Fix to be more portable (and readable).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 09:29:22 +02:00
Takashi Iwai
6d74b86d3c ALSA: ctxfi - Allow 64bit DMA
emu20kx chips support 64bit address PTE.  Allow the DMA bit mask to
accept 64bit address, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 09:26:41 +02:00
Marek Vasut
37330efd4a [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE2
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
2009-06-05 10:41:54 +08:00
Daniel Mack
e3509ff0fb ASoC: fix NULL pointer dereference in soc_suspend()
In case the initalization of an soc_device failed, there is no codec
associated with it. soc_suspend() will still dereference the pointer
and cause an Ooops when entering the sleep mode.

This happens on our board with a multi-target kernel image when booted
on a machine without audio circuits.

This patch makes the code bail out very early in this special case.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-04 13:24:08 +01:00
Alexander Beregalov
65f7598311 ALSA: hda_intel: fix build error when !PM
Fix this build error when CONFIG_PM is not set:
ound/pci/hda/hda_intel.c: In function 'azx_bus_reset':
sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all'
sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend'
sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume'

Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 14:21:11 +02:00
Jean Delvare
82ced6fd28 ALSA: Add missing __devexit_p() markers
3 ISA sound drivers lack their __devexit_p() markers, which would
cause build failures when the kernel is built without hotplug support.

Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 10:52:16 +02:00
Takashi Iwai
d08664fdb5 ASoC: Fix build error in twl4030.c
Fix the (likely cut-n-paste) error by commit
16a30fbb0d, which causes the error below:
  sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache':
  sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 10:01:11 +02:00
Jaroslav Kysela
5fdc18d938 ALSA: Core - clean up snd_card_set_id* calls and remove possible id collision
Move locking outside snd_card_set_id_internal() function and rename it
to snd_card_set_id_no_lock() for better function description.

User defined id is just copied to card structure at allocation time.
The real unique id procedure is called in snd_card_register() to
ensure real atomicity.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 01:22:07 +02:00
Hector Martin
018df41861 ALSA: hda - More Aspire 8930G fixes
Enable all three capture channels, including the missing nid 7 which is
the only one capable of capturing DMIC input

Enable Headphone amp for the HP jack. This causes a volume boost for
headphones, but does not cause any noticeable effect for light loads
like other amps, so there is no need to make it configurable.

Add Input Mix capture mux setting to capture the output of the playback
input mux (that is, what goes out the speakers except for PCM)

Hack another coef register because the stereo DMIC for some reason
produces a nonstandard sum/difference signal. I found a bit to make it
just use the sum signal for both channels, which makes it behave like a
standard mono microphone. The stereo is useless anyway (they're 1cm apart).

Tested working: Three capture channels, mic in, line in, DMIC.

Tested not working: CD. Not sure why, might be unconnected in the actual
hardware or a CD drive issue.

Also looked at SPDIF. It appears to work (emitter lights up inside the
HP out jack) but I lack a proper miniTOSLINK cable to test it.

Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 00:13:40 +02:00
Roel Kluin
13be1bf146 ALSA: burgundy: timeout message is off by one.
Timeout message is off by one.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 00:10:41 +02:00
Mark Brown
872c78202c ALSA: Fix double locking of card list in snd_card_register()
The introduction of snd_card_set_id() added a lock on the card list
to the old choose_default_id() function when using it to implement
the new API call. This lock is needed to allow us to walk the list
and check to see if our new name is a duplicate. Unfortunately this
causes a lockup when called from snd_card_register() (in cases
where no ID is supplied for the card) since the card list is already
locked there.

Fix this fairly hideously by factoring out the implementation and
using a flag to indicate if the lock should be held. A better fix
would probably be to refactor snd_card_register() to move the
_set_id() outside the locking region but I can't immediately see
anything I can convince myself is safe.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03 23:33:28 +02:00
Cliff Cai
f692fce0cf ASoC: SSM2602: assign last substream to the master when shutting down
Fixes crash when shutting down.

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03 11:38:23 +01:00
Sonic Zhang
cf485da15a ASoC: Blackfin: document how anomaly 05000250 is handled
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03 11:31:42 +01:00
Cliff Cai
80d5bd9314 ASoC: Blackfin: set the transfer size according the ac97_frame size
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03 11:30:01 +01:00
Cliff Cai
2552a710f4 ASoC: SSM2602: remove unsupported sample rates
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03 11:25:51 +01:00
Takashi Iwai
3e1e0a5dd5 ALSA: powermac - Replace the rest of __init*
All __initdata should be __devinitdata as platform device is hotpluggable.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03 08:13:15 +02:00
Stephen Rothwell
5c9b6e9e61 ALSA: sound/ppc: update annotations of serveral functions
[I am not sure if this is the correct approach as I don't know if any of
this actual hardware or drivers are really hot pluggable.]

Gets rid of these build warnings:

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_new().
If .snd_pmac_new is only used by .snd_pmac_probe then
annotate .snd_pmac_new with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_burgundy_init().
If .snd_pmac_burgundy_init is only used by .snd_pmac_probe then
annotate .snd_pmac_burgundy_init with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_daca_init().
If .snd_pmac_daca_init is only used by .snd_pmac_probe then
annotate .snd_pmac_daca_init with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_init().
If .snd_pmac_tumbler_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_init with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_post_init().
If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_post_init with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_awacs_init().
If .snd_pmac_awacs_init is only used by .snd_pmac_probe then
annotate .snd_pmac_awacs_init with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_pcm_new().
If .snd_pmac_pcm_new is only used by .snd_pmac_probe then
annotate .snd_pmac_pcm_new with a matching annotation.

WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_attach_beep().
If .snd_pmac_attach_beep is only used by .snd_pmac_probe then
annotate .snd_pmac_attach_beep with a matching annotation.

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03 08:08:44 +02:00
Andrea Borgia
ca85b6ba59 ALSA: usb-audio - errata corrige for quirk
Cut'n'paste mistake, whose likely result was nothing at all.
Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC".

Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03 08:05:32 +02:00
Takashi Iwai
3f08a0e4ab ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'99
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 17:39:52 +02:00
Takashi Iwai
eeaf100d25 ALSA: ca0106 - Add missing card->mixername field setup
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 16:06:26 +02:00
Takashi Iwai
bd05dbd3b2 Merge branch 'topic/ctxfi-fix' into topic/ctxfi 2009-06-02 15:55:22 +02:00
Takashi Iwai
c76157d928 ALSA: ctxfi - Support SG-buffers
Use SG-buffers instead of contiguous pages.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:47 +02:00
Takashi Iwai
cd391e206f ALSA: ctxfi - Remove PAGE_SIZE limitation
Remove the limitation of PAGE_SIZE to be 4k by defining the own
page size and macros for 4k.  8kb page size could be natively supported,
but it's disabled right now for simplicity.

Also, clean up using upper_32_bits() macro.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:46 +02:00
Takashi Iwai
d2b9b96c51 ALSA: ctxfi - Fix supported PCM formats
The device seems supporting only U8, S16, S24_3LE, S32.  Other linear
formats result in bad outputs.

Also, added the support for 32bit float format, which wasn't listed
in the original code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:46 +02:00
Takashi Iwai
8372d4980f ALSA: ctxfi - Fix PCM device naming
PCM names for surround streams should be also fixed as well as the mixer
element names.  Also, a bit clean up for PCM name setup.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:46 +02:00
Takashi Iwai
6585db943a ALSA: ctxfi - Fix surround mixer names
We usually pick up "Surround" mixer for the rear output, and "Side"
for the extra surround.  Fix the channel mapping to follow it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:46 +02:00
Takashi Iwai
822fa19b5c ALSA: ALSA: ctxfi - Release PCM resources at each prepare call
The prepare callback can be called multiple times, thus it needs to
release and acquire the resource again by itself at the second or later
call.

Simply add pcm_release_resources() at the beginning of each prepare
callback in ctatc.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 15:54:46 +02:00
Takashi Iwai
9a83b7453c ALSA: Remove invalid GENERIC_MIX PCM sublass
SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playback
devices, but ca0106 and emu10k1x don't support it (unlike emu10k1).
We shouldn't set that flag to avoid confusion.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 14:23:05 +02:00
Daniel Mack
c6e24d4db8 ALSA: snd_usb_caiaq: bump version number
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 14:03:58 +02:00
Daniel Mack
bafeee5b1f ALSA: snd_usb_caiaq: give better shortname
If not passed as module option, provide an own card ID with the newly
introduced snd_set_card_id() call.

This will prevent ALSA from calling choose_default_name() which only
takes the last part of a name containing whitespaces. This for example
caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very
descriptive.

The implementation now takes the short name and removes all whitespaces
from it which is much nicer.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 12:55:59 +02:00
Takashi Iwai
17db0486d7 Merge branch 'topic/core-id-check' into topic/caiaq 2009-06-02 12:55:40 +02:00
Jaroslav Kysela
10a8ebbb08 ALSA: Core - add snd_card_set_id() function
Introduce snd_card_set_id() function to allow lowlevel drivers to set
default identification name for card slot. The function checks also
for identification name collisions and tries to create unique name.

Also, the snd_card_create() function is simplified, because this new
function is used. As bonus, proper name collision checks are evaluated
at the card create time.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 12:47:46 +02:00
Takashi Iwai
3c4dbda003 Merge branch 'topic/hda-ctl-reset' into topic/hda 2009-06-02 12:15:48 +02:00
Takashi Iwai
601e1cc5df ALSA: ca0106 - Add missing registrations of vmaster controls
Although the vmaster controls are created, they aren't registered thus
they don't appear in the real world.  Added the missing snd_ctl_add()
calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-06-02 11:37:01 +02:00
Hector Martin
3b315d70b0 ALSA: hda - Acer Aspire 8930G support
Short story: this laptop has 5.1 built-in speakers which you *really*
want to use (the not-so-"sub" woofer is what makes the audio above
average for a laptop), so 6-channel support is important (plus a decent
asound.conf to upmix stereo). It also has the 3 typical jacks that ought
to have a selectable mode. And it's based on ALC889, which sucks.

Rationale/explanations:

The const_channel_count stuff was added because, for a laptop like this,
you always have 6 channels available (internal speakers) but still need
to set the mode for the 3 external jacks. Therefore, the device always
needs to be in 6-channel mode but there still needs to be a mixer
control for the jack mode. You could use line/mic-in at the same time as
the 6 internal speakers, for example. You might be tempted to make it
even smarter by dynamically switching the max channel count when
headphones are plugged in (therefore muting the internal speakers and
reducing the physical channel count to the jack channel mode), but as a
user I consider this to be harmful because I want the audio to blow up
to 6 channels / upmixed as soon as I unplug the headphones, and having
opened the device while in 2-channel mode would prevent this from
working (and always making 6-channel mode available doesn't do any harm).

The hardware needs EAPD turned on and the DACs routed to the internal
speaker pins, so the patch adds those verbs.

The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work
by default, at least here. I wasted much time trying to talk to
Realtek/pshou about this, but they just kept sending me useless updates
to patch_realtek.c that did nothing relevant. In the end I gave up and
brute forced the issue by trying to flip every bit in the proprietary
coefficient registers, and eventually found the two magic registers that
need to be cleared to enable all DACs. I have only heard Acer users
complain, but that might be because ALC889 is pretty new and using 5.1
(and noticing the missing center/lfe channels) might not be that common.
If this is a generalized issue with all ALC889 systems then those verbs
should probably be moved to a common verb array.

The internal mic is untested and probably doesn't work.

These settings will probably work for other Acer Gemstone laptops with
the same 5.1 speaker config. When identified, those should be added to
the PCI subsystem ID list.

Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 10:58:37 +02:00
Daniel Mack
1a1df6f043 ALSA: snd_usb_caiaq: give better longname
The serial number is of no interest in the longname, remove it. This
gives space for the usb path information which is more informative.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 09:47:33 +02:00
Daniel Mack
d3873a1be9 ALSA: snd_usb_caiaq: use strlcpy
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 09:41:16 +02:00
Daniel Mack
9318dce503 ALSA: snd_usb_caiaq: clean whitespaces
Cosmetic changes only, no code change.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 09:39:29 +02:00
Takashi Iwai
67fbf88063 ALSA: ctxfi - Fix a typo in MODULE_LICENSE
A space has to be put between GPL and v2.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 09:18:26 +02:00
Takashi Iwai
8a4259bf89 ALSA: ctxfi - Fix Oops at mmapping
Replace a spinlock with a mutex protecting the vm block list at
mmap / munmap calls, which caused Oops like below:

BUG: sleeping function called from invalid context at mm/slub.c:1599
in_atomic(): 0, irqs_disabled(): 1, pid: 32065, name: xine
Pid: 32065, comm: xine Tainted: P           2.6.29.4-75.fc10.x86_64 #1
Call Trace:
  [<ffffffff81040685>] __might_sleep+0x105/0x10a
  [<ffffffff810c9fae>] kmem_cache_alloc+0x32/0xe2
  [<ffffffffa08e3110>] ct_vm_map+0xfa/0x19e [snd_ctxfi]
  [<ffffffffa08e1a07>] ct_map_audio_buffer+0x4c/0x76 [snd_ctxfi]
  [<ffffffffa08e2aa5>] atc_pcm_playback_prepare+0x1d7/0x2a8 [snd_ctxfi]
  [<ffffffff8105ef3f>] ? up_read+0x9/0xb
  [<ffffffff81186b61>] ? __up_read+0x7c/0x87
  [<ffffffffa08e36a6>] ct_pcm_playback_prepare+0x39/0x60 [snd_ctxfi]
  [<ffffffffa0886bcb>] snd_pcm_do_prepare+0x16/0x28 [snd_pcm]
  [<ffffffffa08867c7>] snd_pcm_action_single+0x2d/0x5b [snd_pcm]
  [<ffffffffa08881f3>] snd_pcm_action_nonatomic+0x52/0x6a [snd_pcm]
  [<ffffffffa088a723>] snd_pcm_common_ioctl1+0x404/0xc79 [snd_pcm]
  [<ffffffff810c52c8>] ? alloc_pages_current+0xb9/0xc2
  [<ffffffff810c9402>] ? new_slab+0x1a5/0x1cb
  [<ffffffff810ab9ea>] ? vma_prio_tree_insert+0x23/0xc1
  [<ffffffffa088b411>] snd_pcm_playback_ioctl1+0x213/0x230 [snd_pcm]
  [<ffffffff810b6c20>] ? mmap_region+0x397/0x4c9
  [<ffffffffa088bd9b>] snd_pcm_playback_ioctl+0x2e/0x36 [snd_pcm]
  [<ffffffff810ddc64>] vfs_ioctl+0x2a/0x78
  [<ffffffff810de130>] do_vfs_ioctl+0x462/0x4a2
  [<ffffffff81029cef>] ? default_spin_lock_flags+0x9/0xe
  [<ffffffff81374647>] ? trace_hardirqs_off_thunk+0x3a/0x6c
  [<ffffffff810de1c5>] sys_ioctl+0x55/0x77
  [<ffffffff8101133a>] system_call_fastpath+0x16/0x1b

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:40:51 +02:00
Takashi Iwai
b20f3b8346 ALSA: hda - Limit codec-verb retry to limited hardwares
The reset of a BUS controller during operations is somehow risky and
shouldn't be done inevitably for devices that have apparently no such
codec-communication problems.

This patch adds the check of the hardware and limits the bus-reset
capability.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 01:21:30 +02:00
Takashi Iwai
8dd783304e ALSA: hda - Add codec bus reset and verb-retry at critical errors
Some machines machine cause a severe CORB/RIRB stall in certain
weird conditions, such as PA access at the start up together with
fglrx driver.  This seems unable to be recovered without the controller
reset.

This patch allows the bus controller reset at critical errors so
that the communication gets recovered again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 01:21:23 +02:00
Takashi Iwai
8871e5b915 ALSA: hda - Reorder and clean-up ALC268 quirk table
Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some
entries using SND_PCI_QUIRK_MASK().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 01:02:50 +02:00
Ozan Çağlayan
d22142aa1b ALSA: hda - fix audio on LG R510
Currently, LG R510 is only able to produce sound on headphones, the
internal speakers are not working.

The user tested and confirmed that with model=Dell headphones,
internal speakers and the microphone are working flawlessly.

Tested-by: Serdar Soytetir <tulliana@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 00:58:14 +02:00
Kacper Szczesniak
92b9de8342 ALSA: hda - Macbook[Pro] 5 6ch support
this is a patch against current snapshot that adds:
6 channels support for the MB5 model

Signed-off-by: Kacper Szczesniak <kacper@qwe.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 00:55:19 +02:00
Peter Ujfalusi
eaf1ac8bb5 ASoC: TWL4030: Check the interface format for 4 channel mode
In addition to the operating mode check, also check the
codec's interface format in case of four channel mode.
If the codec is not in TDM (DSP_A) mode, return with error.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-01 23:46:08 +01:00
Jaroslav Kysela
0e4835c198 ALSA: hda-intel: improve initialization for ALC262_HP_BPC model
Fix issues for 3 generations of HP workstations.

The modest modifications do the following:
1. Change the second MIC from device 3 to device 1
2. Init the "boost" values to "0" by default

From: John Brown <john.brown3@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-01 19:14:45 +02:00
Nickolas Lloyd
7c922de709 ALSA: hda - Jack Mode changes for Sigmatel boards
This patch changes Line In as Out Switch and Mic In as Out Switch to
enums for consistency, and causes all mic and line in ports to be probed
and controls to be added appropriately.

Signed-off-by:  Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-01 11:12:29 +02:00
Wei Ni
a3d6ab9723 ALSA: hda - Support NVIDIA 8 channel HDMI audio
Support 8 channel HDMI audio for MCP78/7A

Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-01 11:05:26 +02:00
Takashi Iwai
6efd2cd5e8 ALSA: usb-audio - Add quirk for Roland/Edirol M-16DX
Added a half-working quirk for Roland/Edirol M-16DX.
This enables the capture on the device but the playback on it seems still
problematic becuase of lack of sync with the capture clock.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-01 10:59:51 +02:00
Andrea Borgia
93bfd01227 ALSA: usb-audio - quirk for USB Aureon cards
Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII
USB card.

Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-01 10:48:54 +02:00
Takashi Iwai
8a933ece41 ALSA: hda - Fix a typo in the previous patch
ICH6_GCTL_RESET was wrongly set to another bit by the commit
b21fadb9c1.  This caused a problem when
the codec needs really a reset (e.g. recovering from the communication
error at probe).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-31 09:28:12 +02:00
Takashi Iwai
ba84bfcd2b ALSA: hda - Fix reverted LED setup for HP
The commit 86d190e77c reverted the bit
flip of LED GPIO for HP DX and DV4-1222nr.  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-30 08:59:03 +02:00
Peter Ujfalusi
16a30fbb0d ASoC: TWL4030: Use reg_cache in twl4030_init_chip
Use the codec->reg_cache instead of the array directly
in twl4030_init_chip for setting the default values.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-29 11:02:48 +01:00
Michal Marek
0528c7494e ALSA: clean up the logic for building sequencer modules
Instead of mangling the CONFIG_* variables in the makefiles over and
over, set a few helper variables in Kconfig.

Signed-off-by: Michal Marek <mmarek@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:49:42 +02:00
Jaroslav Kysela
a4444da31e ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value
When hardware has large FIFO, it is necessary to lower jiffies margin
by count of queued samples.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:47:40 +02:00
Jaroslav Kysela
13f040f9e5 ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
Some hardware might have bigger FIFOs and DMA pointer value will be updated
in large chunks. Do not update hw_ptr_jiffies and position timestamp when
hw_ptr value was not changed.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:47:38 +02:00
Jaroslav Kysela
c62a01ad6e ALSA: PCM midlevel: introduce mask for xrun_debug() macro
For debugging purposes, it is better to separate actions.

Bit-values:

	1: show bad PCM ring buffer pointer
	2: show also stack (to debug kernel latency issues)
	4: check pointer against system jiffies

Example:

	5: show bad PCM ring buffer pointer and do jiffies check

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:47:36 +02:00
Jaroslav Kysela
8bea869c5e ALSA: PCM midlevel: improve fifo_size handling
Move the fifo_size assignment to hw->ioctl callback to allow lowlevel
drivers overwrite the default behaviour.

fifo_size is in frames not bytes as specified in asound.h and alsa-lib's
documentation, but most hardware have fixed byte based FIFOs. Introduce
internal SNDRV_PCM_INFO_FIFO_IN_FRAMES.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:47:33 +02:00
Takashi Iwai
e93721a702 Merge branch 'fix/pcm-jiffies-check' into topic/pcm-jiffies-check 2009-05-29 11:46:10 +02:00
Jaroslav Kysela
e9ab33d03e ALSA: au88x0: fix wrong period_elapsed() call
The period_elapsed() call should be called when position moves.

The idea was taken from ALSA bug#4455.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 08:15:59 +02:00
Jaroslav Kysela
3fd43858c7 ALSA: au88x0: fix .pointer callback
Appearently, the used mask in the .pointer callback is invalid. It should
be in period_bytes range. The period_bytes is pow(2), so simple bitwise
operation is used.

Idea was taken from ALSA bug#4455.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 08:15:57 +02:00
Mark Brown
203350c1a8 ASoC: Initialise dev for the dummy S/PDIF DAI
Also include the header to make sure the DAI is prototyped.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-28 18:54:52 +01:00
Chaithrika U S
be461ba836 ASoC: Add dummy S/PDIF codec support
McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed.
This patch provides stub codec that can be used in these configurations.
On DM646x EVM the McASP1 is connected to the S/PDIF out.

Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-28 14:48:38 +01:00
Takashi Iwai
b4f8b5e2f5 Merge branch 'fix/hda' into topic/hda 2009-05-28 13:12:48 +02:00
Takashi Iwai
b21fadb9c1 ALSA: hda - Add more register bits definitions
Added some missing register bits definitions to reduce magic numbers.
Also renamed some to follow the names on the datasheet.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-28 12:26:15 +02:00
Takashi Iwai
817682c11b Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Compaq Presario CQ60 patching for Conexant
2009-05-28 12:02:13 +02:00
Takashi Iwai
b05a7d4fed ALSA: hda - Always sync writes in single_cmd mode
In the single_cmd mode, the hardware cannot store the multiple replies
like on RIRB, thus each verb has to sync and wait for the response no
matter whether the return value is needed or not.  Otherwise it may
result in a wrong return value from the previous verb.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-28 12:01:24 +02:00
Roel Kluin
449bd54dcb ASoC: correct print specifiers for unsigneds
Unsigned variables should use `%u' rather than `%d'.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-28 10:20:26 +01:00
Tony Vroon
1812e67c74 ALSA: hda - Compaq Presario CQ60 patching for Conexant
A docking mic control is shown by default. The Compaq Presario
CQ60 laptop has no docking connector, so designate it as a
CXT5051_HP model.
This makes the phantom mixer slider disappear.

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-28 07:36:20 +02:00
Jon Smirl
ea8b27ad0c ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
The function signature for spin_event_timeout() has changed in version V9.
Adjust the mpc5200 AC97 driver to use the new function.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-27 21:10:37 +01:00
Takashi Iwai
f5219b6195 Merge branch 'fix/pcm-jiffies-check' into for-linus
* fix/pcm-jiffies-check:
  ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mode
  ALSA: Fix invalid jiffies check after pause
2009-05-27 16:51:27 +02:00
Takashi Iwai
f00452cfdc Merge branch 'fix/misc' into for-linus
* fix/misc:
  sound: usb-audio: make the MotU Fastlane work again
2009-05-27 16:51:15 +02:00
Mark Brown
08d15f034e ASoC: Switch FSL SSI DAI over to symmetric_rates
The effect of symmetric_constraints should provide a standard way to
enforce the use of the same sample rate for both directions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
2009-05-27 11:12:45 +01:00
Clemens Ladisch
55de5ef970 sound: usb-audio: make the MotU Fastlane work again
Kernel 2.6.18 broke the MotU Fastlane, which uses duplicate endpoint
numbers in a manner that is not only illegal but also confuses the
kernel's endpoint descriptor caching mechanism.  To work around this, we
have to add a separate usb_set_interface() call to guide the USB core to
the correct descriptors.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: David Fries <david@fries.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-27 11:25:33 +02:00
Takashi Iwai
c87d973200 ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mode
The PCM hw_ptr jiffies check results sometimes in problems when a
hardware doesn't give smooth hw_ptr updates.  So far, au88x0 and some
other drivers appear not working due to this strict check.
However, this check is a nice debug tool, and the capability should be
still kept.

Hence, we disable this check now as default unless the user enables it
by setting the xrun_debug mode to the specific stream via a proc file.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-27 11:04:30 +02:00
Takashi Iwai
6af3fb72d2 ALSA: Fix invalid jiffies check after pause
The hw_ptr_jiffies has to be reset properly to avoid the invalid
check of jiffies delta in snd_pcm_update_hw_ptr*() functions.
Especailly this patch fixes the bogus jiffies check after the puase
and resume.

This patch is a modified version of the original patch by Jaroslav.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-27 11:04:18 +02:00
Mark Brown
0c0e09e21a ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
These drivers use spin_event_timeout() which is only present in the
PowerPC tree at present and which is undergoing some API revisions
so temporarily mark them as BROKEN until these issues are sorted
out.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-26 21:14:59 +01:00
Jon Smirl
6ffee43ecf ASoC: Fabric bindings for STAC9766 on the Efika
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-26 21:13:02 +01:00
Jon Smirl
a9262c4fd4 ASoC: Support for AC97 on Phytec pmc030 base board.
A wm9712 AC97 codec is used.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-26 21:13:01 +01:00
Jon Smirl
20d0e1520e ASoC: AC97 driver for mpc5200
I've implemented retries for when the AC97 hardware doesn't reset on
first try. About 10% of the time both the Efika and pcm030 AC97 codecs
don't reset on first try and need to be poked multiple times.  Failure
is indicated by not having the link clock start ticking. Every once in
a while even five pokes won't get the link started and I have to power
cycle.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-26 21:13:00 +01:00
Jon Smirl
dbcc347562 ASoC: Main rewite of the mpc5200 audio DMA code
Rewrite the mpc5200 audio DMA code to support both I2S and AC97.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-26 21:12:50 +01:00
Linus Torvalds
878a4f521b Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Add missing check of pin vref 50 and others in Realtek codecs
  ALSA: hda - Add 5stack-no-fp model for STAC927x
  ALSA: hda - Add forced codec-slots for ASUS W5Fm
2009-05-26 12:14:46 -07:00
Takashi Iwai
aae80dc24a ALSA: ctxfi - Add missing module parameter definitions
Added missing module_param*() and MODULE_PARM*().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 18:35:27 +02:00
Takashi Iwai
aa2936f5fe ALSA: hda - Support sync after writing a verb
This patch adds a debug mode to make the codec communication
synchronous.  Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c,
and the call of snd_hda_codec_write*() will become synchronous,
i.e. wait for the reply from the codec at each time issuing a verb.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 17:23:46 +02:00
Takashi Iwai
8174086167 ALSA: hda - Allow concurrent RIRB access in single_cmd mode
In the single_cmd mode, the current driver code doesn't do any update
for RIRB just for any safety reason.  But, actually the RIRB and
single_cmd mode don't conflict.  Unsolicited events can be delivered
even while using the single_cmd mode.

This patch allows the handling of unsolicited events with single_cmd
mode, just always checking RIRB independent from single_cmd flag.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 15:22:00 +02:00
Takashi Iwai
20e91c5750 Merge branch 'fix/hda' into topic/hda 2009-05-26 15:19:56 +02:00
Takashi Iwai
86d190e77c ALSA: hda - Minor clean up of patch_sigmatel.c
- Remove unneeded semicolons
- Introduce spec->gpio_led to specify the GPIO bit for LED control

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 15:18:58 +02:00
Takashi Iwai
db1005ec6f ALSA: riptide - Fix joystick resource handling
The current code doesn't handle the multiple gameports properly,
and uses unnecessary global static variables to store the data.
This patch changes the probe / remove routines to use the driver
data assigned to the dedicated pci device, and adds the support of
multiple devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 13:55:42 +02:00
Takashi Iwai
a693a26fe0 ALSA: riptide - Code clean up
A code clean up, coding style fixes.
The firmware loading routine is split to an own function to improve
the readability.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-26 12:58:58 +02:00
Takashi Iwai
4fcd39207f ALSA: hda - Reset CORB/RIRB at retrying the verb communication
When a codec communication error occurs, the CORB/RIRB counters should
be reset first before re-issuing the verb.  Simply call azx_free_cmd_io()
and azx_init_cmd_io() to achieve that.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 18:34:52 +02:00
Peter Ujfalusi
0f89bdcac6 ASoC: TWL4030: HandsfreeL/R mute DAPM switch
Add DAPM switch for HeadsetL/R mute. Since all bits are are needed
for the HFL/R pop removal to work the switch is using the SW_SHADOW
no HW register for the HandsfreeL/R mute.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 11:54:14 +01:00
Peter Ujfalusi
f3b5d3002d ASoC: TWL4030: Add shadow register
Shadow, non HW register for dealing with the HandsfreeL/R
muting.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 11:54:14 +01:00
Peter Ujfalusi
5a2e9a48b1 ASoC: TWL4030: Handsfree pop removal redesign
Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E
to a more appropriate DAPM_PGA_E widget.
Also fix the power-up sequence to match with the TRM.
The power-down sequence is not described in the TRM, so do it
in a way, which seams like the correct sequence.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 11:54:13 +01:00
Clemens Ladisch
04f9890df1 sound: virtuoso: add Xonar Essence ST support
Add support for the Asus Xonar Essence ST and its daughterboard.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 11:49:58 +02:00
Clemens Ladisch
b990ae963a sound: virtuoso: enable HDAV S/PDIF input
The Xonar HDAV1.3 has a digital input jack, so enable the corresponding
device.

This is not related to the HDMI stuff, which stays unsupported.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 11:49:52 +02:00
Clemens Ladisch
53bb705d12 sound: virtuoso: add another DX PCI ID
Add another PCI ID for a second revision of the Xonar DX.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 11:49:44 +02:00
Clemens Ladisch
345c03ef0f sound: oxygen: reset DMA when stream is closed
When a PCM stream is closed, flush the corresponding DMA channel.
Otherwise, the DMA controller would continue to output the last sample
which would result in a DC offset on the output.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 11:49:30 +02:00
Takashi Iwai
461c6c3a0a ALSA: hda - Add missing check of pin vref 50 and others in Realtek codecs
Some Realtek codecs like ALC861 seem to support only VREF50 while the
current driver assumes it's only VREF80.  Check other VREF bits to set
the correct value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-25 08:06:02 +02:00
Jon Smirl
cebe77674c ASoC: Rename the PSC functions to DMA
Rename the functions in the mpc5200 DMA file from i2s based names to dma
ones to reflect the file they are in.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-24 19:31:03 +01:00
Jon Smirl
89dd084252 ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2s
Basic split of mpc5200 DMA code out from i2s into a standalone file.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-24 19:31:03 +01:00
Takashi Iwai
679d92ed14 ALSA: hda - Add 5stack-no-fp model for STAC927x
The recent fix for the headphone volume control on IDT/STAC codecs
resulted in the removal of invalid "Side" volume eventually.  But,
if the front panel doesn't exist, this setup could be regarded as a
sort of regression, as reported in kernel bug #13250.

Now as a workaround, a new model 5stack-no-fp is added so that the user
without the front panel can choose this one explicitly.

Reference: bko#13250
	http://bugzilla.kernel.org/show_bug.cgi?id=13250

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-24 19:00:08 +02:00
Ozan Çağlayan
93574844bc ALSA: hda - Add forced codec-slots for ASUS W5Fm
ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS
problem like W5F.

Tested-by: Alp Kılıç <kilic.alp@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-24 18:45:28 +02:00
Mark Brown
05e1efa2de ASoC: Fix minor issues in STAC9766 driver
Fairly minor issues:
 - Don't register the DAIs, it's not required for AC97 devices.
 - Make unexported functions static.
 - Wrap some excessively long lines.
 - Undo tab/space breakage.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-24 13:32:24 +01:00
Jon Smirl
3c166c7f18 ASoC: Codec for STAC9766 used on the Efika
Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-24 13:15:21 +01:00
Mark Brown
0154724d48 ASoC: Fix WM9081 PowerPC compiler issues
Ensure that we always set a new sysclk when using the FLL in master mode
and pick out the correct value for the sample rate in hw_params().

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-23 10:24:15 +01:00
Takashi Iwai
4986cab555 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control name
  ALSA: pcsp - fix printk format warning again
2009-05-22 19:29:08 +02:00
Andreas Mohr
afe6d7e3c4 ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control name
ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase
Inversion Playback Switch' truncated to 'Sigmatel Surround Phase
Inversion Playback ' bootup message by omitting weird Sigmatel prefix
in this case; also fix up the related ca0106 mixer control removal
part by using identical naming there.

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-22 19:27:13 +02:00
Mark Brown
86ed3669f0 ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 15:11:22 +01:00
Peter Ujfalusi
b4852b793a ASoC: TWL4030: Differentiate the playback streams
Give unique stream names for the two playback streams so
DAPM can figure out which codec_dai is in use.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 15:08:43 +01:00
Peter Ujfalusi
7385ba44f8 ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal
With this patch the initial headset pop-removal related values are
configured for the twl4030 codec (ramp delay and sysclk).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 10:23:22 +01:00
Peter Ujfalusi
9da28c7b38 ASoC: TWL4030: Add support for platform dependent configuration
twl4030_setup_data structure can be passed from platform drivers to
the codec via the snd_soc_device->codec_data pointer.

Currently the setup data has support for the Headset pop-removal
related configuration, which differs from board to board.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 10:23:22 +01:00
Takashi Iwai
89b7161c48 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - fix audio on HP TX25xx series notebooks
2009-05-22 08:23:39 +02:00
Adam Williamson
87488957a6 ALSA: hda - fix audio on HP TX25xx series notebooks
Fixes https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4121

Taken from https://bugzilla.redhat.com/show_bug.cgi?id=498060

Signed-off-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-22 08:22:23 +02:00
Paul Mundt
5f8371cec9 Merge branches 'sh/stable-updates' and 'sh/sparseirq' 2009-05-22 13:29:37 +09:00
Ben Dooks
99ae99533a [ARM] S3C24XX: Merge devel-gpio
Merge branch 'devel-gpio' into for-rmk-devel
2009-05-21 22:10:21 +01:00
Alessandro Rubini
03fbdb15c1 [ARM] 5519/1: amba probe: pass "struct amba_id *" instead of void *
The second argument of the probe method points to the amba_id
structure, so it's better passed with the correct type. None of the
current in-tree drivers uses the pointer, so they have only been
checked for a clean compile.

Change suggested by Russell King.

Signed-off-by: Alessandro Rubini <rubini@unipv.it>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
2009-05-20 23:26:51 +01:00
Takashi Iwai
b3b778b387 ALSA: pcsp - fix printk format warning again
The commit 5a641bcd63 changed the
printk format to '%lu', but the value passed seems to be dependent
on the architecture.  On x86-64, I got a new warning now because an
int value is passed actaully.

As a workaround, just cast the value always to unsigned long.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-20 17:08:00 +02:00
Takashi Iwai
3e3ee6dc94 ALSA: ctxfi - Add depends on X86
The ctxfi driver requires explicitly the 4k page size, and gives a
build error on architectures with non-4k pages.
As a workaround, just add the kconfig dependency on X86, which is
the only architecture ever tested.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-20 16:47:59 +02:00
Peter Ujfalusi
6943c92e87 ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event
This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles
the headset ramp up and down sequences needed for the pop noise
removal.

With this patch the order of the internal components in the twl4030
codec is turned on and off in a correct order.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz  <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-20 09:53:25 +01:00
Peter Ujfalusi
4005d39a5f ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs
Restructuring the twl4030 codec's DAPM routing to be able to handle the power
sequences correctly.

The twl4030 codec internal implementation have this order:
DAC -> Analog PGA -> Mixer/Mux

While the ASoC framework expects the following order:
DAC -> Mixer -> Analog PGA

This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and
adds two levels of mixer to handle the digital and analog loopback
functionality.

Now the analog loopback does not powers on any of the DACs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz  <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-20 09:52:51 +01:00
Atsushi Nemoto
e24805dd85 ASoC: Add TXx9 AC link controller driver (v3)
This patch adds support for the integrated ACLC of the TXx9 family.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 19:54:28 +01:00
Takashi Iwai
fa79796631 ALSA: hda - Fix digital beep tone calculation
The digital beep tone is calculated in two different ways depending
on the codec chip.  The standard one is using a divider, and another
one is a linear tone for IDT/STAC codecs.  Currently, only the
latter type is used for all codecs, which resulted in a wrong tone
pitch.

This patch adds the calculation of the standard HD-audio type.
Also clean-up the fields in hda_beep struct.

Reference: bko#13162
	http://bugzilla.kernel.org/show_bug.cgi?id=13162

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-19 12:54:55 +02:00
Takashi Iwai
4abc1cc2f9 ALSA: hda - Add prefix to kernel messages
Add proper prefix to each kernel message in hda_intel.c.
Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used
together with snd_print*().

Reference: bko#13207
	http://bugzilla.kernel.org/show_bug.cgi?id=13207

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-19 12:16:46 +02:00
Lopez Cruz, Misael
11a7281106 ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3
Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver.
Voice DAI init function enables corresponding interface by
writting directly to VOICE_IF codec register.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 10:35:11 +01:00
Lopez Cruz, Misael
b74bd40fa4 ASoC: TWL4030: Add control for selecting codec operation mode
Add a control for selecting the codec operation mode. TWL4030 codec
has two modes:
- Option 1. Audio only (4 audio DACs)
- Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC)

Control is restricted when a stream is ongoing, since codec's
operation mode cannot be changed on-the-fly.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 10:35:11 +01:00
Peter Ujfalusi
181da78cd0 ASoC: TWL4030: Fix Analog capture path for AUXR
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 09:38:57 +01:00
Ben Dooks
ec976d6eb0 [ARM] S3C24XX: GPIO: Move gpio functions out of <mach/hardware.h>
Move all the gpio functions out of <mach/hardware.h> as
this file is for defining the generic IO base addresses
for the kernel IO calls.

Make a new header <mach/gpio-fns.h> to take this and
include it via the chain from <linux/gpio.h> which is
what most of these files should be using (and will be
changed as soon as possible).

Note, this does make minor changes to some drivers but
should not mess up any pending merges.

CC: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
2009-05-18 16:25:40 +01:00
Mark Brown
f83fba8baa ASoC: Add debug trace for bias level transitions
A standard way of making sure we know when the bias level changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:16 +01:00
Mark Brown
452c5eaa0d ASoC: Integrate bias management with DAPM power management
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:16 +01:00
Mark Brown
aef908434c ASoC: Make DAPM sysfs entries non-optional
sysfs is so standard these days there's no point.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:15 +01:00
Mark Brown
6d3ddc81f5 ASoC: Split DAPM power checks from sequencing of power changes
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.

The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:14 +01:00
Torben Schulz
eb4c41d30b ALSA: hda - Improved MacBook 3,1 support
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.

Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-18 15:13:02 +02:00
Takashi Iwai
313f6e2d40 ALSA: hda - Avoid conflicts with snd-ctxfi driver
The PCI entries of Creative with HD-audio class can be the devices
with emu20k1/emu20k2 chips.  These are supported better by snd-ctxfi
driver.  With that driver, the device will mutate from HD-audio to
its native class.

This patch adds a simple ifdef to avoid the conflict of device probe
between snd-hda-intel and snd-ctxfi drivers.  1102:0009 seems still
OK to be added as it has no emu20kx chip, and is a pure HD-audio
device.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-18 12:40:52 +02:00
Takashi Iwai
6c627f3978 ALSA: hda - Show the actual chip name in 'unkown model' messages
Show the actual chip name in 'unknown model..' info messages for
Realtek codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-18 12:33:36 +02:00
Misael Lopez Cruz
b7a755a8a1 ASoC: TWL4030: Enable/disable voice digital filters
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink
and downlink paths, respectively.

This patch also corrects voice 8/16kHz mode selection bit
(SEL_16K) of CODEC_MODE register.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 11:13:12 +01:00
Julia Lawall
5b740ea975 sound: use dev_set_drvdata
Eliminate direct accesses to the driver_data field.
cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd015

The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
struct device *dev;
expression E;
type T;
@@

- dev->driver_data = (T)E
+ dev_set_drvdata(dev, E)

@@
struct device *dev;
type T;
@@

- (T)dev->driver_data
+ dev_get_drvdata(dev)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-17 19:31:49 +02:00
Takashi Iwai
812a2cca29 ALSA: hda - Split codec->name to vendor and chip name strings
Split the name string in hda_codec struct to vendor_name and chip_name
strings to be stored directly from the preset name.
Since mostly only the chip name is referred in many patch_*.c, this
results in the reduction of many codes in the end.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-16 10:00:49 +02:00
Alexander Beregalov
b04b4f7862 ALSA: parisc/harmony: fix printk format warning
Fix this warning:
sound/parisc/harmony.c:938: warning: format '%lx' expects type 'long unsigned int',
but argument 2 has type 'resource_size_t'

Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-15 16:05:37 +02:00
Takashi Iwai
60befb97f5 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: DaVinci EVM board support buildfixes
  ASoC: DaVinci I2S updates
  ASoC: davinci-pcm buildfixes
  pxa2xx-ac97: fix reset gpio mode setting
  ASoC: soc-core: fix crash when removing not instantiated card
2009-05-15 15:38:26 +02:00
Takashi Iwai
a3d2b755c0 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: pcsp: fix printk format warning
  ALSA: riptide: postfix increment and off by one
2009-05-15 15:38:20 +02:00
Nickolas Lloyd
2fc9989079 ALSA: hda - add controls to toggle DC bias on mic ports
This patch adds a mixer control for the STAC92XX boards to control the
DC bias of mic ports, allowing recording from both powered and
non-powered sources.  It replaces the "Mic Output Switch" with "Mic Jack
Mode" to switch between Mic, Line In, and Line Out.

Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-15 15:34:10 +02:00
Takashi Iwai
2bf2778e0f ASoC: Optimize switch/case in magician.c
Use default to optimize the switch/case in magicial_playback_hw_params(),
which also fixes the compile warnings below:
  sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this function
  sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-15 12:20:52 +02:00
Takashi Iwai
2baaec2806 ASoC: Add missing __devexit in wm8940.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-15 12:18:47 +02:00
Takashi Iwai
5f592bb99b Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-05-15 10:30:21 +02:00
Mark Brown
81a4275bff Merge branch 'for-2.6.30' into for-2.6.31 2009-05-15 08:59:52 +01:00
David Brownell
f492ec9f02 ASoC: DaVinci EVM board support buildfixes
This is a build fix, resyncing the DaVinci EVM ASoC board code
with the version in the DaVinci tree.  That resync includes
support for the DM355 EVM, although that board isn't yet in
mainline.

(NOTE:  also includes a bugfix to the platform_add_resources
call, recently sent by Chaithrika U S <chaithrika@ti.com> but
not yet merged into the DaVinci tree.)

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-15 08:59:07 +01:00
David Brownell
a62114cb90 ASoC: DaVinci I2S updates
This resyncs the DaVinci I2S code with the version in the DaVinci
tree.  The behavioral change uses updated clock interfaces which
recently merged to mainline.  Two other changes include adding a
comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in
order to support more boards than just the DM644x EVM.

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-15 08:58:58 +01:00
David Brownell
82075af6cb ASoC: davinci-pcm buildfixes
This is a buildfix for the DaVinci PCM code, resyncing it with
the version in the DaVinci tree.  The notable change is using
current EDMA interfaces, which recently merged to mainline.
(The older interfaces never made it into mainline.)

NOTE:  open issue, the DMA should be to/from SRAM; see chip
errata for more info.  The artifacts are extremely easy to
hear on DM355 hardware (not yet supported in mainline), but
don't seem as audible on DM6446 hardwaare (which does have
mainline support).

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-15 08:58:05 +01:00
Magnus Damm
639138a991 sound: oss: sh_dac_audio timer fixes.
This patch modifies sh_dac_audio in the following ways:

- use high resolution timer instead of TMU1
- fix cpu/dac.h include
- add future rewrite comment

Signed-off-by: Magnus Damm <damm@igel.co.jp>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
2009-05-15 12:07:17 +09:00
Marek Vasut
63c26baa2a ASoC: Support AC97 link off by default on WM9712
The WM9712 can be configured by resistor strapping GPIO4 to behave like
the WM9713 and default to leaving the AC97 link disabled after cold
reset until a warm reset occurs.  In this configuration we need to issue
a warm reset after cold to bring the link up so do so.  The warm reset
will be harmless on systems that don't need it.

[Changelog rewritten to document the reasoning. -- broonie]

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 20:52:46 +01:00
Mark Brown
c13cb78ace Merge branch 'pxa2xx-i2s' into for-2.6.31 2009-05-14 20:35:51 +01:00
Karl Beldan
916465a841 ASoC: pxa2xx-i2s: Fix suspend/resume
pxa2xx_i2s_resume is :
 - unconditionnaly setting SACR0_ENB
 - unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0
fix these.

In pxa2xx_i2s_{resume,suspend}, save/restore registers even
when !dai->active.

Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 20:35:32 +01:00
Karl Beldan
9bc04fd167 ASoC: pxa2xx-i2s: Fix inappropriate release of i2s clock
i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown.
Now we 'get' i2s_clk at probe and 'put' it at driver removal or when
probe fails.

Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 20:35:31 +01:00
Karl Beldan
34555c1077 ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
- hw_params enables both RPL and REC functions each time : Enable the
	appropriate function in pxa2xx_i2s_trigger.
- pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is
	off : Turn it off only when both functions are off.

Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 20:35:23 +01:00
Karl Beldan
b243b77c70 ASoC: pxa2xx-i2s: Proper hw initialization
Make sure we are in a know good state at end of probe :
Reset FIFO logic and registers, and make sure REC and RPL functions
along with FIFO service are disabled (SACR0_RST enables REC and RPL).

Resetting loses current settings so remove reset from stream startup.
Now reset occurs only at probe.

Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 20:34:40 +01:00
Takashi Iwai
7442f9dadb ALSA: hda - Add a quirk entry for Macbook Pro 5,1
Added the codec SSID for MacBook Pro 5,1 as compatible as MP51.
However, the headphone auto-muting function doesn't work.  So,
this is just a tentative solution.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 18:20:33 +02:00
Randy Dunlap
5a641bcd63 ALSA: pcsp: fix printk format warning
Fix printk format warning:

sound/drivers/pcsp/pcsp_mixer.c:54: warning: format '%d' expects type 'int', but argument 3 has type 'long unsigned int'

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 18:03:10 +02:00
Takashi Iwai
b3e0afe61e ALSA: ctxfi - Add prefix to debug prints
Added ctxfi: prefix to each debug print.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 15:19:30 +02:00
Takashi Iwai
35b053becb ALSA: ctxfi - Avoid unneeded pci_read_config_*() calls
Use struct pci subsystem_device and revision fields instead of
unneeded calls of pci_read_config_*().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 15:17:16 +02:00
Takashi Iwai
9fc20f030b ALSA: ctxfi - Move PCI ID definitions to linux/pci_ids.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 15:14:18 +02:00
Jean Delvare
903dba1eae ALSA: keywest: Get rid of useless i2c_device_name() macro
The i2c_device_name() macro is used only once and doesn't have much
value, it hurts redability more than it helps. Get rid of it.

Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 14:42:19 +02:00
Anuj Aggarwal
14610ce711 ASoC: Added OMAP3 EVM support in ASoC.
Resending the patch after fixing the minor issues.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 12:47:26 +01:00
Takashi Iwai
d0da727e02 ALSA: ctxfi - Add missing inclusion of linux/delay.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 10:56:04 +02:00
Roel Kluin
1b1cc7f21c ALSA: riptide: postfix increment and off by one
With a postfix increment these variables are incremented beyond
CMDIF_TIMEOUT / MAX_WRITE_RETRY.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 08:41:40 +02:00
Wai Yew CHAY
8cc7236148 ALSA: SB X-Fi driver merge
The Sound Blaster X-Fi driver supports Creative solutions based on
20K1 and 20K2 chipsets.

Supported hardware :

Creative Sound Blaster X-Fi Titanium Fatal1ty® Champion Series
Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
Creative Sound Blaster X-Fi Titanium Professional Audio
Creative Sound Blaster X-Fi Titanium
Creative Sound Blaster X-Fi Elite Pro
Creative Sound Blaster X-Fi Platinum
Creative Sound Blaster X-Fi Fatal1ty
Creative Sound Blaster X-Fi XtremeGamer
Creative Sound Blaster X-Fi XtremeMusic

Current release features:

* ALSA PCM Playback
* ALSA Record
* ALSA Mixer

Note:

* External I/O modules detection not included.

Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com>
Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-14 08:24:10 +02:00
Karl Beldan
8c10dc4f54 ASoC: pxa2xx-i2s: Proper initialization
Reset FIFO logic and registers, and make sure REC and RPL functions along
with FIFO service are disabled at probe.

Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-13 19:16:00 +01:00
Takashi Iwai
22b530e028 ALSA: hda - Disable fallback to model=acer for Acer laptops
The model=acer for ALC883/889 doesn't work well for the recent Acer
Aspire laptops.  Since model=auto works better nowadays, it's safer
to use the default fallback instead of the Acer specific one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-13 11:32:52 +02:00
Takashi Iwai
dd4a416442 Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-05-12 12:32:07 +02:00
Takashi Iwai
ddc4097b77 Merge branch 'topic/drvdata-fix' into topic/asoc 2009-05-12 11:57:09 +02:00
Takashi Iwai
511b4c171a ALSA: hdsp - Add a comment about external firmwares for hdsp
When the hdsp driver is built in kernel, the corresponding firmware
files have to be built into the kernel as well (otherwise the boot
will hang up for fairly long time), but there is no way to control
it in Kconfig yet, unfortunately.  Instead, show a comment for user
just to make sure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-12 11:53:45 +02:00
Mark Brown
13e2c86c20 Merge branch 'for-2.6.30' into for-2.6.31 2009-05-12 10:53:33 +01:00
Joonyoung Shim
97b8096dc9 ASoC: TWL4030: change DAPM for analog microphone selection
The inputs of the twl4030 codec can be mixed, so we will use the mixer
DAPM for the analog microphone registers(0x05, 0x06), but if we enable
more than one input at the same time, the input impedance of the input
amplifier will be reduced.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-12 10:53:22 +01:00
Mark Brown
7de0a0aee5 ASoC: Enforce symmetric rates for PXA2xx I2S
There is a single I2S_SYNC pin on the chip.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-12 10:53:22 +01:00
Mike Rapoport
eaaa532883 ASoC: em-x270: make the driver support also eXeda and CM-X300 machines
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-12 10:53:22 +01:00
Raphael Doursenaud
8e365f9b6e ALSA: hdsp: allow firmware loading from inside the kernel
Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and
in-kernel driver.
Also corrected a typo in the comment.

Signed-off-by: Raphael Doursenaud <rdoursenaud@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-12 11:49:39 +02:00