Commit Graph

10509 Commits

Author SHA1 Message Date
Mark Brown
f0f5039c3d ASoC: Handle failed WM8994 FLL lock waits
Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17 13:15:00 +09:00
Arnd Bergmann
6277839602 Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixes 2011-07-15 21:56:37 +02:00
Mark Brown
1479c3fb5f ASoC: Handle spurious wm_hubs DC servo done interrupts
Don't assume the first fire indicates that we're done.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15 23:48:05 +09:00
Dimitris Papastamos
6b3860b0a2 ASoC: WM8983: Initial driver
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15 23:39:53 +09:00
Mark Brown
47d90a03eb Merge branch 'for-3.0' into for-3.1 2011-07-15 22:43:07 +09:00
Mark Brown
b35e160a11 ASoC: Fix shift in WM8958 accessory detection default implementation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-15 22:41:38 +09:00
Daniel T Chen
f21169aa87 ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
BugLink: https://bugs.launchpad.net/bugs/774895

The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.

Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-15 07:54:26 +02:00
Takashi Iwai
00ef9610ac ALSA: hda - Fix krealloc() replacement in hda_codec.c
It was obviously wrong, grr....

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14 15:58:02 +02:00
Takashi Iwai
7b1655f5f2 ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs
During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.

This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14 15:33:59 +02:00
Axel Lin
58499906c8 ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 20:04:26 +09:00
Mark Brown
3b1af3f8c8 ASoC: Log WM8994 FIFO errors from the interrupt
We should spot them anyway on state changes but logging them gives us
better time information about when the misconfiguration happened.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 17:17:23 +09:00
Giridhar Maruthy
b3d7615f2a ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.

Signed-off-by: Giridhar Maruthy <giridhar.maruthy@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 08:25:42 +09:00
Mark Brown
f05bdb8bb6 ASoC: Don't warn on low WM8994/58 AIFnCLKs
We can have valid but very low clocks in accessory detection modes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:30 +09:00
Mark Brown
c7ebf932e5 ASoC: Use WM8994 FLL lock interrupt
If we have interrupts then wait for the FLL lock interrupt rather than
using dead reckoning when waiting for the FLL to start.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:22 +09:00
Mark Brown
b30ead5f39 ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:14 +09:00
Mark Brown
d96ca3cd0b ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:04 +09:00
Mark Brown
b70a51bab9 ASoC: Use late enable handling for direct voice, speaker and headphone
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:37:52 +09:00
Johannes Stezenbach
889ebae537 ASoC: STA32x: Preserve reserved register bits
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched.  It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:24:32 +09:00
Johannes Stezenbach
7968843915 ASoC: STA32x: Add mixer controls for biquad coefficients
The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1.  The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).

These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:24:31 +09:00
Paul Menzel
cf01b73e26 ALSA: hda - fix up typos in Kconfig help for default buffer size introduced in acfa634f
This commit is a fix up for commit acfa634f.

	commit acfa634f7e
	Author: Takashi Iwai <tiwai@suse.de>
	Date:   Tue Jul 12 17:27:46 2011 +0200

		  ALSA: hda - Add Kconfig for the default buffer size

Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 20:17:23 +02:00
Guillaume Pellerin
0f5733b0c8 ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.

Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :

    options snd_usb_audio   vid=0x763 pid=0x2012 device_setup=0x08

Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf

Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 18:15:45 +02:00
Takashi Iwai
acfa634f7e ALSA: hda - Add Kconfig for the default buffer size
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 17:31:46 +02:00
Takashi Iwai
3101ba035c ALSA: Use krealloc() in possible places
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 08:05:16 +02:00
Takashi Iwai
30b4503378 ALSA: hda - Expose secret DAC-AA connection of some VIA codecs
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:45:02 +02:00
Takashi Iwai
9e7717c9eb ALSA: hda - Always read raw connections for proc output
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:45:01 +02:00
Takashi Iwai
b2f934a0df ALSA: hda - Add snd_hda_override_conn_list() helper function
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:44:46 +02:00
Takashi Iwai
19110595c8 ALSA: hda - Turn on extra EAPDs on Conexant codecs
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD.  Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled.  For avoiding such a problem, turn
all extra EAPDs on as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 14:46:44 +02:00
Jiri Kosina
b7e9c223be Merge branch 'master' into for-next
Sync with Linus' tree to be able to apply pending patches that
are based on newer code already present upstream.
2011-07-11 14:15:55 +02:00
Takashi Iwai
9499473463 ALSA: hda - Preserve input pin-ctl bits in HP-automute for VIA codec
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 11:36:44 +02:00
Takashi Iwai
6e969d9155 ALSA: hda - Set line-out pin-ctls properly when indep-HP mode changes
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 11:28:13 +02:00
Takashi Iwai
21ce0b6527 ALSA: hda - Via Fix speaker-mute checks in VIA driver
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack.  Currently
it checks only the line-out state and ignores the headphone.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 10:33:47 +02:00
Mark Brown
5b7396709e ASoC: Conditionalize the enable of WM8994 ADC TDM mode
Future devices will not benefit from this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-09 23:16:48 +09:00
Mark Brown
3db1bbfd4a Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-3.1 2011-07-09 23:16:12 +09:00
Takashi Iwai
017f2a104c ALSA: hda - Implement 44kHz workaround for IdeadPad as fixup
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-09 14:42:25 +02:00
Mark Brown
3f9c42ed6b Merge branch 'for-3.0' into for-3.1 2011-07-09 19:06:33 +09:00
Kuninori Morimoto
2c7beb9285 ASoC: sh: fsi-hdmi: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:06:16 +09:00
Kuninori Morimoto
f15c941331 ASoC: sh: fsi-da7210: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:06:05 +09:00
Kuninori Morimoto
505b04e0f8 ASoC: sh: fsi-ak4642: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:05:55 +09:00
Takashi Iwai
e8fd86efaa Merge branch 'fix/asoc' into for-linus 2011-07-09 11:56:43 +02:00
Takashi Iwai
abaead6ac5 ALSA: hda - Fix a copmile warning
It's harmless but annyoing.
  sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
  sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-09 11:55:28 +02:00
Takashi Iwai
e320bc42be Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-07-09 11:43:04 +02:00
Mark Brown
71ae391d45 Merge branch 'for-3.0' into for-3.1 2011-07-09 18:20:36 +09:00
Takashi Iwai
18361bbe31 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2011-07-09 09:44:09 +02:00
Takashi Iwai
3e6179b844 ALSA: hda - Merge alc*_parse_auto_config() functions in patch_realtek.c
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function.  A good amount of code reduction.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:55:13 +02:00
Takashi Iwai
8452a982fb ALSA: hda - Merge ALC260 auto-parser code
Finally the last one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:19:48 +02:00
Takashi Iwai
4c11398edc ALSA: hda - Merge ALC269 parser code
One more code reduction.  This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:12:05 +02:00
Takashi Iwai
be9bc37bcc ALSA: hda - Merge ALC268/269 auto-parser codes
Now coming to ALC268/269 parser codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:01:47 +02:00
Takashi Iwai
72dcd8e76b ALSA: hda - Merge ALC861 auto-parser code
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 15:16:55 +02:00
Takashi Iwai
44c0240052 ALSA: hda - Fix amp-cap checks in patch_realtek.c
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec.  Thus one needs to check widget-caps first, then check
the corresponding amp-caps.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 15:14:19 +02:00
Takashi Iwai
a1f649d547 ALSA: hda - Merge ALC861-VD auto-parse to the standard parser
The existing standard auto-parser can work well with this codec, too.
Let's merge.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 14:39:03 +02:00
Takashi Iwai
268ff6fbe7 ALSA: hda - Fix auto-mic detection in Realtek codec-parser
A regression fix from commit 21268961d3
  ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs

The auto-mic wasn't detected properly when no ADC-switch is needed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 14:37:35 +02:00
Lydia Wang
28dc10a5f1 ALSA: hda - Fix output-path of VT1812 codec
For VT1812, add dac_mixer_idx for initialization.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 12:37:19 +02:00
Takashi Iwai
21d45d2ba9 ALSA: hda - Fix Oops in smart51 parsing in VIA codec
Typical off-by-one thinko.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:35:11 +02:00
Takashi Iwai
e477062958 ALSA: hda - Provide the standard auto_init for Realtek codecs
Remove redundant definitions.  Ideally, all init functions should be
identical in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:12:09 +02:00
Takashi Iwai
afcd551508 ALSA: hda - Merge ALC680 auto-parser to the standard parser
Improved the standard Realtek auto-parser to support the codec topology
like ALC680.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:07:59 +02:00
Takashi Iwai
e59ea3ed9f ALSA: hda - Add a fix-up for HP RP5800
The BIOS provides bogus pin configs, and also invalid SSID.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:20:18 +02:00
Takashi Iwai
08ef79490d ALSA: pcmcia - Use pcmcia_request_irq()
The drivers don't require the exclusive irqs.  Let's fix the deprecated
warnings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:11:35 +02:00
Pavel Roskin
81b85b6bd9 ALSA: usb-audio: replace "void *" with more specific pointers
Signed-off-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:10:25 +02:00
Lydia Wang
a2a870c827 ALSA: hda - Fix Independent-HP detection on VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:20:05 +02:00
Lydia Wang
5c9a5615de ALSA: hda - Fix DAC checks for VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:19:28 +02:00
Lydia Wang
d69607b3c3 ALSA: hda - Fix VIA output-path init for VT2002P/1802/1812
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:18:29 +02:00
Mark Brown
b5d5f59be2 Merge branch 'for-3.0' into for-3.1 2011-07-07 09:54:19 -07:00
Axel Lin
e12c28a98f ASoC: pxa2xx-pcm: remove unused variable 'dai'
Remove unused variable 'dai' to eliminate below warning.

  CC      sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 09:54:09 -07:00
Kuninori Morimoto
bd7fdbcaa2 ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
mask didn't cover update-data

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-07 09:46:06 -07:00
Takashi Iwai
1d045db96a ALSA: hda - Split quirk codes from patch_realtek.c
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model.  For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.

The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed.  This is just a refactoring.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:27:29 +02:00
Takashi Iwai
0e4a73ae58 ALSA: hda - Use common paser for digital I/O for ALC260
Avoid open-codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:03:12 +02:00
Takashi Iwai
21268961d3 ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.

In the new code, the following strategy is taken:

- When a cap-src can't handle all input-sources, either skip it, or
  switch to the ADC-switching mode.  In ADC-switching mode, like the
  former dual-ADC mode for ALC275, it changes ADC on the fly according
  to the current input source.
- When auto-mic is possible, always assign imux.  If the mic pins are
  set statically via a quirk, rebuild imux according to the pins.
  In the auto-mic mode, the driver always changes the imux (although
  the imux isn't exposed as a mixer element).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:02:43 +02:00
Takashi Iwai
a926757f04 ALSA: hda - Fix warning with ALC882 digital-out detection
The digital out pin on ALC882 may have multiple connections.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 16:08:13 +02:00
Peter Ujfalusi
21385eeb02 ASoC: twl6040: Add back support for legacy mode
The legacy mode has been accidentaly removed by commit:
ASoC: twl6040: add all ABE DAIs

Add back the twl6040-hifi dai.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:47 +03:00
Peter Ujfalusi
ff593ca1a4 ASoC: twl6040: No need to convert the PLL ID
Since the PLL handling has been simplified, and
rebased on 0, there is no longer need for converting
the PLL ID.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:46 +03:00
Peter Ujfalusi
753621c215 ASoC: twl6040: Configure PLL only once
Avoid configuring the PLL several times during audio startup.
We can configure the PLL at prepare time with parameters collected
earlier hw_param, and set_dai_sysclk calls.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:45 +03:00
Peter Ujfalusi
f53c346c08 ASoC: twl6040: Simplify sample rate constraint handling
We can manage the sample rate constraints without the need
to maintain a variable and a pointer.
This simplifies the handling of the constraint, and makes it
more robust.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi
af958c72af ASoC: twl6040: Move PLL selection to codec driver
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi
7cca606794 ASoC: twl6040: Use neutral name for power mode text/enum
Change the variable names to be neutral (not refering to HS).
This will ease up the introduction of PLL selection, which
going to use the same enum strings.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:43 +03:00
Peter Ujfalusi
2a433b9daf ASoC: twl6040: Do not use wrapper for irq request
The twl6040_request_irq/free_irq inline functions are going
to be removed, so replace them with direct calls.
The irq number is provided by the core driver via resource.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Felipe Balbi <balbi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:22 +03:00
Sascha Hauer
6584cb8825 ARM i.MX dma: Fix burstsize settings
dmaengine expects the maxburst parameter in words, not bytes.
The imxdma driver and its users do this wrong. Fix this.

As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work
with imx-dma. This broke the driver with imx-sdma support which
correctly takes the maxburst parameter in words. This patch
puts the sdma based sound back to work.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
2011-07-07 09:55:50 +02:00
Takashi Iwai
c2d986b0d2 ALSA: hda - Clean-up PCM assignments in patch_realtek.c
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:35:17 +02:00
Takashi Iwai
f970de2555 ALSA: hda - Unify alc*_auto_init_input_src() in patch_realtek.c
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:35:14 +02:00
Takashi Iwai
d6cc9fabd5 ALSA: hda - Parse ADCs and CAPSRCs dynamically for Realtek auto-parser
Now with the new code for looking for ADCs and MUXs, we can replace
the whole ADC assignment with the parsed results.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:34:46 +02:00
Takashi Iwai
0a7f532090 ALSA: hda - Unify alc_auto_init_analog_input() calls
All alc*_auto_init_analog_input() calls are identical, so let's use
the same function more clearly without aliases.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:25 +02:00
Takashi Iwai
b78217096b ALSA: hda - Parse ADCs in alc_auto_create_input_ctls()
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller.  By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:21 +02:00
Takashi Iwai
343a04be37 ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:18 +02:00
Takashi Iwai
97aaab7b49 ALSA: hda - Create bind-mutes appropriately for ALC662 auto-parser
When multiple inputs are present on the mixer widget (typically a DAC
and a loopback), mute/unmute both inputs with the corresponding mixer
element.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:15 +02:00
Takashi Iwai
cd51155676 ALSA: hda - Initialize DACs in ALC662 auto-parser mode
The initialization of DACs was missing in ALC662 parser code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:13 +02:00
Takashi Iwai
bb8bf4d40c ALSA: hda - Parse HP and speaker DACs even for multi connections for ALC662
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:09 +02:00
Takashi Iwai
8e89995c58 Merge branch 'fix/hda' into topic/hda 2011-07-07 09:28:47 +02:00
Takashi Iwai
9c7a083d94 ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume.  Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:25:54 +02:00
Kailang Yang
b68785714b ALSA: hda - Add Realtek ALC269VC codec support
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.

In addition, added some aliases:
 - Add ALC269VB alias name ALC277
 - Add ALC269VC alias name ALC259 ALC281X
 - Add ALC269VC for Lenovo device 0x21f3 name ALC3202

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-06 09:53:28 +02:00
Stephen Warren
774fec338b ASoC: Tegra: Implement SPDIF CPU DAI
This is a minimal driver for the Tegra SPDIF controller.

In hardware, the SPDIF output signal is always routed to any active HDMI
display controllers, and may also be routed to external pins on Tegra
using the pinmux.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 12:20:56 -07:00
Liam Girdwood
a82ce2ae0d ASoC: core - Add platform IO tracing
Trace platform IO just like CODEC IO.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 11:08:10 -07:00
Liam Girdwood
cb2cf612fb ASoC: core - Add convenience register for platform kcontrol and DAPM
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 11:07:41 -07:00
Liam Girdwood
b795064137 ASoC: core - Add platform widget IO
Allow platform driver widgets to perform any IO required for DAPM.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 11:07:39 -07:00
Liam Girdwood
a491a5c84f ASoC: core - Add API call to register platform kcontrols.
In preparation for Dynamic PCM (AKA DSP) support.

Allow platform drivers to register kcontrols.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 11:07:34 -07:00
Mark Brown
8a27bd9a33 ASoC: Manage WM8731 ACTIVE bit as a supply widget
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-05 11:07:33 -07:00
Mark Brown
4c7c5374ce ASoC: Manage WM8731 ACTIVE bit as a supply widget
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-05 11:00:21 -07:00
Takashi Iwai
873bd4cb4f ASoC: Don't set invalid name string to snd_card->driver field
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters.  The commit
2b39535b9e changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.

A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.

Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-05 14:39:27 +02:00
Takashi Iwai
f187700c2d Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-07-05 08:20:19 +02:00
Takashi Iwai
8d9afa08fe Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2011-07-05 08:20:00 +02:00
Takashi Iwai
56aa533910 Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2011-07-05 07:33:23 +02:00
Takashi Iwai
63bc975016 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc 2011-07-05 07:33:06 +02:00
Liam Girdwood
f1442bc1e9 ASoC: core - Add platform read and write.
In preparation for ASoC Dynamic PCM (AKA DSP) support.

Allow platform driver to perform IO. Intended for platform DAPM.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 12:41:07 -07:00
Jarkko Nikula
404b566569 ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2011-07-04 19:54:38 +01:00
Axel Castaneda Gonzalez
1fbe99529d ASoC: twl6040: Configure ramp step based on platform
Enable ramp down/up step to be configured based on
platform.

Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:36:29 +03:00
Liam Girdwood
f7026c9996 ASoC: twl6040: set default constraints.
Set default sysclk constraints to high performance mode.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:36:28 +03:00
Misael Lopez Cruz
6bba63b68d ASoC: twl6040: Remove pll and headset mode dependency
Remove dependency between pll (hppll, lppll) and headset power
mode (low-power, high-performance), as headset power mode can
be used with any pll.

A new control is created to allow headset power mode configuration
from userspace. Changing headset power mode during earpiece related
usecases is not propagated down to the codec as earpiece requires
HS DAC in HP mode.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:36:27 +03:00
Liam Girdwood
e17e4ab801 ASoC: twl6040: Support other sample rates in constraints.
Add other supported sample rates to LP and HP modes.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:36:26 +03:00
Liam Girdwood
6510bdc3f4 ASoC: twl6040: add all ABE DAIs
Add all DAIs to fully support OMAP4 ABE.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:36:25 +03:00
Misael Lopez Cruz
fb34d3d505 ASoC: twl6040: Convert into TWL6040 MFD child
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 19:35:06 +03:00
Mark Brown
469bb638dc Merge branch 'for-3.0' into for-3.1 2011-07-04 08:54:40 -07:00
Mark Brown
8e9ddf811b ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
This delay is very conservative.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-07-04 08:51:44 -07:00
Stephen Warren
b5f9cfed12 ASoC: Tegra: I2S: s/clk_get_sys/clk_get/
The clock needed by the I2S driver is associated with the I2S device name
in the standard fashion. Hence, use clk_get(dev) instead of clk_get_sys(clk_name).

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 08:49:24 -07:00
Stephen Warren
713d136978 ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.

Luckily, the dropped register writes historically had no effect:

TEGRA_I2S_TIMING: The value we wrote was the reset default.

TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.

However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-04 08:49:05 -07:00
Peter Ujfalusi
4ae6df5e10 MFD: twl4030-audio: Rename platform data
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver

To avoid breakage, change all depending drivers, files
to use the new types.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
2011-07-04 18:44:02 +03:00
Peter Ujfalusi
57fe7251f5 MFD: twl4030-codec -> twl4030-audio: Rename the driver
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
2011-07-04 18:43:56 +03:00
Takashi Iwai
bac4b92cf7 ALSA: hda - Don't add aa-mix for VIA surrounds
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals.  For avoiding this, the aa-mix routes have to be disabled
for surround paths.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 17:37:57 +02:00
Takashi Iwai
18bd2c44b9 ALSA: hda - Create HP-vol control properly for VIA codecs
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 15:55:44 +02:00
Takashi Iwai
de6c74f3e3 ALSA: hda - Define some constants in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 14:53:30 +02:00
Lydia Wang
b89596a160 ALSA: hda - Fix invalid multi-channel amplifiers for VT1718S
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;

But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 14:53:25 +02:00
Lydia Wang
c4394f5b80 ALSA: hda - Fix issue that front can't output sound for VT1718S
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 14:33:23 +02:00
Liam Girdwood
956245e9cd ASoC: core - Make platform probe more like codec probe.
In preparation for ASoC dynamic PCM support (AKA ASoC DSP)

Platform will also support DAPM so separate out the probe function
to simplify the code (just like the codec probe).

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-02 11:50:16 -07:00
Lydia Wang
e5e1468140 ALSA: hda - Fix the silent front with independent-HP for VIA codecs
Unmute DAC on front speaker path when Independent HP is enabled.

When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().

If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-01 08:33:06 +02:00
Mark Brown
67d0c479d9 ASoC: Improve error reporting in Speyside WM8962 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-30 13:17:49 -07:00
Takashi Iwai
350434ee53 ALSA: hda - Fix missing initialization in alc662 auto-parser
A missing initialization resulted in wrong DAC assignments in
ALC662 (and other) auto-parsers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-30 21:29:12 +02:00
Takashi Iwai
2525050518 ALSA: hda - Re-implementation of VIA Independent-HP sharing with side stream
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.

When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.

One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-30 17:24:47 +02:00
Takashi Iwai
286bed0f0c ALSA: hdspm - Fix compile warnings with PPC
The char can be unsigned on some architectures.  Since the code checks
the negative values, they should be declared as signed char explicitly.

  sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
  sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-30 12:45:36 +02:00
Takashi Iwai
71276410e1 ALSA: cs5535 - Fix invalid big-endian conversions
Fix the wrongly converted short values:
  sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
  sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-30 12:35:45 +02:00
Mark Brown
57cc2432e1 Merge branch 'for-3.0' into for-3.1 2011-06-29 09:49:04 -07:00
Mark Brown
4e8e78e37c ASoC: Change WM9081 speaker output enable to _OUT_DRV
More for neatness than any actual performance improvement.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-29 09:48:55 -07:00
Mark Brown
d5b040c92d ASoC: Correct left/right swap in wm_hubs DC offset correction
It was consistently wrong for everything except WM8993 so should be no
functional change.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.comm>
2011-06-29 09:48:36 -07:00
Mark Brown
e999dc5040 ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.

Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-29 09:47:53 -07:00
Wu Fengguang
f5b2d0ef63 ALSA: HDMI - fix ELD monitor name length
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.

This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.

The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:48:24 +02:00
Lydia Wang
e322a36d39 ALSA: hda - Fix jack-detection on non-VT1708 VIA codecs
Move codec init verb which is only applicatable for VT1708.

I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:02:46 +02:00
Takashi Iwai
94230c11da ALSA: hda - Fix unused variable warning
sound/pci/hda/patch_cmedia.c: In function ‘cmi9880_fill_multi_init’:
sound/pci/hda/patch_cmedia.c:401:15: warning: unused variable ‘len’

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:02:33 +02:00
Takashi Iwai
c82693db52 ALSA: hda - Enable auto-parser as default for Conexant codecs
Let's use auto-parser as default now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:02:21 +02:00
Takashi Iwai
c2549312d2 Merge branch 'fix/hda' into topic/hda 2011-06-29 08:02:09 +02:00
Takashi Iwai
8d087c7600 ALSA: hda - Create snd_hda_get_conn_index() helper function
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget.  Replaced the similar codes used in
several codec-drivers with this common helper.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:01:46 +02:00
Lydia Wang
63f10d2ca7 ALSA: hda - Fix unsol event initializations for VIA codecs
Fix a issue to enable unsolicited response to line-out pins.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:01:23 +02:00
Lars-Peter Clausen
aef05294df ASoC: Blackfin: Add machine driver for EVAL-ADAV80X boards
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-28 17:12:44 +01:00
Lars-Peter Clausen
cc52688a08 ASoC: Add ADAV80x codec driver
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-28 17:11:20 +01:00
Hans-Christian Egtvedt
0cfae7c937 ALSA: atmel - update author email for ABDAC, AC97C and AT73C213
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.

Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 16:56:07 +02:00
David Henningsson
9966db22ca ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TG
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 14:09:57 +02:00
David Henningsson
f0ca89b031 ALSA: HDA: Add a new Conexant codec ID (506c)
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 14:09:41 +02:00
Takashi Iwai
ff2b7e2a3f ALSA: hda - Fix warnings with CONFIG_SND_POWER_SAVE=n
Use static inline for dummy function to fix the warnings like below
  sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
  sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
  sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
  sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 08:59:30 +02:00
Stephen Rothwell
880a050f4a ALSA: hda - remove SND_HDA_POWER_SAVE protection of struct hda_loopback_check
to fix build problems when it is disabled.

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 08:55:56 +02:00
Takashi Iwai
4f574b7b1a ALSA: hda - More volume-init fixes for ALC267 codec
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.

Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 16:17:07 +02:00
Takashi Iwai
7ec9c6ccc6 ALSA: hda - Fix volume-init for ALC259 with invalid widget caps
ALC259 seems to provide an invalid widget capability for the input-src
selector widget.  The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too.  For fixing this, check the amp capability and handle
the connection selection individually.

Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 15:53:38 +02:00
Takashi Iwai
050ea75317 ALSA: hda - Fix volume-init of ALC299 & co
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is.  Fixed by checking the amp caps and
picking up the right widget for initialization.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 15:48:17 +02:00
Takashi Iwai
39fa84e94a ALSA: hda - Simplify EAPD control in patch_realtek.c
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.

This will also avoid the forgotten entries of newly added codec ids
in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 15:28:57 +02:00
Takashi Iwai
6d86b4fb40 ALSA: hda - Fix auto-init of output volumes of Realtek codecs
Fix the regression introduced by the commit
1f0f4b8036
  ALSA: hda - Reduce static init verbs for Realtek auto-parsers

The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).

More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 15:07:28 +02:00
Takashi Iwai
00c6850dde Merge branch 'topic/via-cleanup' into topic/hda 2011-06-27 14:32:50 +02:00
Takashi Iwai
3af9ee6b83 ALSA: hda - Check hard-wired DACs at first for ALC662 & co
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.

Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 12:34:01 +02:00
Takashi Iwai
cb053a8265 ALSA: hda - Call proper DAC-filler function for Realtek auto-parser
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way.  This isn't pretty suitable for other codecs, of course.

Now we call it with fill_dac function pointer so that the proper
function can be called at that point.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 11:32:07 +02:00
Takashi Iwai
1f0f4b8036 ALSA: hda - Reduce static init verbs for Realtek auto-parsers
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-27 10:52:59 +02:00
Sangbeom Kim
766705eef5 ASoC: SAMSUNG: Add idma related register definition
This patch add idma related register definitions to support idma.

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-26 12:05:23 +01:00
Sangbeom Kim
172a453dcf ASoC: SAMSUNG: Move I2S common register definition
I2S registers can be used for control idma.
Previously, register is defined in i2s.c.
For sharing the registers, It is moved to i2s-regs.h

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-26 12:05:17 +01:00
Sangbeom Kim
d2ec3ababa ASoC: SAMSUNG: Add WM8994 PCM Machine driver
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-26 12:05:12 +01:00
Takashi Iwai
dce2079b89 ALSA: hda - Add snd_hda_get_conn_list() helper function
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-25 09:19:49 +02:00
Tim Blechmann
80b52490cd ALSA: lx6464es - include mac address in device name
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-25 09:15:31 +02:00
Markus Bollinger
1470579913 ALSA: lola - Fix for Lola280 board
- add/fix comments and debug messages
- fix incomplete matrix init
- comment out creation of buggy lola_dest_gain_mixer controls
- minor optimisations

Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 12:54:43 +02:00
Takashi Iwai
e093811857 Merge branch 'fix/asoc' into for-linus 2011-06-24 12:36:25 +02:00
Jesper Juhl
16866741bd ALSA: Remove unneeded version.h includes from sound/
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 11:28:17 +02:00
Takashi Iwai
2e925ddeb9 ALSA: hda - Use alc_get_pfx_name() for all Realtek codecs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 11:27:22 +02:00
Joe Perches
7c9d440e90 treewide: transciever/transceiver spelling fixes
Just tyops.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2011-06-24 11:20:14 +02:00
Joe Perches
dbc6221be7 treewide: Fix recieve/receive typos
Just spelling fixes.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2011-06-24 11:18:57 +02:00
Takashi Iwai
6843ca16f5 ALSA: hda - Clean up multi-channel mixer name assignment in patch_realtek.c
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.

Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 11:06:17 +02:00
Takashi Iwai
1af7c5f0d4 ALSA: hda - Add a workaround for invalid line-out setups
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out.  In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 10:43:03 +02:00
Takashi Iwai
f9a09e003f Merge branch 'fix/hda' into topic/hda 2011-06-24 10:36:25 +02:00
Takashi Iwai
3fccdfd891 ALSA: hda - Allow multi-io with HP output for ALC662 & co
Even if the machine has no line-out but only HP-out, try to detect the
multi-io.  It'll allow more possibilities for 5.1 outputs on laptops.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-24 10:35:05 +02:00
Mark Brown
1ae4e9b996 Merge branch 'for-3.0' into for-3.1 2011-06-23 20:10:28 +01:00
Daniel Mack
53dea36c70 ASoC: pxa-ssp: Correct check for stream presence
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-23 20:09:56 +01:00
Arnaud Patard (Rtp)
96dcabb99b ASoC: imx: add missing module informations
- add some modules aliases
- add module license to avoid tainted kernel when loading the imx-pcm-audio
  driver

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-23 20:08:50 +01:00
Arnaud Patard (Rtp)
f6d96e0da1 ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry
SND_MXC_SOC_SSI looks to be unused, so kill it.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-23 20:08:30 +01:00
Mark Brown
df1553c8de ASoC: Add basic WM8918 support
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-23 20:05:38 +01:00
Mark Brown
780b75b4a0 ASoC: Allow suppression of series updates on wm_hubs devices
Some devices do not support manual updates of the DC servo.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-23 20:05:31 +01:00
Mark Brown
f9acf9fe5b ASoC: Trigger wm_hubs series update startup off a separate flag
Allowing the two to be used independently.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-23 20:05:23 +01:00
David Henningsson
d2a19da79d ALSA: HDA: Pinfix quirk for HP Z200 Workstation
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.

BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-23 09:25:20 +02:00
Johannes Stezenbach
ec3ea54c6c ASoC: add WM8782 ADC Codec Driver
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-22 19:11:02 +01:00
Johannes Stezenbach
c034abf6e5 ASoC: add STA32X codec driver
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-22 19:10:59 +01:00
Takashi Iwai
a86a88eaf6 ALSA: hda - Implement dynamic-ADC switching for VIA codecs
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs.  These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.

This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case.  When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs.  The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-22 15:23:25 +02:00
Taylor Hutt
bab3b59d53 ASoC: codecs: Max98095: Fix logging of hardware revision.
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.

Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().

Signed-off-by: Taylor Hutt <thutt@chromium.org>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-22 00:27:39 +01:00
Takashi Iwai
f2b1c9f031 ALSA: hda - Auto-mute smart51 surround pins for VIA codecs
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out.  Also this patch includes minor
clean-ups.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 16:52:39 +02:00
Takashi Iwai
ddd304d8be ALSA: hda - Remove redundant VT1709 and VT1708B codes
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers.  They have no difference now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 16:33:55 +02:00
Takashi Iwai
09a9ad69a5 ALSA: hda - VT1708 independent HP routing fix
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 16:02:32 +02:00
Takashi Iwai
a934d5a983 ALSA: hda - Fix surround-volume parsing for VT1708B codecs
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead.  Fix the parser to
check these selector widgets for the possible mixer controls as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 14:22:14 +02:00
Takashi Iwai
1e11cae143 ALSA: hda - Fix the check of loopback-mixer element index in patch_via.c
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic.  Now check the label properly for avoid duplication.

Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 12:59:24 +02:00
Takashi Iwai
0f98c24b80 ALSA: hda - Assign smart51 only in the same stack for VIA codecs
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other.  Also, prefer line-in as the surround
to mic-in.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 12:51:33 +02:00
Takashi Iwai
8df2a3129d ALSA: hda - Fix re-routing of HP-independent mode in patch_via.c
Re-route the whole output path when HP-independent mode is changed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 12:20:13 +02:00
Lydia Wang
a00a5fad9d ALSA: hda - Fix creations of playback volume controls in patch_via.c
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 10:24:18 +02:00
Takashi Iwai
8e3679dca2 ALSA: hda - Revisit output_path parsing in patch_via.c
Change the order of the output-path list in a way from the DAC to the
target pin.  Also now the list include the target pin, too.

Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 09:01:36 +02:00
Takashi Iwai
30f7c5d491 ALSA: hda - Use xxx Boost Volume for VIA
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 08:37:41 +02:00
Takashi Iwai
efb9f469b6 ALSA: hda - Fix a compile error in patch_ca0132.c for the recent SPDIF change
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:44:51 +02:00
Takashi Iwai
216c7a0f22 Merge branch 'fix/hda' into topic/via-cleanup
Conflicts:
	sound/pci/hda/patch_via.c
2011-06-21 07:34:45 +02:00
Ian Minett
95c6e9cb77 ALSA: hda - Add Creative CA0132 HDA codec support
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.

This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
   i - Mic-in
   ii- Line-in
* mixer device

Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:31:25 +02:00
Lydia Wang
e905a83acd ALSA: VIA HDA: Create a master amplifier control for VT1718S.
Create a master volume and mute control of playback for VT1718S.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:24:56 +02:00
Lydia Wang
ba31a60d0f ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.
When switch HP independent mode, mute/unmute connctions of mixer  which is
connected to headphone for VT1718S.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:23:19 +02:00
Lydia Wang
42467b32ce ALSA: VIA HDA: Modify initial verbs list for VT1718S.
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:22:57 +02:00
Tony Vroon
c933790614 ALSA: hda - Remove ALC268 model override for CPR2000
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.

Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:18:36 +02:00
Dan Carpenter
c20974090e ASoC: adau1701: signedness bug in adau1701_write()
"ret" is supposed to be signed here.  The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-20 18:46:33 +01:00
David Henningsson
6f2e810ad5 ALSA: HDA: Remove quirk for an HP device
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.

BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:32:50 +02:00
Takashi Iwai
ada509ec00 ALSA: hda - Simplify analog-low-current mode check for VIA codecs
Use the existing aa-loop list for simplifying the check for analog
low-current mode.  Also fix the stream count test for playback streams.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:16 +02:00
Takashi Iwai
47be05ce0a ALSA: hda - Remove NID_MAPPING hacks in patch_via.c
There is no longer virtual kmixer element for NID mapping.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:16 +02:00
Takashi Iwai
c619160787 ALSA: hda - Remove unused defines and struct fields in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:15 +02:00
Takashi Iwai
6aadf41d6b ALSA: hda - Name the primary out as Speaker when needed for VIA codecs
When the primary output is the speaker output, rather name it as
"Speaker".  This will be more intuitive.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:14 +02:00
Takashi Iwai
13af8e77ea ALSA: hda - Create loopback-list dynamically in patch_via.c
Create loopback list dynamically from the parsed input pins for VIA
codecs instead of the fixed arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:13 +02:00
Takashi Iwai
e3d7a1431f ALSA: hda - Fix smart51 handling again
Fix the broken detection of smart51 and its handling.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:12 +02:00
Takashi Iwai
370bafbdae ALSA: hda - Create virtual-master control for VIA codecs
Now let's add the missing Master control to VIA codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:12 +02:00
Takashi Iwai
4a918ffeaa ALSA: hda - Initialize unsol events dynamically in patch_via.c
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs.  Also, consolidate the unsol handlers for HP
and line-out mutes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:11 +02:00
Takashi Iwai
096a885494 ALSA: hda - Initialize input-path dynamically in patch_via.c
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:10 +02:00
Takashi Iwai
5d41762a21 ALSA: hda - Initialize output path dynamically in patch_via.c
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:09 +02:00
Takashi Iwai
0fe0adf82f ALSA: hda - Replace with standard consts in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:09 +02:00
Takashi Iwai
ece8d0431f ALSA: hda - Fix independent-HP handling in patch_via.c
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.

This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:08 +02:00
Takashi Iwai
12daef65fd ALSA: hda - Unify auto-parser in patch_via.c
Now all codecs use the same parser-path, so we can reduce into a single
auto-parser function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:07 +02:00
Takashi Iwai
7f0df88ce0 ALSA: hda - Return error for invalid setup for VIA
Instead of ignoring the invalid pin configuration, return the error.
This will avoid unexpected crash, anyway.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:06 +02:00
Takashi Iwai
d7a99cce55 ALSA: hda - Unify capture-mixer creations in patch_via.c
Create capture-related mixer elements dynamically from the parsed
ADCs and input-pins instead of fixed values for each codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:05 +02:00
Takashi Iwai
7eb56e84e6 ALSA: hda - Assign HP-independent PCM to individual stream
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM.  Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:05 +02:00
Takashi Iwai
9af7421091 ALSA: hda - Unify PCM assignments in patch_via.c
Unify PCM streams for all codecs by assigning the NID dynamically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:04 +02:00
Takashi Iwai
57307bf24a ALSA: hda - Don't create secondary substream when no independent-hp is used
For VIA codecs, we shouldn't create a substream for independent HP mode,
when no individual HP DAC is found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:03 +02:00
Takashi Iwai
f4a7828bc1 ALSA: hda - Re-implement smart51 detection for VIA codecs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:02 +02:00
Takashi Iwai
4a79616d07 ALSA: hda - Unify output-control parsing in patch_via.c
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.

The smart51 setup doesn't work with this change, and will be fixed
in the next commits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:01 +02:00
Takashi Iwai
620e2b28b7 ALSA: hda - Unify input-volume creations in patch_via.c
Now storing the analog-mixer widget in spec, we can simplify the rest
parts.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:01 +02:00
Takashi Iwai
64be285b66 ALSA: hda - Auto-mute all LO and speakers in patch_via.c
Muting all line-outs and/or speakers is more common in other drivers,
so we should follow it, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:00 +02:00
Takashi Iwai
3e0693e278 ALSA: hda - Change pin-ctl for auto-muting in patch_via.c
Mute the outputs via pin-controls instead of amps for the auto-mute
handling.  This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.

With this change, we can use vmaster for the master control easily now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:59 +02:00
Takashi Iwai
82673bc895 ALSA: hda - Generate PCM names dynamically in patch_via.c
This reduces lots of static strings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:58 +02:00
Takashi Iwai
291c9e33bf ALSA: hda - Refactor ctl array handling in patch_via.c
No functional change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:57 +02:00
Takashi Iwai
e06e5a2974 ALSA: hda - Defer mixer element creation to the right time in patch_via.c
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.

Also, fixed typos of vt1708_jack_detect in all places.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:56 +02:00
Takashi Iwai
a766d0d763 ALSA: hda - Fill ADCs dynamically for VIA codecs
Instead of giving the fixed ADC list, parse the widgets and fill in
ADCs dynamically.

Also, probe the stereo-mixer input more dynamically, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:55 +02:00
Takashi Iwai
24088a58d6 ALSA: hda - Add control to suppress the dynamic pin-power for VIA
Currently VIA driver controls the power-state of each pin per jack
detection.  But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.

For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:55 +02:00
Takashi Iwai
5f4b36d64d ALSA: hda - Remove superfluous NID_MAPPING use for smart51 mixer
Just a minor clean up; nid-mapping can be set directly to the smart51
mixer element.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:07:02 +02:00
Mark Brown
29376bc7e2 ASoC: Fix DAPM sequence run for per-widget I/O methods
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.

Fix originally suggested by Liam on IM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-20 11:27:10 +01:00
Mark Brown
ee8c7e9744 ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others.  There's a pending patch to make the directory build as standard
but it's not merged yet.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-18 15:31:38 +01:00
David Henningsson
b13e552d37 ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 18:48:23 +02:00
Grant Likely
f8db4cc4f2 Merge branch 'spi/merge' into spi/next 2011-06-17 08:32:26 -06:00
Takashi Iwai
3409fcd1f7 Merge branch 'fix/hda' into topic/misc 2011-06-17 14:54:47 +02:00
Takashi Iwai
ad2409413d ALSA: hda - Fix no NID error with VIA codecs
The via driver spews warnigs like
	hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 14:23:46 +02:00
Mike Frysinger
4d1e46b7ef ASoC: Blackfin: allow SPI for SSM2602 parts
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-17 10:40:16 +01:00
Clemens Ladisch
cf6f1ff17f ALSA: isight: adjust for new queueing API
Since commit 13882a82ee (optimize iso queueing by setting
wake only after the last packet), drivers are required to call
fw_iso_context_queue_flush() after queueing a batch of packets.
The missing call would have an effect only if the controller
queue underruns, but then the DMA would stop completely.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 11:37:29 +02:00
Clemens Ladisch
f4b1e98aa9 ALSA: firewire-speakers, oxygen, ua101: allow > 10 s periods
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 11:36:30 +02:00
Torsten Schenk
0ec5258d68 ALSA: 6fire - Fix signedness bug
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-16 21:31:33 +02:00
Barry Song
5d0e7f6170 ASoC: AD1836: rename suspend/resume funcs
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Scott Jiang <scott.jiang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:42:10 +01:00
Mike Frysinger
0679059a41 ASoC: AD1836: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD.  This was incorrectly changed during the multi-component updated.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:42:01 +01:00
Mike Frysinger
d4d80f5e46 ASoC: AD1836: fix intermixed tab/space indentation
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:53 +01:00
Mike Frysinger
15e8705129 ASoC: AD1836: drop unnecessary spi register check
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us.  So drop the redundant checking/message.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:46 +01:00
Mike Frysinger
42f32c5591 ASoC: AD1836: clean up comment headers
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:38 +01:00
Lars-Peter Clausen
53a93d58fb ASoC: Blackfin: Add bf5xx-adau1701 machine driver
Add a machine driver to support the ADAU1701 SigmaDSP processors on
Analog Devices BF5XX evaluation boards.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:24:25 +01:00
Lars-Peter Clausen
631ed8a213 ASoC: Add ADAU1701 codec driver
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:24:17 +01:00
Mark Brown
b83e60c000 ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-16 11:24:16 +01:00
Ryan Mallon
1c5454eed8 Change Ryan Mallon's email address across the kernel
I no longer work at Bluewater Systems. Update my email address accordingly. I
have deleted my email address from C files rather than change it. This
was suggested by several people, since the commit from my new email
address will cause scripts/get_maintainer.pl to function properly. I
have not added the .mailmap entry as suggested by Joe because I think
it is no longer necessary if I touch all the files which had my name
in them.

Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Cc: Andre Renaud <andre@bluewatersys.com>
Cc: H Hartley Sweeten <hsweeten@visionengravers.com>
Cc: Russell King <linux@arm.linux.org.uk>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Andrew Victor <avictor.za@gmail.com>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Anton Vorontsov <cbou@mail.ru>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Jesper Juhl <jj@chaosbits.net>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: trivial@kernel.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2011-06-16 11:22:05 +02:00
Harry Butterworth
030aba53ea ALSA: ctxfi: Change PLL initialization code
This is a reworked patch from Creative to change the PLL code to address
unreliable 44100Hz initialization.

Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-16 07:28:57 +02:00
Takashi Iwai
e72888e91c ALSA: lola - Fix section mismatch
Add missing __devinit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-15 15:14:49 +02:00
Liam Girdwood
0445bdf4ae ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
This time with soc_widget_update_bits reflecting recent soc_update_bits changes.

Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP
devices have mixer components that can alter power usage and hence require full
DAPM support.

This provides a generic widget IO operation wrapper in preparation for
future patches that implement platform driver DAPM.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-14 14:25:32 +01:00