Commit graph

5073 commits

Author SHA1 Message Date
Takashi Iwai
afad17c0ae Merge branch 'topic/usb-audio' into for-linus 2009-03-24 00:36:22 +01:00
Takashi Iwai
d080732334 Merge branch 'topic/sscape-fix' into for-linus 2009-03-24 00:36:21 +01:00
Takashi Iwai
d7b6df5d1a Merge branch 'topic/powermac' into for-linus 2009-03-24 00:36:20 +01:00
Takashi Iwai
7df4f69435 Merge branch 'topic/pcxhr' into for-linus 2009-03-24 00:36:19 +01:00
Takashi Iwai
b54fc8dd2c Merge branch 'topic/oxygen' into for-linus 2009-03-24 00:36:17 +01:00
Takashi Iwai
9fb5430c3d Merge branch 'topic/oss-fix' into for-linus 2009-03-24 00:36:16 +01:00
Takashi Iwai
4c5ecb7e59 Merge branch 'topic/msnd' into for-linus 2009-03-24 00:36:15 +01:00
Takashi Iwai
fa15fdeffa Merge branch 'topic/isa-misc' into for-linus 2009-03-24 00:36:13 +01:00
Takashi Iwai
843ad02fa4 Merge branch 'topic/intel8x0' into for-linus 2009-03-24 00:36:13 +01:00
Takashi Iwai
60c4e7c120 Merge branch 'topic/ice' into for-linus 2009-03-24 00:36:12 +01:00
Takashi Iwai
593aff6c50 Merge branch 'topic/hdsp' into for-linus 2009-03-24 00:36:10 +01:00
Takashi Iwai
e7bfbb0215 Merge branch 'topic/hda' into for-linus 2009-03-24 00:36:09 +01:00
Takashi Iwai
fe506d6bc5 Merge branch 'topic/emu10k1' into for-linus 2009-03-24 00:36:08 +01:00
Takashi Iwai
c9294e4b37 Merge branch 'topic/echoaudio' into for-linus 2009-03-24 00:36:07 +01:00
Takashi Iwai
ae02cde7e9 Merge branch 'topic/drop-l3' into for-linus 2009-03-24 00:36:05 +01:00
Takashi Iwai
a3c6048dcf Merge branch 'topic/cs423x-merge' into for-linus 2009-03-24 00:35:59 +01:00
Takashi Iwai
87cd9d7c85 Merge branch 'topic/ca0106' into for-linus 2009-03-24 00:35:58 +01:00
Takashi Iwai
158c1529fe Merge branch 'topic/atmel' into for-linus 2009-03-24 00:35:56 +01:00
Takashi Iwai
b5c784894c Merge branch 'topic/asoc' into for-linus 2009-03-24 00:35:53 +01:00
Takashi Iwai
ff4fc3656e Merge branch 'topic/aoa' into for-linus 2009-03-24 00:35:51 +01:00
Takashi Iwai
e0d2054fd3 Merge branch 'topic/misc' into for-linus 2009-03-24 00:35:50 +01:00
Takashi Iwai
d807500a24 Merge branch 'topic/pcm-cleanup' into for-linus 2009-03-24 00:35:49 +01:00
Takashi Iwai
ec6659c389 Merge branch 'topic/vmaster-update' into for-linus 2009-03-24 00:35:47 +01:00
Takashi Iwai
c944a93df0 Merge branch 'topic/rawmidi-fix' into for-linus 2009-03-24 00:35:46 +01:00
Takashi Iwai
65b3864b85 Merge branch 'topic/ctl-list-cleanup' into for-linus 2009-03-24 00:35:45 +01:00
Takashi Iwai
bafdb7278c Merge branch 'topic/quirk-cleanup' into for-linus 2009-03-24 00:35:44 +01:00
Takashi Iwai
5b56eec774 Merge branch 'topic/jack' into for-linus 2009-03-24 00:35:43 +01:00
Takashi Iwai
c2f43981e5 Merge branch 'topic/hwdep-cleanup' into for-linus 2009-03-24 00:35:41 +01:00
Takashi Iwai
dec14f8c0e Merge branch 'topic/snd_card_new-err' into for-linus 2009-03-24 00:35:35 +01:00
Takashi Iwai
9b6682ff4c ALSA: hda - Add quirk for Acer Ferrari 5000
Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).

Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 22:50:52 +01:00
Takashi Iwai
14bafe3278 ALSA: hda - Use cached calls to get widget caps and pin caps
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.

hda_proc.c is still using the direct verbs to get raw values as
much as possible.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 16:36:55 +01:00
Takashi Iwai
a23b688f4d ALSA: hda - Don't create empty/single-item input source
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 15:24:15 +01:00
Takashi Iwai
e82c025b50 ALSA: hda - Fix the wrong pin-cap check in patch_realtek.c
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change...  Simply a thinko.

Also, add the similar checks to all places that put output-amp mutes
in the initialization.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 15:20:53 +01:00
Takashi Iwai
1327a32b87 ALSA: hda - Cache pin-cap values
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 13:08:33 +01:00
Takashi Iwai
52ca15b7c0 ALSA: hda - Avoid output amp manipulation to digital mic pins
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 12:51:55 +01:00
Pascal de Bruijn
234b4346a0 ALSA: hda - Add function id to proc output
This patch does two things:
 Output Intel HDA Function Id in /proc/asound/cardX/codec#X
 Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros

Before:
 Vendor Id: 0x11d41884
 Subsystem Id: 0x103c281a
 Revision Id: 0x100100

After:
 Function Id: 0x1
 Vendor Id: 0x11d41884
 Subsystem Id: 0x103c281a
 Revision Id: 0x0100100

As report on the Kernel Bugzilla #12888

Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-23 11:55:36 +01:00
Takashi Iwai
c9840cf4cf Merge branch 'topic/hda-optimize' into topic/hda 2009-03-20 16:33:30 +01:00
Takashi Iwai
8b22d943c3 ALSA: pcm - Safer boundary checks
Make the boundary checks a bit safer.
These caese are rare or theoretically won't happen, but nothing
bad to keep the checks safer...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-20 16:26:15 +01:00
Takashi Iwai
2d864c499a ALSA: hda - Detect digital-mic inputs on ALC663 / ALC272
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode.  The automatic mic switch via plugging
isn't implemented yet, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-20 12:52:47 +01:00
Wolfram Sang
c468ac29e6 ALSA: sound/ali5451: typo: s/resouces/resources/
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-20 12:47:50 +01:00
Takashi Iwai
07a1e81355 ALSA: hda - Don't show the current connection for power widgets
The power-widgets have no connection selection, so skip the check
in proc output, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 17:08:19 +01:00
Takashi Iwai
1f2186951e ALSA: Fix wrong pointer to dev_err() in arm/pxa2xx-ac97-lib.c
Fix the wrong device pointer passed to dev_err().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 14:16:19 +01:00
Lopez Cruz, Misael
632087748c ASoC: Declare Headset as Mic and Headphone widgets for SDP3430
Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.

Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:

- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-19 11:56:16 +00:00
Jarkko Nikula
f8d5fc924b ASoC: OMAP: N810: Add more jack functions
Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.

Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-19 11:56:16 +00:00
Jarkko Nikula
13b9d2ab59 ASoC: OMAP: N810: Mark not connected input pins
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-19 11:56:15 +00:00
Mark Brown
e8523b641c ASoC: Add FLL support for WM8400
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-19 11:56:11 +00:00
Takashi Iwai
1dddab400b ALSA: hda - Don't reset stream at each prepare callback
Don't reset the stream at each prepare callback but do it only once
after the open.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 12:54:23 +01:00
Takashi Iwai
97b71c94d6 ALSA: hda - Don't reset BDL unnecessarily
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.

This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 12:53:58 +01:00
Takashi Iwai
ded652f702 ALSA: pcm - Fix delta calculation at boundary overlap
When the hw_ptr_interrupt reaches the boundary, it must check whether
the hw_base was already lapped and corret the delta value appropriately.

Also, rebasing the hw_ptr needs a correction because buffer_size isn't
always aligned to period_size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 10:08:49 +01:00
Takashi Iwai
5f513e1197 ALSA: pcm - Reset invalid position even without debug option
Always reset the invalind hw_ptr position returned by the pointer
callback.  The behavior should be consitent independently from the
debug option.

Also, add the printk_ratelimit() check to avoid flooding debug
prints.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 10:01:47 +01:00
Takashi Iwai
98204646f2 ALSA: pcm - avoid unnecessary inline
Remove unnecessary explicit inlininig of internal functions.
Let compiler optimize.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 09:59:21 +01:00
Takashi Iwai
cad377acf3 ALSA: pcm - Fix a typo in error messages
Fix a typo in error messages; forgotten after a copy&paste error.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 09:57:45 +01:00
Giuliano Pochini
a2328d0249 ALSA: Echoaudio: add support for Indigo express cards
This patch adds support for IndigoIOx and IndigoDJx.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-19 08:17:57 +01:00
Mark Brown
24a51029fc ASoC: Add separate AVDD for WM8400
There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-18 18:31:54 +00:00
Mark Brown
e3598f6e42 ASoC: Further optimise WM8400 bias configuration sequence
The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-18 18:31:53 +00:00
Daniel Mack
28514fe5bb ALSA: snd-usb-caiaq: bump version number
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 11:31:26 +01:00
Daniel Mack
9311c9b4f1 ALSA: snd-usb-caiaq: drop bogus iso packets
Drop inbound packets that are smaller than expected. This has been
observed at the very beginning of the streaming transaction.

And when the hardware is in panic mode (which can only very rarely
happen in case of massive EMI chaos), mute the input channels.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Tested-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 11:31:08 +01:00
Daniel Mack
1313e70414 ALSA: snd-usb-caiaq: only warn once on streaming errors
Limit the number of printed warnings to one in case of streaming errors.
printk() happens to be expensive, especially in code called as often as
here.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 11:27:51 +01:00
Takashi Iwai
f1aa298679 Merge branch 'fix/opl3sa2-suspend' into for-linus 2009-03-18 08:04:36 +01:00
Takashi Iwai
a232ee66e0 Merge branch 'fix/hda' into for-linus 2009-03-18 08:04:16 +01:00
Takashi Iwai
6af845e4eb ALSA: Fix vunmap and free order in snd_free_sgbuf_pages()
In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG
pages, and it causes errors on Xen as Xen manages the pages
differently.  Although no significant errors have been reported on
the actual hardware, this order should be fixed other way round,
first vunmap() then free pages.

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

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

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

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

Have tested two usb audio mic devices.

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

Also, clean a bit the code.

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

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

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

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

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

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

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

     return fasync_helper(...);

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Also, keep power-up during hwdep reconfiguration.

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

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

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

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

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

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

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

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

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

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

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

There's no such (standard) clock interface.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-02 10:35:29 +01:00