Commit graph

7000 commits

Author SHA1 Message Date
Takashi Iwai
f1cf9a666d Merge branch 'topic/hda' into for-linus 2010-03-08 09:35:43 +01:00
Wu Fengguang
2abbf4391f ALSA: hdmi - show debug message on changing audio infoframe
Also change printk level for the two others.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-08 08:21:25 +01:00
Wu Fengguang
079d88ccc3 ALSA: hdmi - merge common code for intelhdmi and nvhdmi
Create patch_hdmi.c to hold common code from intelhdmi and nvhdmi.

For now the patch_hdmi.c file is simply included by patch_intelhdmi.c
and patch_nvhdmi.c, and does not represent a real codec.

There are no behavior changes to intelhdmi. However nvhdmi made several
changes when copying code out of intelhdmi, which are all reverted in
this patch. Wei Ni confirmed that the reverted code actually works fine.

Tested-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-08 08:21:08 +01:00
Michele Ballabio
4193d13b2c ALSA: hda - Add ASRock mobo to MSI blacklist
This avoids a lockup at boot.

Signed-off-by: Michele Ballabio <barra_cuda@katamail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-07 09:29:39 +01:00
Takashi Iwai
7484399fe2 Merge branch 'fix/hda' into topic/hda 2010-03-07 09:29:29 +01:00
Frederik Deweerdt
d2db09b87e ALSA: hda: uninitialized variable fix
Commit eaa9b3a748 introduced the following
uninitialized warning:

	sound/pci/hda/patch_realtek.c: In function 'set_capture_mixer':
	sound/pci/hda/patch_realtek.c:4928: warning: 'pin' is used uninitialized in this function
	sound/pci/hda/patch_realtek.c:4918: note: 'pin' was declared here

It appears indeed that 'pin' needs to be initialized to 0.

Signed-off-by: Frederik Deweerdt <frederik.deweerdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-05 16:40:26 +01:00
Daniel T Chen
0321b69569 ALSA: hda: Use LPIB for a Biostar Microtech board
BugLink: https://launchpad.net/bugs/523953

The OR has verified that position_fix=1 is necessary to work around
errors on his machine.

Reported-by: MMarking
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-05 16:06:01 +01:00
Daniel T Chen
9919c7619c ALSA: hda: Use LPIB for Dell Latitude 131L
BugLink: https://launchpad.net/bugs/530346

The OR has verified that position_fix=1 is necessary to work around
errors on his machine.

Reported-by: Tom Louwrier
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-04 16:32:01 +01:00
Takashi Iwai
dd74b46535 ALSA: hda - Build hda_eld into snd-hda-codec module
Now two modules require hda_eld.o, so we need to put it to the common
place instead of building into two individual modules.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-04 16:05:24 +01:00
Wei Ni
25045705d4 ALSA: hda - Support NVIDIA MCP89 and GT21x hdmi audio
Support nvidia MCP89 and GT21x 8ch hdmi audio.
Add some eld support.

Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-04 15:54:12 +01:00
Wei Ni
7445dfc159 ALSA: hda - Support max codecs to 8 for nvidia hda controller
Support max codecs to 8 for nvidia hda controller.
Change AZX_MAX_CODECS to 8, and add
"#define AZX_DEFAULT_CODECS 4" for default driver.
Set azx_max_codecs to 8 for nvidia controller.

Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-04 15:53:56 +01:00
Norberto Lopes
28aedaf7bf ALSA: sound/pci/hda/hda_codec.c: various coding style fixes
Signed-off-by: Norberto Lopes <nlopes.ml@gmail.com>
Acked-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-02 11:21:18 +01:00
Takashi Iwai
20645d70bd ALSA: hda - Add missing hp_pins definitions for ALC269 quirks
In 2.6.33 ACL269 unsol event handler was changed to look up the pre-defined
pins, but the headphone pins aren't defined properly in each quirk.
This patch adds the missing definitions, and fixes the speaker auto-mute
regression on some ASUS (and possibly other) laptops.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2010-03-02 11:14:01 +01:00
Takashi Iwai
6679ee1870 Merge branch 'topic/asoc' into for-linus 2010-03-01 12:38:59 +01:00
Takashi Iwai
a91a4aa1ee Merge branch 'topic/hda' into for-linus 2010-03-01 12:38:54 +01:00
Takashi Iwai
12c2a682b5 Merge branch 'topic/misc' into for-linus 2010-03-01 12:38:49 +01:00
Takashi Iwai
a86ba28583 Merge branch 'fix/misc' into for-linus 2010-03-01 12:38:39 +01:00
Takashi Iwai
a0b62329bb Merge branch 'for-2.6.34' of git://opensource.wolfsonmicro.com/linux-2.6-asoc into topic/asoc 2010-02-25 19:44:00 +01:00
Mark Brown
b4e82b5b78 ASoC: Check progress when reporting periods from i.MX FIQ handler
Currently the i.MX FIQ handler is reporting periods as elapsed based
purely on a timer running in the CPU. This means that any clock
mismatch between the CPU and the audio subsystem can result in the
status reported to applications drifting away from the actual status
of the hardware. This is particularly likely at present since the
SSI driver is only capable of operating in slave mode so it's very
likely that the interface will be clocked from a different source.

Instead check the offset reported by the FIQ and only notify when we
have transferred at least one period, re-firing the timer if we didn't
do so. Also factor out the calculation of the timer expiry time for
make it a bit easier to experiment with.

Note that this only improves the situation, problems can still be
triggered.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-25 15:25:07 +00:00
Mark Brown
9e4a10d27e ASoC: Remove a unused variables from i.MX FIQ runtime data
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-25 15:25:07 +00:00
Kailang Yang
61c2d2b5e7 ALSA: hda - Add/fix ALC269 FSC and Quanta models
Specify proper quirk models for FSC and Quanta machines with ALC269 codec.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-25 08:49:06 +01:00
Kailang Yang
6227cdced0 ALSA: hda - Add ALC670 codec support
- Fixed alc_subsystem_id( ) typo and add new function.
   - !(ass & 0x100000)) ==> Delete this check. It is unnecessary check.
   - Add porti
- ALC670 support

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-25 08:48:44 +01:00
Zhang, Rui
dd2b4a7abf ALSA: hda - remove unnecessary msleep on power state transitions
This will save ~15ms boot time.

The first 10ms sleep was introduced in commit d2595d86e5 for (buggy)
Cxt codecs, so better to limit the sleep to the problem hardware.

For the second 10ms sleep, the HDA spec says:

Power State[1:0]:
00: Node Power state (D0) is fully on.
01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog
playback) which must remain fully on.
10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state.
11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software
control. Note that any low power state set by software must retain sufficient operational capability to properly
respond to subsequent software Power State command.

So 10ms is actually the max wait time. It should be safe to
remove/reduce it and rely on the loop of 1ms-sleeps.

CC: Marc Boucher <marc@linuxant.com>
CC: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Zhang Rui <rui.zhang@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-24 09:12:57 +01:00
Kuninori Morimoto
47fc9a0a80 ASoC: fsi: Modify over/under run error settlement
In current FSI driver, playback function cares only overrun,
and capture function cares only underrun.

But playback function should had cared about underrun,
and capture function should had cared about overrun too.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:42:07 +00:00
Misael Lopez Cruz
db72c2f897 ASoC: OMAP4: Add McPDM platform driver
McPDM platform driver is configured to use sDMA in order to transfer
to/from memory. Support for interfacing with ABE will be added later.

McPDM dai currently supports up to 4 downlink channels and 2 uplink
channels simultaneously, as well as 88.2 and 96 KHz, and a sample
size of 32 bits.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:41:05 +00:00
Candelaria Villareal, Jorge
b3b0b4580b ASoC: OMAP4: Add support for McPDM
McPDM is the interface between Phoenix audio codec
and the OMAP4430 processor. It enables data to be transfered
to/from Phoenix at sample rates of 88.4 or 96 KHz.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:39:48 +00:00
Misael Lopez Cruz
e17dd32f34 ASoC: OMAP: data_type and sync_mode configurable in audio dma
Allow client drivers to set the data_type (16, 32) and the
sync_mode (element, packet, etc) of the audio dma transferences.

McBSP dai driver configures it for a data type of 16 bits and
element sync mode.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:38:52 +00:00
Reimundo Heluani
76e6f5a9ef ALSA: add support for Macbook Air 2,1 internal speaker
Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.

Signed-off-by: Reimundo Heluani <rheluani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 10:55:03 +01:00
Daniel Mack
de48c7bc6f ALSA: usbaudio: consolidate header files
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.

Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.

Now things are also nicely prefixed which makes understanding the code
easier.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:51:56 +01:00
Daniel Mack
7b8a043f26 ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.

However, it allows using these devices for now, without mixer support.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:26 +01:00
Daniel Mack
53ee98fe8a ALSA: usbaudio: implement basic set of class v2.0 parser
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:

* the number of streaming interfaces is now reported by an interface
  association descriptor. The old approach using a proprietary
  descriptor is deprecated.

* The number of channels per interface is now stored in the AS_GENERAL
  descriptor (used to be part of the FORMAT_TYPE descriptor).

* The list of supported sample rates is no longer stored in a variable
  length appendix of the format_type descriptor but is retrieved from
  the device using a class specific GET_RANGE command.

* Supported sample formats are now reported as 32bit bitmap rather than
  a fixed value. For now, this is worked around by choosing just one of
  them.

* A devices needs to have at least one CLOCK_SOURCE descriptor which
  denotes a clockID that is needed im the class request command.

* Many descriptors (format_type, ...) have changed their layout. Handle
  this by casting the descriptors to the appropriate structs.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:24 +01:00
Daniel Mack
8fee4aff8c ALSA: usbaudio: introduce new types for audio class v2
This patch adds some definitions for audio class v2.

Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:20 +01:00
Daniel Mack
28e1b77308 ALSA: usbaudio: parse USB descriptors with structs
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.

Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:12 +01:00
Seth Heasley
32679f95ca ALSA: hda - enable snoop for Intel Cougar Point
This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.

Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:15:37 +01:00
Takashi Iwai
d01aecdf90 ALSA: hda - Remove identical definitions for macmini3 model
The channel mode definitions for macmini3 model are identical with mb5.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:07:15 +01:00
Takashi Iwai
ad6cfc2ac7 Merge remote branch 'alsa/fixes' into fix/misc 2010-02-22 18:45:34 +01:00
Peter Ujfalusi
b9dd94a87e ASoC: core: On resume also check the soc device state
Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22 14:39:42 +00:00
Clemens Ladisch
bf30a4309d ALSA: via82xx: add quirk for D1289 motherboard
Add a headphones-only quirk for the Fujitsu Siemens D1289.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-22 11:15:11 +01:00
Chris J Arges
40717382e0 ALSA: usbaudio Mbox support, output only
Signed-off-by: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 09:56:26 +01:00
Paul Menzel
0708cc582f ALSA: hda-intel: Add position_fix quirk for ASUS M2V-MX SE.
With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1].

Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE.

The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker.

$ lspci -vvnn | grep -A10 Audio
20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)
	Subsystem: ASUSTeK Computer Inc. Device [1043:8290]
	Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
	Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
	Latency: 0, Cache Line Size: 64 bytes
	Interrupt: pin A routed to IRQ 17
	Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K]
	Capabilities: <access denied>
	Kernel driver in use: HDA Intel

[1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user

Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:37:15 +01:00
Paul Menzel
2448158ed2 ALSA: Typo. s/distrubs/disturbs/
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:36:56 +01:00
Takashi Iwai
9d54f08bc7 ALSA: hda - Clean up Intel Mac unsol codes
Use the standard unsol_event callback with each setup callback for
IntelMac models with Realtek ALC885 codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:34:40 +01:00
Luke Yelavich
e458b1fadf ALSA: hda - Add Macmini 3,1 support
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989

Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The
pinout is almost identical to the mb5 quirk, except for no microphone and
the line-in mixer controls being on a different index. Everything works in
2ch mode, but as I am not sure what needs to be changed for 6ch mode, or
whether the Mac Mini's chip supports 6ch mode, I have simply duplicated
the code from the mb5 quirk for the mac mini chmode management. The new
model parameter for this quirk is "macmini3".

Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:27:57 +01:00
Daniel T Chen
ba579eb7b3 ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q
BugLink: https://bugs.launchpad.net/bugs/524948

The OR has verified that the existing model=laptop-eapd quirk does not
function correctly but instead needs model=3stack.  Make this change
so that manual corrections to module-init-tools file(s) are not
required.

Reported-by: Lasse Havelund <lasse@havelund.org>
CC: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:15:21 +01:00
Florian Zumbiehl
04510a74bf ALSA: cs46xx - fix some typos
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:12:30 +01:00
Florian Zumbiehl
7fb2d723e6 ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.

Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:10:54 +01:00
Mark Brown
6c5f1fed49 ASoC: Make pmdown_time a long
Fixes a warning.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-17 14:37:20 +00:00
Peter Ujfalusi
e47c796d58 ASoC: TWL4030: Use codec defaults for Headset initial configuration
Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-17 14:37:20 +00:00
Takashi Iwai
7fb3a069bc Merge branch 'fix/misc' into topic/misc
Conflicts:
	sound/pci/hda/patch_realtek.c
2010-02-17 14:24:46 +01:00
Takashi Iwai
9d3415a8cc Merge remote branch 'alsa/fixes' into fix/misc 2010-02-17 14:22:21 +01:00