Also change printk level for the two others.
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NWei Ni <wni@nvidia.com>
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This avoids a lockup at boot.
Signed-off-by: NMichele Ballabio <barra_cuda@katamail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit eaa9b3a7 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: NFrederik Deweerdt <frederik.deweerdt@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

Support nvidia MCP89 and GT21x 8ch hdmi audio.
Add some eld support.
Signed-off-by: NWei Ni <wni@nvidia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NWei Ni <wni@nvidia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NNorberto Lopes <nlopes.ml@gmail.com>
Acked-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

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: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NSascha Hauer <s.hauer@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NSascha Hauer <s.hauer@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Specify proper quirk models for FSC and Quanta machines with ALC269 codec.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- 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: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This will save ~15ms boot time.

The first 10ms sleep was introduced in commit d2595d86 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: NZhang Rui <rui.zhang@intel.com>
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NKuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMargarita Olaya <x0080101@ti.com>
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya <x0080101@ti.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.
Signed-off-by: NReimundo Heluani <rheluani@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.
Signed-off-by: NSeth Heasley <seth.heasley@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The channel mode definitions for macmini3 model are identical with mb5.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add a headphones-only quirk for the Fujitsu Siemens D1289.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: NMarc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

Signed-off-by: NChris J Arges <christopherarges@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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-userSigned-off-by: NPaul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NPaul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Use the standard unsol_event callback with each setup callback for
IntelMac models with Realtek ALC885 codecs.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NLuke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NLasse Havelund <lasse@havelund.org>
CC: <stable@kernel.org>
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NFlorian Zumbiehl <florz@florz.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.
Signed-off-by: NFlorian Zumbiehl <florz@florz.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixes a warning.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>