Merge branch 'topic/asoc' into for-next

Merge branch 'topic/hda' into for-next

ALSA: hda - remove static intelhdmi configurations

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - auto parse intelhdmi cvt/pin configurations

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - get intelhdmi max channels from widget caps

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - vectorize intelhdmi

The Intel IbexPeak HDMI codec supports 2 converters and 3 pins,
which requires converting the cvt_nid/pin_nid to arrays.

The active pin number (the one connected with a live HDMI monitor/sink)
will be dynamically identified on hotplug events.

It exports two HDMI devices, so that user space can choose the A/V pipe
for sending the audio samples.

It's still undefined behavior when there are two active monitors
connected and routed to the same audio converter.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - reorder intelhdmi prepare/cleanup callbacks

No behavior change.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - use pcm prepare/cleanup callbacks for intelhdmi

Remove pcm callbacks open/close in favor of the prepare/cleanup.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - remove intelhdmi dependency on multiout

We'll be managing multiple HDMI audio sources/sinks on our own.
So remove multiout dependency from intelhdmi.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - convert intelhdmi global references to local parameters

No behavior change.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - allow up to 4 HDMI devices

The new Intel HDMI codec supports 2 independant HDMI/DisplayPort pipes.
We'll be exporting them as 2 pcm devices. So bump up the allowed number
of HDMI devices.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - vectorize get_empty_pcm_device()

This unifies the code and data structure,
and makes it easy to add more HDMI devices.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'for-2.6.32' into for-2.6.33

ASoC: sh: FSI: Add capture support

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: sh: FSI: Remove DMA support

SuperH FSI device have the hardware limitation to use DMA.
If DMA is used, LCD output will be broken.
Maybe there are some solution. But I don't know how to do it now.
This patch remove DMA support and use soft transfer.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ALSA: hda - select IbexPeak handler for Calpella

An earlier patch merely adds id for 0x80862804.
It has 2/3 cvt/pin nodes and shall be tied to the IbexPeak handler.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/asoc' into for-next

Merge branch 'fix/asoc' into for-next

ASoC: Modifying Kconfig/Makefile for AM3517 EVM

Modifying the Kconfig and Makefile in sound/soc/omap folder
to add support for OMAP3517 / AM3517 EVM in Alsa SoC.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC

Adding support for OMAP3517 / AM3517 EVM in Alsa SoC.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal

The pop-removal specific values are configured for TWL4030 codec
for OMAP3EVM through this patch.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Add APLL supply for the capture path

Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Change APLL powering sequence

It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.

If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.

Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Vibra motor stop fix when it is driven with audio

This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.

Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: CS4270: export de-emphasis filter as ALSA control

The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Minor SMDK64xx WM8580 cleanups

Fix up some comments, remove all enable_pin() calls (edge widgets
are all enabled by default) and mark the microphone as disabled by
default since it requires a resistor fit to connect it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Change codec_muted to apll_enabled

codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Remove bypass tracking

Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.

Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.

Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Add regulator support for WM8731

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Driver registration via twl4030_codec MFD

Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: use the twl4030-codec.h for register descriptions

Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

OMAP: Platform support for twl4030_codec MFD

Add needed platform data for the twl4030_codec MFD on boards,
where the audio part of the twl4030 codec is used.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

MFD: twl4030: add twl4030_codec MFD as a new child to the core

New MFD child to twl4030 MFD device.

Reason for the twl4030_codec MFD: the vibra control is actually in the codec
part of the twl4030. If both the vibra and the audio functionality is needed
from the twl4030 at the same time, than they need to control the codec power
and APLL at the same time without breaking the other driver.
Also these two has to be able to work without the need for the other driver.

This MFD device will be used by the drivers, which needs resources
from the twl4030 codec like audio and vibra.

The platform specific configuration data is passed along to the
child drivers (audio, vibra).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text

I thought it could be usefull to add some information on how to get the device
fully supported by loading a line discipline on the modem line.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1

After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:

        omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
        omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);

Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.

The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.

Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: tlv320dac33: typo fix in the header

Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Amstrad Delta minor cleanups

Hi Mark,

Here is a patch that corrects small omissions I have found in my code.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.32' into for-2.6.33

ASoC: Fix possible codec_dai->ops NULL pointer problems

Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Move dereference after NULL test

If the NULL test on jack is needed, then the derefernce should be after the
NULL test.

A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):

// <smpl>
@match exists@
expression x, E;
identifier fld;
@@

* x->fld
  ... when != \(x = E\|&x\)
* x == NULL
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: au1x: psc-ac97: reorganize timeouts

Codec read/write functions: wait 21us between the pokings of hardware.
Add timeouts to unbounded loops waiting for bits to change.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: au1x: psc-ac97: verify correct codec register was read

Verify that the correct register has been received from the codec.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk

Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/asoc' into for-next

Merge branch 'fix/asoc' into for-next

Merge branch 'for-2.6.32' into for-2.6.33

ASoC: Codec driver for Texas Instruments tlv320dac33 codec

Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.

TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.

The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.

Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).

b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: finally enable support for eXeda and CM-X300

Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Remove snd_soc_suspend_device()

The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'fix/hda' into for-next

ALSA: hda - Fix capture source checks for ALC662/663 codecs

The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls()
to check the capture source selections.  This should be alc882, instead.

Reference: Novell bnc#546918
http://bugzilla.novell.com/show_bug.cgi?id=546918

Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/hda' into for-next

ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF

48 kHz limit is for slightly better stability, and sample rates other
than 48k (like 96k/192k) are for better sound quality.
We choose better quality, so remove the 48k limit.

Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/hda' into for-next

ALSA: hda - Allow all formats as default for Nvidia HDMI

In the commit f0613d5752d8f7d1d02e6d40947f38877fdf9c90
    ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.

Let's enable all formats/rates as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/misc' into for-next

ALSA: aaci: ARM1176 aaci-pl041 AC97 register read timeout

After a reboot on an ARM1176 which amounts to a softreset, it has been
noted that the ALSA driver does not get registered and the probe fails
with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process
of reading from a register the SL1TxBusy bit is set indicating that the
transceiver is busy and remains so until the default timeout occurs.
Set the Power down register 0x26 to an arbitrary value as specified in
the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take
their default state.

Signed-off-by: Philby John <pjohn@in.mvista.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/hda' into for-next

ALSA: hda - Fix volume-knob setup for Dell laptops with STAC9228

The volume-knob widget needs to be set with 0x7f instead of 0xff
for Dell laptops with STAC9228 codec, too, like the previous commit.

Reference: Novell bnc#545013
http://bugzilla.novell.com/show_bug.cgi?id=545013

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - Fix mute sound with STAC9227/9228 codecs

On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup.  The delta bit (bit 7)
shouldn't be set for these devices.

This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.

Reference: Novell bnc#546006
http://bugzilla.novell.com/show_bug.cgi?id=546006

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: S3C: Remove <plat/audio.h>

Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Simtec Linux Team <linux@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Serialize access to dapm_power_widgets()

Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.

Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'fix/misc' into for-next

ALSA: bt87x - Add a whitelist for Pinnacle PCTV (11bd:0012)

Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/misc' into for-next

ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd

snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd
This is the correct error number for telling the USB system that this
driver is not for the device.

Signed-off-by: Tobias Hansen <Tobias.Hansen@physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/hda' into for-next

ALSA: hda - Fix overflow of spec->init_verbs in patch_realtek.c

ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to
the recent changes.  Simply increase the array size to avoid the overflow.

Reported-by: Luca Tettamanti <kronos.it@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: TPA6130A2: Make tpa6130a2_power as static

The power for the amplifier should be handled internally
by the tpa6130a2 driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/misc' into for-next

ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency

SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.

Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/hda' into for-next

sound: use semicolons to end statements

Fixes:

sound/pci/hda/patch_via.c: In function 'patch_vt1718S':
sound/pci/hda/patch_via.c:4951: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1716S':
sound/pci/hda/patch_via.c:5441: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt2002P':
sound/pci/hda/patch_via.c:5794: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1812':
sound/pci/hda/patch_via.c:6148: error: expected expression before 'return'

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/misc' into for-next

ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@

If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.

Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/misc' into for-next

ALSA: snd_dma_pointer workaround for chipsets with buggy DMA

The chipsets with the isa_dma_bridge_buggy set do not stop DMA during
DMA counter reads. The DMA counter is read in two 8-bit read steps
on x86 platform. Sometimes, such reads happen during higher byte
change so the lower byte is already decremented (rolled over) but
the higher byte is not. It introduces an error that position is
moved 256 bytes ahead of the true position. Thus, the next DMA
position read can return a lower value then the previous read.
If the DMA position is decreased (reversed) the ALSA subsystem is
tricked into the playback underrun error and resets the playback.
It results in a "pop" during a playback.

Work around the issue by reading the counter twice and choosing a higher
value.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: wss: reuse CS4231 controls for AD1848

The C4231 control set is a superset of the AD1848 control
set so reuse the CS4231 controls definitions for the AD1848.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/hda' into for-next

ALSA: HDA VIA: Only cosmetic changes

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: comments: update copyright, changeset, etc.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Change PW4 connect select default to to MW0.

According to customer request, hp should be default to redirected mode,
i.e. PW4 connect select default to to MW0.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: rename vt1708_control_templates[].

To via_control_templates[].

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VT1812 support.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VT2002P support.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VT1716S support.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VT1828S and VT2020 support.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VT1718S support.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb

As init verbs, vt17xx_volume_init_verb is a better place to hold them.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Replace MIC_BOOST_VOLUME.

With snd_hda_override_amp_caps.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls.

Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls.

Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup

Replaced with via_playback_multi_pcm_prepare/cleanup to support
multi-stream operations

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function.

like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port
Connectivity field into 'AC_JACK_PORT_COMPLEX'

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add Jack detect feature for VT1708.

VT1708 does not support unsolicited response, but we need hp detect to
automute speaker. Implemented in workqueue.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Refresh front playback mute in via_hp_automute.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: When changing input source, update power state.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Add smart5.1 function.

Smart 5.1 is for 3-jacks model, to reuse input pins as outputs.
While off, they act as "line out" / "line in" / "mic in".
While on, they acts as "line out" / "back left/right" / "center/lfe".

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA VIA: Rewrite via_independent_hp_put

Use hp_independent_mode_index to store hp index, and simplify function
via_independent_hp_put with it.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>