The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.

[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs.  Thus it needs own definitions of capture elements.]

This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).

Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.
Signed-off-by: NHerton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Move static codes to setup from init_hook for each model.

Also, use the common auto-mic selection helper for devices that support
auto-mic selection.  They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC.  Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.

This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.

Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition.  This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).

This patch avoids the overriding by adding the proper checks.

Reference: Novell bnc#529467
	https://bugzilla.novell.com/show_bug.cgi?id=529467Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the auto-mic feature is enabled, we should support only one
capture stream.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode).  The auto-mic mode is enabled
only when one internal mic and one external mic are present.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.

Reference: Novell bnc#527361
	https://bugzilla.novell.com/show_bug.cgi?id=527361Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

Check whether index is within bounds before testing the element.
Signed-off-by: NRoel Kluin <roel.kluin@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.

Reference: Novell bnc#526325
	https://bugzilla.novell.com/show_bug.cgi?id=526325Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo".  Now the size is doubled.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It auto mutes all 8-channel outputs at rear panel when
the front panel headphone is connected.
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This 2-channel mode is useful in that it will broadcast
a 2-channel audio stream to all front/side/... ports.
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.

Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a helper macro to retrieve the widget type from wiget cap bits.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted.  This patch fixes
the issue.

Reference: Novell bnc#404873
	https://bugzilla.novell.com/show_bug.cgi?id=404873Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

Here are the new sound enabling patches for IbexPeak.

Summary of tested features:

  - playback
    - Front Headphone: OK
    - 8 channel audio: Front/Rear/CLFE/Side all OK

  - recording
    - Front Mic/Rear Mic: both OK
      (front/rear/line mics are selectable in the "Input source" alsamixer control)
    - Line In: not working
      (in 6ch mode, its amp/mute, direction and route all looks fine,
       so I'm a little puzzled)
      (hopefully no one will care this feature)

  - digital SPDIF input/output: not tested (no equipment)
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.

Also, unmute the unused output connections to avoid noises on inputs.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.

The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).

This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the merge error at the commit 305355aa,
an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and
ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id
64a8be74Signed-off-by: NDavid Heidelberger <d.okias@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There is a regression, introduced in aa202455
(in alsa-kernel) which I noticed when trying to use the headphone socket on
my EeeCPC 901: the output was *very* quiet, practically silent.

This patch corrects the control types to that which was obviously intended in
the referenced commit.
Signed-off-by: NDarren Salt <linux@youmustbejoking.demon.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NWilliam Weston <weston@sysex.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The mixer elements created for ASUS eeePC 1000 with ALC269 aren't
standard but strange words like "LineOut".  Rename the element names
to follow the standard one like "Headphone" and "Speaker".
Also, split the volumes to each so that the virtual master can control
them.

The alc269_fujitsu_mixer is removed because it's now identical with
the new eeepc mixer.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of expanding alc882_init_verbs to two elements via a macro,
manually expand to each entry.  This makes clear that some have already
the full slot for init_verbs array (currently 5).
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

After merting patch_alc882() and patch_alc883(), the initialization of
mixer amp 0x0b was missing in alc882_base_init_verbs[].
This is usually no critical problem, but it can disable the power-saving
as the default state, so better to put to mute these channels.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless
of the input pin types although it shouldn't be for line-in pins.

This patch fixes the behavior to follow other codecs.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

During the changes to clean up / fix the realtek codec initialization
routines in commit 4a79ba34,
I forgot to add the check for ALC268 and ALC269.
This resulted in the missing EAPD and COEF setup for these codecs.

This patch adds the missing checks for these codecs.

Reference: bko#13633
	http://bugzilla.kernel.org/show_bug.cgi?id=13633Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Line In connector is set up as PIN_IN by default, using
VREF_HIZ. It is connected to both ADCs, so add it to both
input selectors.
Also add the ability to use the input mix (on a SoundBlaster
one would call this "What You Hear").
Signed-off-by: NTony Vroon <tony@linx.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For Acer Aspire 6930G (1025:015e), acre-aspire-6530g model matches
obviously better.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the following bugs of acer-aspire-6530g model with ALC888:
- HP jack to mute all speaker outputs including LFE
- Make digital built-in mic working
Signed-off-by: NEmilio López <buhitoescolar@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Realtek codecs require the pin-sense trigger call before actually
reading the pin-sense.  Without this, the pin-detection might not be
done accurately.

This patch adds the pin-capability check and issues the trigger call
if required.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add the digital-mic support with ALC262 auto model.
The new ALC262 models have the digital mic at NID 0x12.  This widget
isn't checked in the current alc262_auto_create_analog_input_ctls()
since it's under 0x18.  So, just reuse the routine for alc269 to fix
the behavior.

But, it doesn't suffice: the digital mic is supported only with the
ADC0, we have to exclude other ADCs when d-mic is detected.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the check of the input-source type by checking the widget type of
each capture-source item.  Since some codecs can have both the mixer
and selector types depending on the ADC, alc_mux_enum_put() needs to
check each widget.

With this change, spec->capture_style gets unneeded, so it's removed,
too.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It needs model=toshiba-s06 to work with the digital-mic.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

The commit f9e336f6
    ALSA: hda - Unify capture mixer creation in realtek codes
removed the "Input Source" mixer element creation for toshiba-s06 model
because it contains a digital-mic input.

This patch take the control back.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>