1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
       black  pink         blue
2ch: front   ext mic     line in
4ch: front   ext mic     surround
6ch: front   CLFE        surround
  Can be changed in mixer.
5) Sound can be recorded from:
 Internal mic
 Ext mic
 Cd
 Line in
6) 2 separate capture channels.
Signed-off-by: NDenis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The last patch in this series is for ALC662; pretty similar as the
previous patch for ALC861-VD.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec.  The change is simple, just checking the
pin connection whether it's a speaker-out.  When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.

However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins.  The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c.  Thus, there
are two possible volumes.

When only one of them is used, we can name it as "Master".  OTOH, when
both are used at the same time, they have to be named uniquely.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec.  The new helper parses the codec connections dynamically
isntead of fixed indicies.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.

This patch adds the identification of such laptops to apply the
standard BIOS-probing method.  Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.
Tested-by: NAlexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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>