The Acer Aspire AO756 has an analog built-in mic on nid 0x1b and an
external mic on nid 0x19.  The BIOS doesn't set this up.

The mic detect on this Acer Aspire netbook and Acer C7 ChromeBook is
only valid when the headphone is plugged.  The detect circuit relies on
the tip detect switch being closed on the jack.  Tell hda_jack to ignore
the mic sense unless the headphones are plugged.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Introduce the concept of a "gated" jack.  The gated jack's pin sense
is
only valid when the "gating" jack is plugged.  This requires checking
the gating jack when the gated jack changes and re-checking the gated
jack when the gating jack is plugged/unplugged.

This allows handling of devices where the mic jack detect floats when
the headphone jack is unplugged.

[Rewritten for fixing the possible snd_array reallocation, covering
 the missing callback calls and jack sync operations, as well as some
 code cleanups -- tiwai]
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

We set "s" before we have capped "speed" so it could be the wrong value.
This could lead to a divide by zero bug.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Use bitmap_weight to count the total number of bits set in bitmap.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails.  This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():

  usbaudio: unit 51: unexpected type 0x09
  snd-usb-audio: probe of 1-8:1.0 failed with error -5

Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out.  It would be nice to
avoid that.

While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces.  This allows audio and MIDI interfaces to be
used without any obvious issues.

Interface 0 is the main one to ignore.  It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this.  Interface 3 is some sort of control interface that
I don't understand:

    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        3
      bAlternateSetting       0
      bNumEndpoints           0
      bInterfaceClass         1 Audio
      bInterfaceSubClass      1 Control Device
      bInterfaceProtocol      0
      iInterface              0
      AudioControl Interface Descriptor:
        bLength                 9
        bDescriptorType        36
        bDescriptorSubtype      1 (HEADER)
        bcdADC               1.00
        wTotalLength            9
        bInCollection           1
        baInterfaceNr( 0)       1
Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add a MIDI driver for the Stanton FireWire DJ controllers.
Tested-by: Sean M. Pappalardo - D.J. Pegasus <spappalardo@mixxx.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>

Bah, forgot this again...
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Assume that unknown ICE1724-based cards are AC97-only that can suspend
without any additional card-specific code.

This fixes suspend on Gainward Hollywood@Home 7.1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

A closer look shows that the name is not even used and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When two built-in speakers are found on the machine, we can suppose
it's rather a 2.1 speaker system with a bass output instead of
front/surround channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL

This patch fixes these places.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

alc269_toggle_power_output() was only use in ALC269VB.  I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback.  It turned out that the problem is that we don't
wait until all URBs are killed.

This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:

    status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);

    lock = (status & (0x1<<idx)) ? 1 : 0;
    sync = (status & (0x100<<idx)) ? 1 : 0;

The index is given in kcontrol->private_value:

    HDSPM_SYNC_CHECK("WC SyncCheck", 0),
    HDSPM_SYNC_CHECK("AES SyncCheck", 1),
    HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
    HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
    HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
    HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
    HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
    HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
    HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),

The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e.  This confuses
the auto-parser.  Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i).  When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[].  This confuses is_empty_dac() and trims the detected DAC
in later reference.

This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

On some of the PantherPoint HDMI machines we currently enable, we're seeing
trouble with unsol events, i e detecting monitor presence, especially when
on battery and after suspend/resume.

BugLink: https://bugs.launchpad.net/bugs/1075882Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This patch adds support for ASUS - Xonar DSX sound cards. Tested on
openSUSE 12.2 with kernel:
Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux
Works:
 - play sounds
 - adjust volume on master channel.
 - mute .

Since Xonar DS uses the same chip, everything that works for DS should
work for DSX as well.
Signed-off-by: Sergiu Giurgiu <sgiurgiu11@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.

Original fix credit to Javeed Shaikh.

BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add generic ESS vendor ID to pm_whitelist. This should fix suspend on
all Maestro-2 and Maestro-2E based PCI cards.
Tested on Terratec DMX and SF64-PCE2.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Mark structures that won't change const.
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.

This fixes the digital mic on the Macbook Pro 10,1/Retina.
Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Alexander Stein <alexander.stein@systec-electronic.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.
Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Correct spelling typo in debug messages within drivers/sound
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Starting audio or seeking in various music players causes
setup_dig_out_stream() to be called, which resets the SPDIF stream,
which caused one DAC (but not another) to make a clicking noise every
time.

This patch ensures the reset only happens when it needs to, which is
when the format changes, and makes the code a little more readable.
Signed-off-by: Laurence Darby <ldarby@tuffmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The rate isn't restored properly after resume since it's only set up
in hw_params, and not in prepare callback.  For fixing it, put the
corresponding call to resume callback as well.
Reported-and-tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

... for migrating the core changes for USB-audio disconnection fixes

When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.

The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.

Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.

The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
 mixer.c and others; the device speed is now cached in subs->speed
 instead of accessing to chip->dev

The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.

The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks.  They'll be covered by the
upcoming change to rwsem.

Also the mixer codes are untouched, too.  These will be fixed in
another patch, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>