It makes no longer sense to keep the old hwdep user-space firmware
loading, which has been deprecated since ages ago.

Just add a hard dependency on CONFIG_FW_LOADER and drop the useless
code.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since we keep the pin default config values anyway internally, we
don't have to set the values in the codec.  This patch removes the
code writing the pincfg values.

As a gratis bonus, we can remove also the code restoring the original
pincfg values at PM resume or module free.  This will give us more
benefit, as it can reduce the unnecessary power-up of codecs.

This won't change the driver functionality.  The only difference would
be that the codec proc file will show the original pincfg values
instead of the actually referred values.  The actually referred values
can be determined from sysfs *_pin_configs files.
(Also hda-emu was updated to follow this change.)
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The free callback is called at the state where no extra verbs are
executed, thus calling *_shutup() is useless, as it's checking the
shutdown flag.  Remove such superfluous calls.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

We've got a report that the runtime PM may make the codec the
unresponsive on AMD platforms.  Since the feature has been tested only
on the recent Intel platforms, it's safer to limit the support to such
devices for now.

This patch adds a new DCAPS bit flag indicating the runtime PM
support, and mark it for Intel controllers.
Reported-and-tested-by: NJulian Wollrath <jwollrath@web.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi 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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the bus reset is performed during the suspend/resume (including
the power-saving too), it calls snd_hda_suspend() and
snd_hda_resume() again, and deadlocks eventually.

For avoiding the recursive call, add a new flag indicating that the PM
is being performed, and don't go to the bus reset mode when it's on.
Reported-and-tested-by: NJulian Wollrath <jwollrath@web.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There is a precedence bug because | has higher precedence than ?:.  This
code was cut and pasted and I fixed a similar bug a few days ago.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I don't think this works as intended.  '|' higher precedence than ?: so
the bitwize OR "0 | (val & STR_MOST)" is a no-op.

I have re-written it to be more clear.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Bah, forgot this again...
Signed-off-by: NTakashi 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: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is another variant of iMac 9,1 with a different codec SSID.
Reported-and-tested-by: NEveraldo Canuto <everaldo.canuto@gmail.com>
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

alc269_toggle_power_output() was only use in ALC269VB.  I rename it to
alc269vb_toggle_power_output().
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NAdrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: NTakashi 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: NMassimo Del Fedele <max@veneto.com>
Signed-off-by: NTakashi 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: NMassimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NMassimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NCyrus Lien <cyrus.lien@canonical.com>
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi 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: NSergiu Giurgiu <sgiurgiu11@gmail.com>
Signed-off-by: NTakashi 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: NRobin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Mark structures that won't change const.
Signed-off-by: NDaniel J Blueman <daniel@quora.org>
Signed-off-by: NTakashi 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: NAlexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: NDaniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAlexander Stein <alexander.stein@systec-electronic.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NLars R. Damerow <lars@pixar.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Correct spelling typo in debug messages within drivers/sound
Signed-off-by: NMasanari Iida <standby24x7@gmail.com>
Signed-off-by: NTakashi 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: NLaurence Darby <ldarby@tuffmail.com>
Signed-off-by: NTakashi 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: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a couple of tracepoints to snd-hda-intel for tracing the position
and the trigger timings.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

dev_<level> calls take less code than dev_printk(KERN_<LEVEL>
and reducing object size is good.
Coalesce multiline formats for easier grep.
Coalesce segmented single line formats too.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The BIOS on HP dv5 doesn't have the DMI string to guide the setup of
mute led GPIO and polarity. Associate this laptop with the hp-inv-led
model.
Signed-off-by: NGustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: NVinícius Angiolucci <angiolucci@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The check of the return value from vortex_wtdma_bufshft() in
vortex_interrupt() is useless as it executes no code.  Actually the
return value is intentionally ignored because the delta calculation
for wavetable doesn't work always correctly.  For avoiding the
confusion, a comment is added and the superfluous if () is removed.
Reported-by: NFengguang Wu <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: NKees Cook <keescook@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix build errors by using correct kconfig symbol name:

sound/pci/ice1712/psc724.c:417:5: error: 'struct snd_ice1712' has no member named 'pm_resume'
sound/pci/ice1712/psc724.c:418:5: error: 'struct snd_ice1712' has no member named 'pm_suspend_enabled'

[Fixed another #ifdef CONFIG_PM in the same file, too, by tiwai]
Signed-off-by: NRandy Dunlap <rdunlap@xenotime.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Reuse code from clocksource to handle wall clock counter.
Since wrapparound occurs, the audio timestamp is reinitialized
to zero on a trigger. Synchronized linked devices will
start counting from same reference to avoid any drift.

Max buffer time is limited to 178 seconds to make sure
wall clock counter does not overflow

Wallclock timestamps are disabled on capture streams
until we figure out how to handle digital inputs.
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>