Add "Internal" node type.
Remove GPI and GPO node types.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The channel count can be queried to determine which.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Unlike other streams which support 1..max channels,
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

and update HPI version to 4.10
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Elapsed counter should only count data committed to snd_pcm_period_elapsed,
rather than all data available
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

These drviers defined only variables but didn't declare as module
parameters.  Also fix the enable variable to bool type.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

module_param(bool) used to counter-intuitively take an int.  In
fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy
trick.

It's time to remove the int/unsigned int option.  For this version
it'll simply give a warning, but it'll break next kernel version.
Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Sergiusz Urbaniak <sergiusz.urbaniak@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The advantage of kcalloc is, that will prevent integer overflows which could
result from the multiplication of number of elements and size and it is also
a bit nicer to read.

The semantic patch that makes this change is available
in https://lkml.org/lkml/2011/11/25/107Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The advantage of kcalloc is, that will prevent integer overflows which could
result from the multiplication of number of elements and size and it is also
a bit nicer to read.

The semantic patch that makes this change is available
in https://lkml.org/lkml/2011/11/25/107Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: David Dillow <dave@thedillows.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are some AC97 codec and board combinations that have been observed
to take a very long time to respond after the cold reset has completed.
In one case, more than 350 ms was required. To allow users to have sound
on those platforms, we'll wait up to 500ms for the codec to become
ready.

As a board may have multiple codecs, with some faster than others to
reset, we add a module parameter to inform the driver which codecs
should be present.
Reported-by: KotCzarny <tjosko@yahoo.com>
Signed-off-by: David Dillow <dave@thedillows.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The verb command in stac92xx_post_suspend caused the audio to stop
working after resuming from S3 mode on HP laptops with the VREF-pin
mute-LED control.  Removing relevant post_suspend registering.

Although removing D3 on AFG is no optimal solution, the impact should
be small in comparison with the broken S3/S4.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This quirk sould be reverted. It has the following probems:

1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the
ID used matches a much broader range, potentially all boards containing a
VIA chipset model in the family of vendor VIA 0x1106 and audio device ID
0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets.

2) VIA chipsets rely on azx_via_get_position() to handle correctly dma
transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO
leads to partially corrupted input buffers during capture. The effects
of this bug are not immediately visible, it took strong DSP expertise,
some expensive signal generator and a spectrum analyzer to identify it
and verify correct behaviour using original default.

3) It's almost certain that the quirk did not fix the real problem,
if there was one. Refer to original submission:
http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.htmlSigned-of-by: Marc Vertes <mvertes@sigfox.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Some HP laptops use a pin VREF for controlling the mute LED, and such a
pin shouldn't be powered off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Currently the SigmaDSP firmware loader only works correctly on little-endian
systems. Fix this by using the proper endianess conversion functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

The firmware header is not part of the CRC, so skip it. Otherwise the firmware
will be rejected due to non-matching CRCs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

The SigmaDSP firmware loader currently does not perform enough boundary size
checks when processing the firmware. As a result it is possible that a
malformed firmware can cause an out of bounds memory access.

This patch adds checks which ensure that both the action header and the payload
are completely inside the firmware data boundaries before processing them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon
Roland SH-201 table entry as template.  USB MIDI and audio was tested
with Muse and Audacity.
Signed-off-by: John F Leach <jfleach@jfleach.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Based on initial data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

This patch converts the drivers in sound/* to use the
module_platform_driver() macro which makes the code smaller and a bit
simpler.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

VT1708 has no support for unsolicited events per jack-plug, the driver
implements the workq for polling the jack-detection.  The mixer element
"Jack Detect" was supposed to control this behavior on/off, but this
doesn't work properly as is now.  The workq is always started and the
HP automute is always enabled.

This patch fixes the jack-detect control behavior by triggering / stopping
the work appropriately at the state change.  Also the work checks the
internal state to continue scheduling or not.

Cc: <stable@kernel.org> [v3.1]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The CS420X_IMAC27 was copied from the line before but CS420X_APPLE
was clearly intented.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

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

Factor out some boilerplate code.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are cases where there is no working codec on the soc-audio devices,
and snd_soc_suspend() will skip such device when suspending. Yet its
counterpart snd_soc_resume() does not check this, causing complaints
about spinlock lockup:

[  176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8
[  176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24)
[  176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158)
[  176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68)
[  176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c)
[  176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0)
[  176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c)
[  176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0)
[  176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0)
[  176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8)
Signed-off-by: Eric Miao <eric.miao@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Just checking the code in cs42l51_fill_cache():
The cache pointer points to codec->reg_cache + 1.
I think it is because CS42L51_FIRSTREG is 0x01,
so codec->reg_cache[0] is not used here.

Then we read CS42L51_NUMREGS bytes to cache.
So we need reg_cache_size to be CS42L51_NUMREGS + 1.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca968
("[ALSA] Revised AT32 ASoC Patch"). That support depends on
BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got
added in v2.6.27. It still doesn't. It has never been possible to even
build this driver. Drop it.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Use an inline function for the common pattern for assigning a capsrc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When the imux entries are rebuilt in alc_rebuild_imux_for_auto_mic(),
the initialization of index field is missing.  It may work without it
casually when the original imux was created by the auto-parser, but
it's definitely broken in the case of static configs where no imux was
parsed beforehand.  Because of this, the auto-mic switching doesn't
work properly on some model options.

This patch adds the missing initialization of index field.
Reported-by: Dmitry Nezhevenko <dion@inhex.net>
Cc: <stable@kernel.org> [v3.1]
Signed-off-by: Takashi Iwai <tiwai@suse.de>