Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add experimental support for the Asus Xonar HDAV1.3 Slim sound card.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add experimental support for the Asus Xonar DG sound card.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add support for the AuzenTech X-Meridian 7.1 2G sound card.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add PCI IDs for some unknown models.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate
modes (64-96 kHz) can be reduced to 128x without reducing sound quality.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Replace the get_i2s_mclk callback with tables of MCLK values.  This
simplifies the MCLK-handling code in both the framework and the model-
specific drivers.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Do not apply the headphone gain offset to any but the front DAC.  These
DACs would not be used in headphone mode, so this saves a few register
writes.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Remove the DAC Oversampling mixer control because this setting does not
make much sense.

For cards with the H6 daughterboard, 128x oversampling was disabled
anyway because these high MCLK frequency would not be compatible with
the connector cable.

For cards without the H6 daughterboard, 128x gives a slightly higher
output quality; there is no reason to reduce it to 64x except for saving
power, but then these cards have not been designed to be power efficient
anyway (the D2's blinkenlights cannot be disabled).
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Because of the unshielded connector cable, it is important to use as low
a master clock frequency as possible with the H6.

For double rate modes (64-96 kHz), the MCLK rate is unconditionally
lowered from 512x to 256x because the higher rate would not improve
anything.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The clock output of the CS2000, which is used as master clock for the
DACs, was using half the actual master clock frequency for some reason.
Using the theoretically correct frequency seems also to work in practice.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On the Xonar Essence ST Deluxe, remove all mixer controls that would
require I2C communication with the third DAC, which does not work
because of an addressing conflict with the CS2000 chip.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change the PCM format used for the PCM1796 from left-justified to I2S to
ensure that the correct format is used even for the Essence ST Deluxe's
center/LFE DAC, where I2C does not work because of an address conflict
with the CS2000 chip.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The PCM1796 needs the master clock for I2C communication to work, so
add delays after clock changes to ensure that the clock is stable when
we try to write the DACs' registers.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

To make the I2C communication reliable when using the H6 daughterboard,
reduce the I2C clock frequency.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix wrong register bits for SPI clock cycle times longer than 160 ns,
and adjust the polling loop timeout for these speeds.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The number of DACs can now be deduced from the dac_channels_mixer field,
so the private_data field is no longer needed.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For cards like the Xonar HDAV1.3, differentiate between the number of
PCM channels that can be played and the number of channels whose volume
can be adjusted.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- add some WARN_ONCE
- add multi-I/O helper (and use helper struct)
- fix off-by-1 DMA length bug
- better variable naming
Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- much improved implementation due to clean codec hierarchy
- preparation for potential per-codec spinlock change

NOTE: additionally removes a chip->pcm[codec_type] NULL ptr check
(due to it requiring access to external chip struct),
however I believe this to be ok since this condition should not occur
and most drivers don't check against that either.
Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- use a separate variable for the frequency part, don't always "or" it
- use a "clever"(?) macro to shorten the code
Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- correct samples to be POSIX shell compatible
- add logging of jiffies value in _pointer()
- several comments
- cleanup
Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

flush_scheduled_work() is deprecated and scheduled to be removed.

* cancel[_delayed]_work() + flush_scheduled_work() ->
  cancel[_delayed]_work_sync().

* wm8350, wm8753 and soc-core use custom code to cancel a delayed
  work, execute it immediately if it was pending and wait for its
  completion.  This is equivalent to flush_delayed_work_sync().  Use
  it instead.
Signed-off-by: NTejun Heo <tj@kernel.org>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2cc), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.

ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1).  There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: NBrian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Reformat and update the comments that describe the hardware connections
on the various models.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of the hardcoded "CMI8788", show the actual chip name.

Note: This is neither what the chip is (it's always the same),
      nor what the chip is actually called.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

To help with debugging, add the registers of the model-specific
codecs to the controller and AC97 register dump in the proc file.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The "Front Panel" switch on the Xonar D1/DX actually switches only the
output direction, so mark it appropriately.

The front panel microphone is controlled by the FMIC2MIC bit of the
CM9780.  It was unconditionally enabled on the D1/DX and never set on
the ST(X); add a control for it.  Selecting the front panel microphone
as source does not actually disable the microphone jack, but this is
bug-compatible with the Windows driver, and users rely on it.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The GPIO bit that enables analog output on the Xonar HDAV1.3 also
disables the HDMI audio output, so we better add a switch for it.
Hopefully, this is sufficient to make the HDMI output work.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Initialize the configuration of some unknown GPIO output bits (that
might not be used at all) to be the same as in the Windows driver, just
to be sure.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: https://launchpad.net/bugs/683695

The original reporter states that headphone jacks do not appear to
work.  Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.

Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: NFlorian Faber <faberman@linuxproaudio.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: https://launchpad.net/bugs/682199

A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c of sound-2.6.git, a new model was added with this
machine's PCI SSID.  Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.

Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.

Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far.  This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.

Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.

This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.

Reference:
	https://bugzilla.redhat.com/show_bug.cgi?id=657388Reported-and-tested-by: NKyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).

Reference: https://qa.mandriva.com/show_bug.cgi?id=61159Signed-off-by: NHerton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixes automatic EAPD configuration on Acer 7730G laptop.
Signed-off-by: NDenis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Give more correct chip names for ALC269-variant codecs.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>