Convert the bitfield coded enums to the new VALUE_ENUM type.
Remove the enum check, since the VALUE_ENUM type can handle
the bitfield coding and also handles the 'holes' in the bitfield.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch introduces a new enum type.
In this enum type each enumerated items referred with a value.

This new enum type can handle enums encoded in bitfield, or any other
weird ways. twl4030 codec has several mux selection register, where the
input/output mux is coded in a bitfield. With the normal enum type this type
of mux can not be handled in a clean way.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If AC97 unit is in partially enabled state, early request_irq can trigger
IRQ storm or even full hang up. Workaround this by forcibly switching ACLINK off
at the start of the probe.
Signed-off-by: NDmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Let's have audio playback not sound like chipmunks, 'k? :)

ASP1 on the DM355 EVM uses a 27 MHz external audio clock, not
the slower clock used with ASP0 on the DM6446 EVM.

Also, that slower ASP0 clock on the DM6446 is 12.288 MHz,
not 22.5792 MHz ... 48 KHz sample rate (x256), not a double
speed 44.1 KHz sample rate (which could be done, but isn't
what the board init code now sets up).
Signed-off-by: NDavid Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Pandora has all TWL4030 output pins floating, it uses
external DAC for playback. Mark those outputs as not
connected using DAPM calls.
Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

N810 bootloader muxes I2S pins for OMAP2420 EAC block while N810 ASoC
drivers are using McBSP block so the kernel have to change configuration
runtime.

Author has not seen problems using kernel pin multiplexing on N810 but very
many times unworking audio after forgotten to enable it and spending
15 minutes each time to figure it out again...

This change makes it easier for other users as well. If problems arise, then
they are better to find and fix in OMAP pin multiplexing framework.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Minor bugfix:  now that DaVinci kernels can support multiple
boards, board-specific ASoC components need to verify they're
running on the right board before initializing.
Signed-off-by: NDavid Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: NKevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Almost all parameters that have been misnamed in the comments.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Set the invalid dma channel to -1 (and check properly for it) in
pxa2xx_pcm_hw_free().  Was assuming 0 is an invalid channel number but 0
is a valid pxa dma channel num.
Signed-off-by: Nstephen <stephen.ware@eqware.net>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds DAPM implementaion for the capture path
on twlx030.

TWL has two physical ADC and two digital microphone (stereo) connections.
The CPU interface has four microphone channels.
For simplicity the microphone channel paths are named as:
TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid
TX2 (Left/Right)

Input routing (simplified version):
There is two levels of mux settings for TWL in input path:
Analog input mux:
 ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic}
 ADCR <- {Off, Sub mic, AUXR}

Analog/Digital mux:
TX1 Analog mode:
 TX1L <- ADCL
 TX1R <- ADCR
TX1 Digital mode:
 TX1L <- Digimic0 (Left)
 TX1R <- Digimic0 (Right)

TX2 Analog mode:
 TX2L <- ADCL
 TX2R <- ADCR
TX2 Digital mode:
 TX2L <- Digimic1 (Left)
 TX2R <- Digimic1 (Right)

The patch provides the following user controls for the capture path:
Mux settings:
"TX1 Capture Route": {Analog, Digimic0}
"TX2 Capture Route": {Analog, Digimic1}
"Analog Left Capture Route":  {Off, Main Mic, Headset Mic, AUXL, Carkit Mic}
"Analog Right Capture Route": {Off, Sub Mic, AUXR}

Volume/Gain controls:
"TX1 Digital Capture Volume": Stereo gain control for TX1 path
"TX2 Digital Capture Volume": Stereo gain control for TX2 path
"Analog Capture Volume":      Stereo gain control for the analog path only

Important things for the board files:
Microphone bias:
"Mic Bias 1":       Bias for Main mic or for digimic0 (analog or digital path)
"Mic Bias 2":       Bias for Sub mic or for digimic1 (analog or digital path)
"Headset Mic Bias": Bias for Headset mic

When the routing configured correctly only the needed components will be
powered/enabled.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Modify the enum filter to more generic that it will filter
out the enums with text "Invalid".
The enum filter also required for the capture path.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The card based on stv0299 or stv0288 demodulators.
Signed-off-by: NIgor M. Liplianin <liplianin@me.by>
Signed-off-by: NMauro Carvalho Chehab <mchehab@redhat.com>

Signed-off-by: NHerton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: stable@kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix a typo (& and &&)
Signed-off-by: NRoel Kluin <roel.kluin@gmail.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.

- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
  delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
  error is populated also there
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: NArun KS <arunks@mistralsolutions.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fixed incorrect mixer index values for 92hd83xx codec's audio
input mixer.
Signed-off-by: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add check to determine if dinput_mux is set by any of patch_stac*() functions,
otherwise a invalid pointer my be referenced causing gibberish to mixer values.
Signed-off-by: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added the model=hp-m4 quirk for another HP dv7 (103c:30fc) with IDT
92HD71b* codec.

Reference: Novell bnc#461108
	https://bugzilla.novell.com/show_bug.cgi?id=461108

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

Added the missing __devexit annotation to wm8350_codec_remove():
  sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add SND_SOC_DAIFMT_DSP_A mode option.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.

Add dsp_a, dsp_b mode options
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Minor, just move a block of code to make next patch clearer.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Just at little cleanup of davinci_i2s_set_dai_fmt
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Document the current polarity choices.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add constants with a value of 0 to show more explicitly
what is being requested.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The capture with 44.1kHz on ca0106 seems to cause loud noises on
later playbacks, which doesn't support 44.1kHz.  A simple fix is to
disable 44.1kHz, as the "default" PCM with dsnoop is anyway only with
48kHz.

Reference: Novell bnc#447624
	https://bugzilla.novell.com/show_bug.cgi?id=447624Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added the missing card->private_data initialization that caused obvious
problems at PM.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Check the availability of ac97 at PM suspend/resume callbacks.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When no jack detection is available, the pins should be always
turned on since it can't be turned on/off dynamically via unsol
events.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixed "unused varible" warnings in patch_sigmatel.c that have been
introduced by the last changes.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.
Signed-off-by: NStanley Miao <stanley.miao@windriver.com>
Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Cc: stable@kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added probe_only module option to hd-audio driver.
This option specifies whether the driver creates and initializes the
codec-parser after probing.  When this option is set, the driver skips
the codec parsing and initialization but gives you proc and other
accesses.  It's useful to see the initial codec state for debugging.

The default of this value is off, so the default behavior is as same
as before.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the line_out has only one DAC and it's unique (i.e. not shared
by other outputs), assign a more reasonable and distinct mixer name
such as "Headphone" or "Speaker".
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The current auto-configuration code has several problems especially
for the new IDT codecs, e.g. wrong assignment of pins and DACs or
coupled volume for speaker and headphone.

This patch is a fairly large rewrite of the auto-configuration code.
Some remaks

- mic_switch and line_switch contain NIDs instead of bool
- dac_list isn't fixed for IDT 92HD* codecs now, they are all probed
- extra HP and speakers are stored in extra_dacs[].
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The previous commit re-enabled hp_nid setup for IDT92HD73*, but
it's unneeded indeed for Dell laptops that have multiple headphones.
Setting the extra hp_nid results in a non-working "Headpohne" mixer
control.  Thus hp_nid should be 0 for these dell models.

Also, the automatic addition of hp_nid should check whether it's
a dual-HP model or not.  For dual-HPs, the pins are already checked
by the early workaround.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The register and channel_id pair were twisted in the pm code...
Oh my.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added "IEC958 PCM Stream" controls for the per-stream IEC958 status
bits.  Using this instead of "IEC958 Default" is safer since the status
bits will be recovered to the default states after closing the PCM
stream.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>