This driver adds support for the wm9705 ac97 codec. The driver supports
audio input and output.
Signed-off-by: NIan Molton <ian@mnementh.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove dependency on sffsdr_fpga_set_codec_fs() when the
SFFSDR FPGA module is not selected.
Signed-off-by: NHugo Villeneuve <hugo@hugovil.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Modify the check for the mux type to also handle the
snd_soc_dapm_value_mux type in a same way as the snd_soc_dapm_mux.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Call the snd_soc_free_pcm and snd_soc_dapm_free when the
codec driver is unloaded.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.

This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.

[Edited to raise priority of error log message and document parameters.
 -- broonie]
Signed-off-by: NIan Molton <ian@mnementh.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

This patch adds a jack reporting interface to ASoC.  This wraps the ALSA
core jack detection functionality and provides integration with DAPM to
automatically update the power state of pins based on the jack state.

Since embedded platforms can have multiple detecton methods used for a
single jack (eg, separate microphone and headphone detection) the report
function allows specification of which bits are being updated on a given
report.

The expected usage is that machine drivers will create jack objects and
then configure jack detection methods to update that jack.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Cc: Ingo Molnar <mingo@elte.hu>
Cc: Thomas Gleixner <tglx@linutronix.de>
Acked-by: NTheodore Ts'o <tytso@mit.edu>
Acked-by: NMark Fasheh <mfasheh@suse.com>
Acked-by: NDavid S. Miller <davem@davemloft.net>
Cc: James Morris <jmorris@namei.org>
Acked-by: NCasey Schaufler <casey@schaufler-ca.com>
Acked-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

The soc_value_enum has been merged to soc_enum.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge the recently introduced soc_value_enum structure to the soc_enum.
The value based enums are still handled separately from the normal enum types,
but with the merge some of the newly introduced functions can be removed.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch allows you to define the mixer paths as having the same name as the
paths they represent.

This is required to support codecs such as the wm9705 neatly without extra
controls in the alsa mixer.
Signed-off-by: NIan Molton <ian@mnementh.co.uk>

For codecs that have both SPI and I2C support we need to ensure that we
don't try to make the codec driver built in when I2C is modular since
that won't link.  Do this by creating a helper variable which uses
conditional defaults to pick up the correct value for all combinations.

We don't need to do anything special for I2C-only codecs since a
conditional select passes on the full value for a tristate.
Reported-by: NIngo Molnar <mingo@elte.hu>
Tested-by: NIngo Molnar <mingo@elte.hu>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

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>

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>

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>

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>

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>

The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.

The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.

This driver was originally written by Liam Girdwood with further updates
from me.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This should never happen and it's helpful to identify the specific control
that failed when it does happen.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Rather than listing lots of architectures per line in Kconfig and
Makefile, causing merge conflicts all the time, have one per line
in alphabetical order.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>