Add support for 24 bit audio (with S32_LE msbits 24).
The reason to limit the msbits to 24, is that the FIFO
can be configured for 16 or 24 bit layout.
It is unknown how the codec would downsample from 32 to
24 bit, if the interface is configured to receive 32
bit data.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Change the structure of FIFO handling in order to
pave the way for adding 32/24 bit audio support.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The manual FIFO configuration was the first version to enable
the use of the FIFO in the codec.
It had served it's purpose as debugging aid, but the automatic
FIFO configuration is much safer to use.
The removal of the manual controls, and configuration makes
it easier to add new features for the codec later, since
the manual mode neded different ways to calculate, and
protect against misconfiguration.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Make the phoenix HS and HF drivers use the new DAPM driver
widget in order to guarantee power ON/OFF order sequence.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds McBSP support for the OMAP4 CPU
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add ramp functions for the headset and handsfree outputs
in order to reduce the pops during power on/off sequences.

In order to give more control to volume ramp, step size and delay
between steps can be specified.

The patches are based on wm8350 implementation from Liam
Girdwood.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: NMisael Lopez Cruz <misael.lopez@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Updated default values to improve power consumption.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Gain for LineInAmp Right uses LINEGAIN[5:3], which means that
offset for right channel should be 4.
Signed-off-by: NMisael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Some gains were incorrectly configured for dB values.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

After coming back from suspend, the timeout waiting for Phoenix
chip to complete its power up sequence is not enough, which leaves
the codec cache value for some registers in an outdated state.

Increase the timeout value to wait for the power up sequence
to correclty complete.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Enable plug detection interrupt mask in order to get headset
PLUGINT/UNPLUGINT interrupts.
Signed-off-by: NMisael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

On Phoenix 1.1, the INTID register default value is an invalid
one, causing the interrupt handler to think the phoenix power on
sequence is ready before it actually finishes.

This causes some i2c errors when trying to configure twl.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Phoenix 1.1 supports automatic power on sequence, a
verification is added to use it with new revision of
the chip.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The mux control has 4 elements not 3
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The twl6040 can support more sample rates other than 88.2 and 96k.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch moves all the PCM error handling for clock config
out of trigger() and startup() and into prepare().
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch restores the CODEC bias level at resume().
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds support for the twl6040 headset and handset
MUX controls.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Multiples interrupts can be received. The irq handler is modified
to attend all of them.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Update the codec to use the new twl core register macros
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Use jack framework to enable detection for the headset microphone
and stereo output in the sdp4430.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NDavid Anders <x0132446@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds support for reporting twl6040 headset and
handset jack events.

The machine driver retrieves and report the status  through
twl6040_hs_jack_detect.

A workq is used to debounce of the irq.
Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: NMisael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

twl4030 series of codecs supports S32_LE with msbits=24.
Replace the S24_LE with S32_LE format, and add constraint
for 24msbit in case of 32 S32_LE format.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Remove redundant parentheses/spaces in the use of the sizeof
operator.
Signed-off-by: NDimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This helps ensure that the ramp logic is reset when powering back up.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.

Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

soc_unregister_ac97_dai_link() takes a CODEC as an argument, not a
rtd like the registration function, so give it what it's looking for.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Fix "ASoC: Fix bias power down of non-DAPM codec" for 3.6.37 will cause a
build error when merging into ASoC for-2.6.38. Fix the issue by doing a
change that commit ce6120cc "ASoC: Decouple DAPM from CODECs" would do.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.

Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

delayed_work was moved to dapm in the commit
ce6120cc
    ASoC: Decouple DAPM from CODECs
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The compiler really ought to have been warning about unreferenced
variables...
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

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>

Provide the user with a boolean control then automatically select
the deemphasis filter most closely matching the sample rate.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

We're already flagged as using symmetric rates so we don't need to
have a custom implementation.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

If the following scenario has been followed:
1. Enable analog bypass
amixer sset 'Analog Left Bypass' on
amixer sset 'Analog Right Bypass' on

2. Start playback
aplay -fdat -d3 /dev/zero

After the playback stopped (3 sec), and the soc timeout (5 sec),
the digital parts of the codec will remain powered up.
This means that the DAI clocks are continue to run, the
oscillator remain operational, etc.

Use the SND_SOC_DAPM_POST_PMD widget to get notification
about the stopped stream, and power down the digital
part of the codec.
If the analog bypass is enabled, than the codec will remain in
BIAS_ON level, and things will work correctly.
In case, if the bypass is disabled, than the codec will
fall to BIAS_STANDBY than to BIAS_OFF level, as it used
to.

The digital part of DAC33 is initialized at every stream start
(DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec)
will have working DAI.
When the codec is coming out from BIAS_OFF, the full power-up
sequence followed by the same DAPM_PRE widget event will power up
the digital part.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on
H1940.
Signed-off-by: NVasily Khoruzhick <anarsoul@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The oversampling rate of the DAC and ADC can be controlled to optimise
for either low power consumption or maximum performance.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Tune the FLL gain for optimal performance according to evaluation
results.
Signed-off-by: NMario Becroft <mb@gem.win.co.nz>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>