If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

In case we have a pending completion, for example due to a problem with
the input clock which got corrected after we timed out.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

Ensure that we don't spuriously trigger early.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

This should never happen with level triggered IRQs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

The WM8962 features a DSP providing a number of signal processing
features including HD Bass and Virtual Surround Sound (VSS).  Enable
initial support for this, allowing users to enable and disable the
algorithms using the default coefficient sets.  Further patches will
add support for runtime configuration of the DSP coefficients.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

We get called once per DAPM context but only need to run once. When DAPM
was serialized this was a series of noops but now it can run in parallel
we need to take proper care.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

This occurs frequently if we are in edge triggered mode as we must poll the
interrupt status register until we get no more interrupts so it's worth
the effort - it means we skip writing null acknowledgements to the chip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

For non-audio uses like accessory detection we can use a lower quality,
unregulated microphone bias, saving a little power. As the hardware can
manually enable and disable the biases we can select regulating mode
automatically with supply widgets connected to the biases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

There's no need for separate widgets for the enables (as the map already
shows).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

Make AIFCLK supply the record paths otherwise record will not work unless
there is a simultaneous playback.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

This patch includes below small fixes:

1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove().
2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(),
   as we will call regulator_bulk_free() in sta32x_remove().
3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove.
   The i2c core will set the clientdata to NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Johannes Stezenbach <js@sig21.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Fix "error: too few arguments to function 'ams_delta_set_bias_level'"
build errors in ams-delta.c that were introduced after commit d4c6005f ("ASoC:
Add context parameter to card DAPM callbacks") by adding dapm context
to ams_delta_set_bias_level calls.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The ASoC core tries to not enforce symmetric rates when
two streams open simultaneously. It does so by checking
rtd->rate being zero. This works exactly once after booting
because it is not set to zero again when the streams close.
Fix this by setting rtd->rate when no active stream is left.

[This leads to lots of warnings about not enforcing the symmetry in some
situations as there's a race in the userspace API where we know we've
got two applications but don't know what rates they want to set.
-- broonie ]
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The previous patch to fsl_dma.c ("fix initialization of DMA buffers")
left behind an unused local variable that causes a build warning.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>

The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when
the device is opened, which means that the /proc/interrupts entry for
the SSI exists only during playback or capture.  This also meant that
the user won't know that the IRQ number is wrong until he tries to use
the device.  Instead, we should claim the IRQ when the device is probed.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add a machine driver to support the EVAL-ADAU1373 board connected to a
Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds support for the Analog Devices ADAU1373 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ALSA doesn't really have good mechanisms for dealing with these so we just
log them - the hardware already has automatic shutdown support.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

Ensure we have access to all the advanced DSP functinality offered by the
WM8962 by running the system clock at 512fs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Currently it is only possible to route one source per switch into a mixer.
This patch modifies the code, so that it is possible to route multiple sources
into a mixer via the same switch. One use-case for this is routing a stereo
channel pair into a mono-mixer via the same switch.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Pull control is availalbe for WM8994 AIF2, generally disabled as part of
the GPIO configuration in order to save power after system startup. As on
newer devices in the series there is no GPIO functionality on these pins
this will happen less naturally so have the driver disable the pulls as the
AIF is probed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The hardware documentation uses revision numbers starting at 1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

There is exactly one match or no match at all during the for loop iteration,
thus we can break from the for loop once a match is found.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Remove unused variable 'dai' to eliminate below warning.

  CC      sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_new':
sound/soc/nuc900/nuc900-pcm.c:321: warning: unused variable 'dai'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Since commit b8c0dab9
"ASoC: core - PCM mutex per rtd",
the global pcm_mutex is not being used any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Since commit aea170a0
"ASoC: soc-cache: Add reg_size as a member to snd_soc_codec",
the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Since commit d779fce5
"ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache",
the codec_drv pointer variable is not used any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Remove the bodge for ad193x.

Conflicts:
	sound/soc/soc-io.c

asoc cache layer can't support this kind of spi registers well.
remove cache support and read/write registers directly
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>