Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar', 'asoc/topic/rk3036' and 'asoc/topic/rockchip' into asoc-next

Merge remote-tracking branches 'asoc/topic/kcontrol', 'asoc/topic/max98357a' and 'asoc/topic/mtk' into asoc-next

Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/img' and 'asoc/topic/intel' into asoc-next

Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/dwc', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc' and 'asoc/topic/fsl-esai' into asoc-next

Merge remote-tracking branches 'asoc/topic/da7219', 'asoc/topic/dai-link' and 'asoc/topic/doc' into asoc-next

Merge remote-tracking branches 'asoc/topic/atmel-classd', 'asoc/topic/const' and 'asoc/topic/da7218' into asoc-next

Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/adsp', 'asoc/topic/ak4613' and 'asoc/topic/atmel' into asoc-next

Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus

As we want gpio_chip .get() calls to be able to return negative
error codes and propagate to drivers, we need to go over all
drivers and make sure their return values are clamped to [0,1].
We do this by using the ret = !!(val) design pattern.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

The stream is created whilst the compressed stream is opened and a
buffer is created when the DSP powers up. It is necessary at a point
once both the DSP has powered up and the the stream has been opened to
connect a stream to a buffer on the DSP. This is done in the trigger
callback as this is after the DSP has been powered and obviously the
stream must be open. Note that whilst the connect is currently trivial
it is expected that this will get more complex when support for multiple
buffers/streams per DSP is added.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add code that locates and initialises the buffer of compressed data on
the DSP if the firmware supported compressed data capture. The buffer
struct (wm_adsp_compr_buf) is kept separate from the stream struct
(wm_adsp_compr) this will allow much easier support of multiple
streams of data from the one DSP in the future, although support for
this will not be added in this patch chain.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Allow user-space to open a compressed stream, although no data will be
passed yet, as part of this adding the ability to define supported
capabilities per firmware and check these match the stream being opened.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

When working with the compressed framework occasionally vendors will
use esoteric internal audio formats. For such formats it doesn't really
make sense to add an new define to the kernel as their use is not
sufficiently general.

This patch adds a new define SND_AUDIOCODEC_BESPOKE that vendors can use
in such situations.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Register a platform driver for the CODEC and add DAIs that will be used
to connect a compressed record path for the voice control functionality.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge branch 'topic/cs47l24' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-adsp

Merge branch 'topic/arizona' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-adsp

PLL mode based on 32KHz master clock not supported in
AB silicon so remove support from the driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

HW can provide 1.6V micbias level as well the existing levels
already provided in the driver. This patch adds support for 1.6V
to the DT binding.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

In AB silicon, the internal LDO is not supported so remove
DT and driver references to this (digital voltage direct from
'VDD' supply)
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

In current AB silicon, BIAS_EN field is enabled by default in the
REFERENCES register, so the regmap default value should reflect
this.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

If codec probe() function fails after supplies have been enabled
it should really tidy up and disable them again. This patch updates
the probe function to do just that.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Previously Sidetone would operate only when capture to DAI was in
progress, due to DAPM path configuration. There is no reason why
this should not operate without DAI capture, so this patch updates
the DAPM path accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture")
in AC'97 mode so in this case fsl-asoc-card route map should
also be using them.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support.

According referance menual, "Limited support for the case when
output sampling rates is between 8kHz and 30kHz. The limitation
is the supported ratio (Fsin/Fsout) range as between 1/24 to 8."
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Sometimes the audio play can not be resumed after it is
suspended. Add snd_soc_pm_ops to execute power management
operations, then this issue is fixed.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add NULL test on call to devm_kzalloc.

The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression x;
identifier fld;
@@

* x = devm_kzalloc(...);
  ... when != x == NULL
  x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add NULL test on call to devm_kzalloc.

The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression x;
identifier fld;
@@

* x = devm_kzalloc(...);
  ... when != x == NULL
  x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add NULL test on call to devm_kzalloc.

The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression x;
@@

* x = devm_kzalloc(...);
  ... when != x == NULL
  *x
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

This patch adds Multi channel support on Renesas R-Car sound.
This patch is tested on Salvator-X board, but it can't use
Multi channel, because supported format is different between
codec chip and R-Car.
Thus, it was tested on board which doesn't mount codec chip,
with oscilloscope.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

This patch adds support to the codec driver to handle mic level
detect related IRQs, and report these to user-space using a uevent
variable.

The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to
user-space, if the mic level detect feature is enabled, and the
audio captured at the chosen mic(s) is above a certain threshold.
User-space can then handle the event accordingly (e.g. process
audio capture stream).

This method was chosen over ALSA control notification for a couple
of reasons:

 1) There's no requirement here for a control to read state from.
    The event is the only thing that's required and of interest.
 2) tinyalsa support for control notifications does not exist so on
    platforms using this over alsa-lib there is a need to add code
    to support this event handling.

Another possible option would be to use the standard Jack reporting
framework but this really does not fit for this kind of event.

Finally, use of the input device framework is not being encouraged,
due to difficulties in enabling apps to access input devices, so
this has also been avoided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

An external amp (if any) is connected to the external outputs of the SoC
of course, rather then directly to the internal amp.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

AFE is actually allowed to be turn on before configuration of DAIs
since each DAI has its own enabling control. Turn on/off AFE in
runtime resume/suspend to avoid AFE being shut down when closing a DAI
while other DAIs are still active.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>