Merge remote-tracking branches 'asoc/topic/ad1980', 'asoc/topic/adsp', 'asoc/topic/ak4104', 'asoc/topic/ak4642', 'asoc/topic/alc5623', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/cache' into asoc-next

ASoC: Intel fixes for v3.15

This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be.  Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.

# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

ASoC: Driver fixes for v3.15

A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.

# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

ASoC: Core fixes for v3.15

A few things here:

 - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
   have audio paths which shouldn't be present causing spurious powerups
   and potential audible issues for users.
 - Ensure the suspend->off transition doesn't have spurious transitions
   to prepare added to the sequence.
 - Fix incorrect skipping of PCM suspension for active audio streams.
 - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
   this and Timur no longer has the boards that he was using.

# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.

Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>

Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.

Also fixes the following warnings from smatch:
	sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
	sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
	sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.

While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Make sure the cache is fully flushed at resume time.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

The return statement was not indented correctly.  I lined up the
condition a bit as well.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus

Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

Currently unused, this is done to let the driver distinguish between
the different supported codec types in later patches.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

ALC and ALC Zero crossing detection has been enabled unconditionally.
Add controls for this.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.

To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.
Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

Commit 10df3509 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:

baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware

Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.

So this patch moves PCRC and PRRC settings to the end of hw_params().
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.

So this patch adds an extra check in the code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.

So this patch fixes the condition here and adds one line comments to
make the purpose here clear.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

This was initially removed in commit 6423c187 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

Suspend/resume support for the atmel-pcm-pdc driver was broken in commit
f0fba2ad ("ASoC: multi-component - ASoC Multi-Component Support"). It
essentially reverted the modifications done in commit 10cab262 ("ASoC: Change
how suspend and resume obtain the PCM runtime"). The suspend and resume handlers
at the beginning check if dai->runtime is not NULL, but dai->runtime is always
NULL, hence the code never runs. Considering that nobody noticed any problems in
the last 4 years since the code was broken and that the driver does not set
SNDRV_PCM_INFO_RESUME, which means applications are expected to stop and restart
the audio stream during suspend/resume, it is probably safe to assume that his
code is not needed and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.

This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

Commit commit de9ba98b ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

If we find a widget who's stream name matches the name of a DAI widget then
thats the one it should be connected to. Based on the widget id we can say in
which direction the path should be. No need to go back to the DAI and check the
stream names.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>

We already know the widgets we want to connect, so use snd_soc_dapm_add_path()
instead of snd_soc_dapm_add_route() in snd_soc_dapm_new_pcm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>