The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.

Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.

The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.

Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.

The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.

This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)

SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).

This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: NDaniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Signed-off-by: NAlexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.

The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.

The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.

A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

According to the data sheet data is clocked out on the falling edge
and latched on the rising edge of the bit clock. While the left sample
is transmitted the word clock line is low.
Signed-off-by: NDaniel Glöckner <dg@emlix.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ak4535_remove() from sound/soc/codecs/ak4535.c calls
i2c_unregister_device() with a possibly null pointer.

This bug was found by smatch (http://repo.or.cz/w/smatch.git/).
Signed-off-by: NDan Carpenter <error27@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds machine specific code for the audio part of the Stretch
s6105 IP camera reference design.

The device uses the tlv320aic31(01) codec to generate the clock for
both I2S ports of the soc. While the master clock is generated by a
configurable PLL chip, the code assumes the factory default settings.

An additional kcontrol has been added to handle the special routing of
the board, connecting both HPLCOM and HPROUT to the same pin of the audio
jack. One of these should always be switched off.
Signed-off-by: NDaniel Glöckner <dg@emlix.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds a driver for the I2S interface found on Stretch s6000
family processors.
Signed-off-by: NDaniel Glöckner <dg@emlix.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds the needed code to be able to use 96KHz playback.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Without this the WM9705 driver fails badly when resuming.
Tested-by: NRussell King <linux@arm.linux.org.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Ensure that any AC97 devices that bind to the CODEC are below the
ASoC device in the device tree so the suspend and resume code can
figure out what order to handle them in.
Reported-by: NRussell King <linux@arm.linux.org.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

AC97 devices may have other drivers hanging off them directly so need to
have resumed when the resume function returns meaning that we can't defer
the resume - complete it immediately for them. Non-AC97 devices should
not have other drivers hanging directly off the ASoC devices.

We only really need the deferral for non-AC97 devices - it's there since
some I2C buses are very slow and non-AC97 codecs often have large numbers
of registers to restore and require delays to bring the codec up cleanly
leading to a substantial impact on overall resume time.
Reported-by: NRussell King <linux@arm.linux.org.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During
initial playback startup, this FIFO is keeping the DMA request active
until the FIFO is full.

So now if ALSA buffer size is smaller, DMA is looping around it while
filling up the HW FIFO, generating burst of interrupts as well and SW
doesn't have any change to fill enough data.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In case of duplex mode (capture and playback at the same time), the second
stream has to have the same parameters (rate, sample size) as the already
running stream.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Optimize the display of SSI statistics in the Freescale MPC8610 sound driver
to display the status count only of the interrupts that were actually enabled.
Previously, it would display the counts of all SISR status bits, even those
that were not enabled.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

the variable gsr_bit is set in isr. It is however set to 0 and interrupts are
disabled prior to reset. Hence it doesn't make a lot of sense to show the
content of gsr_bit in case of a reset timeout.
Signed-off-by: NLuotao Fu <l.fu@pengutronix.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove the delay from the trigger function in the Freescale MPC8610 sound
driver when capture is started.  This delay was used to ensure that the DMA
controller was active when ALSA call the .pointer function to request a
DMA transfer status.  A better approach is for the .pointer function to detect
that DMA has not started, and return zero instead.  This change eliminates
the need for the delay.

Also add some related code to check for a DMA programming error, and report
XRUN if it occurs.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

HTC Magician has a Philips UDA1380 codec connected via
SSP1 (playback) and I2S (capture).
There is a flip-flop between the SSP frame clock output
and the codec's word select input pin. To make the codec
see proper I2S input, the SSP has to send two frames per
sample.
Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Now magician and similar boards can use network mode with only one
active slot to explicitly set 16 bit frame width, even for S16_LE
stereo sound.
Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix the wrong device pointer passed to dev_err().
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.

Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:

- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.

Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The commit 14fa43f5 ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister().  This patch adds same condition for
soc_ac97_dev_unregister().  Without this fix, kernel crashes when
unloading an asoc driver.
Signed-off-by: NAtsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

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

CC      sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.

This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.
Signed-off-by: NRobert Jarzmik <rjarzmik@free.fr>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This also simplifies the code a bit.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>