The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.
in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF
Signed-off-by: Eric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for the JZ4740 internal codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.ul>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.
If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.
If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.
As safety measure do the register reset at remove time also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Restructure the codec power code in order to be able to hit
off when the codec is not in use.
Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.
According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.
Move all chip initialization code under chip_init, and do
it when the codec is initialized.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.
Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for the power down wrapper.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reset most of the codec registers to their chip reset
value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These scales should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These should be regular rather than linear scales.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When using MCLK is configured for 19.2 Mhz, clock slicer should be
enabled and HPPLL should be bypassed in clock path.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.
Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Log the values we're getting back from the DC servo and the values we
write to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This allows more flexible integration with subsystem features.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 6f3991152f.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control
It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.
The codec reset values are considered safe in all environmnts.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.
Provide different mixer control for the chips with correct
TLV mapping.
User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140
The way machine drivers are using this amplifier remained
the same.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).
After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).
The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.
There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, but we does not
need to execute the playback related configuration
2. Playback caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, and also we need
to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
still on.
Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.
Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.
The substream must be easily available in other places than
pcm_* callbacks.
Manage a pointer in _startup, and _shutdown for this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
OUTL/R are leftovers from the original driver, and they
are no longer needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.
We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */
/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */
/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */
The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).
The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The SYSCLK source is automatically managed when configuring the PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.
Bypass mode: FIFO is bypassed, report 0 as delay
Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.
Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.
Phase1: when we received the alarm threshold, but our workqueue has
not been executed (safeguard phase). Just count the played out
samples since ts1 and subtract it from the alarm threshold
value.
Phase2: During nSample burst (after writing to nSample register), count
the played out samples since ts1, count the samples received
since ts2 (in a burst). Estimate the FIFO depth using these and
alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
samples since ts1. Estimate the FIFO depth using the nSample
configuration and the alarm threshold value.
Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received
Interrupts are coming when DAC33 FIFO depth reaches upper threshold.
Phase1: Draining phase (after the burst), counting the played out
samples since ts1, and subtract it from the upper threshold
value.
Phase2: During burst operation. Using the pre calculated time needed to
play out samples from the buffer during the drain period (from
upper to lower threshold), move the time window to cover the
estimated time from the burst start to the current time.
Calculate the samples played out since lower threshold and also
the samples received during the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An index equal to the array size may not be accessed.
Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Bill Gatliff <bgat@billgatliff.com>
Acked-by: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.
Also, this patch fixes the Jive and Spitz machine.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.
Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise we may try to power down multiple times when the using
idle bias off and the driver is removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
