module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the process of moving over from static board files to the device
tree, reset pins of peripheral reset pins should be handled by their
corresponding drivers.
Add a reset-gpio DT property to the cs4270 driver, and de-assert it
before probing the chip. The logic could be augmented some day to
re-assert it when codec is put to suspend.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There's never any need to cast away from void.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Lee Jones <lee.jones@linaro.org>
It's currently only used in one place but another user will be added
shortly and there's an argument it's clearer anyway.
Also add support for readback in mode 1, though it's not currently used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since jackdet only reports deltas it won't generate an interrupt on startup
when a jack is not present. This doesn't make a difference to userspace
but does mean we don't generate a notification via the internal notifier
chains. Fix that by scheduling a work to poll the chip after the clock is
enabled. Use an extremely large timeout since there's no urgency and we
don't want to report a false negative.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The FLLs in the WM8994 series devices can be started without any reference
being supplied, mainly for use in analogue bypass cases. Implement support
for this mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
version.h header file inclusion is no longer needed for this file.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few updates for issues discovered during the merge window, the main
one being the fix for the issues with defaulting to use of regmap
without properly checking if there was I/O in place already.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-3.6
ASoC: Additional updates for 3.6
A few updates for issues discovered during the merge window, the main
one being the fix for the issues with defaulting to use of regmap
without properly checking if there was I/O in place already.
A bunch of small fixes for ASoC, mainly against regressions due to the
defaulting regmap i/o, in addition to a HD-audio fixup.
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A bunch of small fixes for ASoC, mainly against regressions due to the
defaulting regmap i/o, in addition to a HD-audio fixup."
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: core: Fix check before defaulting to regmap
ALSA: hda - Support dock on Lenovo Thinkpad T530 with ALC269VC
ASoC: wm8962: Allow VMID time to fully ramp
ASoC: AC97 doesn't use regmap by default
ASoC: sgtl5000: enable VAG_POWER for LINE_IN
ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements
ASoC: omap: Add missing modules aliases to get sound working on omap devices
sound: tegra_alc5632: Adjust to of_get_named_gpio() change
sound: tegra_wm8903: Adjust to of_get_named_gpio() change
ASoC: mc13783: Provide codec->control_data
ASoC: ux500: Include the correct header files
ASoC: wm8994: Hold runtime PM reference while handling mic and jack IRQs
ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe
ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work
ASoC: mxs-saif: fix clock prepare and enable unbalance issue
ASoC: wm8994: Ensure there are enough BCLKs for four channels
Since commit 38cbf9598f
("ASoC: core: Try to use regmap if the driver doesn't set up any I/O")
any ASoC codec which doesn't set codec::control_data is assumed
to use regmap. That doesn't work with AC97 so this workaround
sets the codec::control_data member to a random value to restore
proper behaviour.
Tested with WM9712.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
LINE_IN also needs VAG_POWER on or we may hear noise when directly
route LINE_IN to Headphone Mux.
Tested on imx28evk.
Signed-off-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If codec->control_data is not populated SoC Core assumes we want to
use regmap, which fails catastrophically, as we don't have one:
Unable to handle kernel NULL pointer dereference at virtual address 00000080
pgd = c0004000
[00000080] *pgd=00000000
Internal error: Oops: 17 [#1] PREEMPT SMP ARM
Modules linked in:
CPU: 1 Not tainted (3.5.0-rc6-00884-g0b2419e-dirty #130)
PC is at regmap_read+0x10/0x5c
LR is at hw_read+0x80/0x90
pc : [<c01a91b8>] lr : [<c0216804>] psr: 60000013
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have support for a few new drivers:
- Samsung s2mps11
- Wolfson Microelectronics wm5102 and wm5110
- Marvell 88PM800 and 88PM805
- TI twl6041
We also have our regular driver improvements:
- Device tree and IRQ domain support for STE AB8500
- Regmap and devm_* API conversion for TI tps6586x
- Device tree support for Samsung max77686
- devm_* API conversion for STE AB3100
Besides that, quite a lot of fixing and cleanup for mc13xxx, tps65910,
tps65090, da9052 and twl-core.
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Merge tag 'mfd-3.6-1' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6
Pull MFD bits from Samuel Ortiz:
"We have support for a few new drivers:
- Samsung s2mps11
- Wolfson Microelectronics wm5102 and wm5110
- Marvell 88PM800 and 88PM805
- TI twl6041
We also have our regular driver improvements:
- Device tree and IRQ domain support for STE AB8500
- Regmap and devm_* API conversion for TI tps6586x
- Device tree support for Samsung max77686
- devm_* API conversion for STE AB3100
Besides that, quite a lot of fixing and cleanup for mc13xxx, tps65910,
tps65090, da9052 and twl-core."
Fix up mostly trivial conflicts, with the exception of
drivers/usb/host/ehci-omap.c in particular, which had some
re-organization of the reset sequence (commit 1a49e2ac96: "EHCI:
centralize controller initialization") that clashed with commit
2761a63945 ("mfd: USB: Fix the omap-usb EHCI ULPI PHY reset fix
issues").
In particular, commit 2761a63945 moved the usb_add_hcd() to the
*middle* of the reset sequence, which clashes fairly badly with the
reset sequence re-organization (although it could have been done inside
the new omap_ehci_init() function).
I left that part of commit 2761a63945 just undone.
* tag 'mfd-3.6-1' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6: (110 commits)
mfd: Ensure AB8500 platform data is passed through db8500-prcmu to MFD Core
mfd: Arizone core should select MFD_CORE
mfd: Fix arizona-irq.c build by selecting REGMAP_IRQ
mfd: Add debug trace on entering and leaving arizone runtime suspend
mfd: Correct tps65090 cell names
mfd: Remove gpio support from tps6586x core driver
ARM: tegra: defconfig: Enable tps6586x gpio
gpio: tps6586x: Add gpio support through platform driver
mfd: Cache tps6586x register through regmap
mfd: Use regmap for tps6586x register access.
mfd: Use devm managed resources for tps6586x
input: Add onkey support for 88PM80X PMIC
mfd: Add support for twl6041
mfd: Fix twl6040 revision information
mfd: Matches should be NULL when populate anatop child devices
input: ab8500-ponkey: Create AB8500 domain IRQ mapping
mfd: Add missing out of memory check for pcf50633
Documentation: Describe the AB8500 Device Tree bindings
mfd: Add tps65910 32-kHz-crystal-input init
mfd: Drop modifying mc13xxx driver's id_table in probe
...
If codec->control_data is not provided, the following crash happens:
Unable to handle kernel NULL pointer dereference at virtual address 00000078
pgd = 80004000
[00000078] *pgd=00000000
Internal error: Oops: 5 [#1] SMP ARM
Modules linked in:
CPU: 0 Tainted: G W (3.5.0-next-20120725+ #1263)
PC is at regmap_read+0x18/0x64
LR is at hw_read+0x50/0x98
pc : [<802bcd90>] lr : [<803cad18>] psr: 60000013
...
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
twl6040 ES1.1 and ES1.2 have the same revid (0x01).
ES1.3 of twl6040 REVID is 0x02.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Ensures that we don't interact badly with the power management framework,
especially in the cases where we're doing deferred work or we're using a
direct GPIO for these signals.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There's a driver bug that sgtl5000 dapm widget kcontrols do not work.
e.g. can not select capture mux with amixer tool(no error info prompted).
The root cause is that we still call snd_soc_dapm_new_widgets in
codec driver probe function afer converting to table based widgets.
This will cause the card dapm widgets are instantiated before the
dapm_routes are registered.
Then, no available dapm widget pathes can be found during instantiation
which finally will cause soc_dapm_mux_update_power to fail(can not find
correct path with kcontrol) in snd_soc_dapm_put_enum_double function.
Here we remove the unneeded snd_soc_dapm_new_widgets in codec probe
and let the soc core to handle the register sequence properly.
Then we can fix above issue.
Signed-off-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise if someone tries to use all four channels on AIF1 with the
device in master mode we won't be able to clock out all the data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
The conversion of the core driver to irqdomains means that we don't need
and irq_base to have working interrupts so use wm8994_request_irq() to
deal with looking up the interrupt number for the micdet IRQ.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure robust startup of the part by going through the reset procedure
prior to resyncing the full register cache, avoiding potential intermittent
faults in some designs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The WM5110 is a highly integrated low power audio subsystem for
smartphones, tablets and other portable audio devices. It combines
an advanced DSP feature set with a flexible, high performance audio
hub CODEC.
This patch adds the audio CODEC driver for the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GCC complains that "ret" is uninitialized here.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-By: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some Arizona chips have a higher frequency for the FLL VCO, support this
in the common code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds input selection of main codec clock - from what pin.
Both registers set same value since codec uses clock divider or pll at one time.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds AGC target level and times settings for TLV320AIC3x.
Enums uses small arrays of two channels left and right since it uses different registers.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds missing deemphasis switch.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The analogue PGA shifts were used; this makes no practical difference as
the values are the same.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix spelling mistake in "High-Performance" option of twl6040 power mode.
Signed-off-by: Simon Wilson <simonwilson@google.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If an audio interface is configured to use ASYNCCLK then update the
asynchronous sample rate rather than one of our primary sample rates.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow any BCLK which can be divided down to generate LRCLK, not just the
lowest possible BCLK to clock out the samples.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the user to select which of the system clocks each AIF is referenced
to and constran the DAI to the set of frequencies which can be generated
from that clock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They're the rates for the BCLK, not for the sample rate, so rename so that
we don't confuse ourselves.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The STA529 is a digital stereo class-D audio amplifier. It includes an
audio DSP, an ST proprietary high-efficiency class-D driver and CMOS
power output stage. It is intended for high-efficiency
digital-to-power-audio conversion for portable applications.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds missing register default values to cache.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds register and bit shift definitions in header file.
Changes are for TLV320AIC310x based on data sheet.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Leave it up to the machine driver to disable accessory detection if
desired, the common pattern is to have accessory detection be a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In sound/soc/codecs/tlv320aic3x.c
data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
data | (pll_p << PLLP_SHIFT));
In the above code, pll-p value is OR'ed with previous value without
clearing it. Bug is not seen if pll-p value doesn't change across
Sampling frequency.
However on some platforms (like AM335x EVM-SK), pll-p may have different
values across different sampling frequencies. In such case, above code
configures the pll with a wrong value.
Because of this bug, when a audio stream is played with pll value
different from previous stream, audio is heard as differently(like its
stretched).
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Without this very high BCLKs will be configured incorrectly.
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Otherwise we fail to link when building as modules due to multiple
init/exit functions.
Reported-by: Fengguang Wu <wfg@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the support for spdif in audio codec.
Signed-off-by: vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seems that the code duplication was added at a merge operation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This gets the registers set up as early as possible, mainly useful for the
GPIOs to ensure that they're in the correct mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is only one caller and this allows us to cleanly leave the CODEC
with the internal LDO powered down which is the default state we're
looking for and means that we can robustly disable the register cache
only when we either disable the LDO or power down the external
regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Get it done as early as possible, it's neater and minimises the time the
pins aren't configured as requested.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Even though the WM8904 is able to use idle_bias_off during both probe
and resume we were needlessly leaving the device in standby mode. Instead
power the device down as soon as we've confirmed that we can talk to it
and don't manage the bias level at all over suspend and resume, the core
will take us down to our minimum power level.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Merge tag 'v3.5-rc4' into for-3.6
Linux 3.5-rc4 contains some bug fixes which overlap with new features.
The WM5102 is a highly-integrated low-power audio system for smartphones,
tablets and other portable audio devices based on the Arizona platform.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Wolfson Arizona series of audio hub CODECs can share a large amount
of their driver code as the result of a common register map. This patch
adds some of this core support, providing a basis for the initial WM5102
audio driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a missing mutex_unlock() here. The cleanup path also has more
debug output.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a leak if snd_soc_register_codec fails.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GPLv2 -> GPL v2
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also convert to use update_bits() while we're at it. No great need to do
this, it's just a bit neater to do as much as possible in the I2C probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
There's no reason to defer requesting of the interrupt until the CODEC
probe and doing so results in more work if we hit an error as we'll have
registered the CODEC with the core. It's neater to acquire as many of the
resources we'll need as we can in the bus probe function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
There's no urgent need for the interrupt handler to use the ASoC I/O
functions and it'll support a further move in where we request the
interrupt so call the regmap APIs directly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
Ensure that the device pins are configured as soon as possible by moving
the pin configration (including MICBIAS) into the I2C probe() function.
This had been done in the CODEC probe() function when we were relying on
the ASoC register I/O code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
This patch adds support for Dialog DA732x audio codecs.
Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They aren't marked as readable and the feature is never used so they'll
never get referenced.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we don't have control over the LDO but do have control over the other
regulators then we may end up trying to write to a powered off device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're using demv_regulator_bulk_get() so don't need to manually free and
this is in the CODEC driver not the I2C driver anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should be using the regmap API consistently for all the cache only
configuration and we should be going cache only before we power down
the supplies.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We no longer have a flat ASoC cache so can't peer directly into the array
any more but should instead use the register I/O functions to update the
cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org (v3.4)
Rather than having varying timeouts depending on the transition always
use a 4s timeout. This provides better diagnostics for clocking errors
and ensures compatibility with current calibration firmwares.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should set "isabelle_regmap" before using it. GCC complains.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Volume updates may not be acted upon if there is no clock applied when
the volume update is written. Ensure this doesn't happen by writing out
registers with volume updates after we enable each of the clocks.
There are more registers updated than before as previously we were
relying on wm_hubs to set those for controls it manages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Ensure that all the actions get taken at appropriate times by calling the
_PRE and _POST events for the aifNclk_ev functions explicitly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The core fills in some blanks which makes it annoying to do the right thing
and constify the calls in the core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With devm_ APIs regmap_exit() not needed, so remove regmap_exit().
Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Support TI Isabelle Audio driver
The Isabelle Audio IC is a complete low power high fidelity CODEC with integrated
ADCs, DACs, decimation and interpolation filters, PLL, and power providers. This
device supports 2 analog and 2 digital microphone channels, a mono earpiece driver,
stereo class G headphone drivers with ultra low power and best SNR in the industry,
stereo Class D speaker drivers, and 2 high performance Line drivers.
The below patch is a basic driver code for TI Isabelle audio codec. The
functionalities like headset detection, etc., will be included incrementally
in the up-coming patches.
Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
version.h header file is no longer required.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
version.h header file is no longer needed.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output
gain control as a playback control. Fix this by changing the name as
"Class-D Playback Volume".
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows the GPIOs to be available as soon as the I2C device has
probed, which in turn enables machine drivers to request the GPIOs in
their probe(), rather than deferring this to their ASoC machine init
function, i.e. after the whole sound card has been constructed, and
hence the WM8903 codec is available.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
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Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute updates
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
The major thing here is the addition of some helpers to factor code out
of drivers, making a fair proportion of regulators much more just data
rather than code which is nice.
- Helpers in the core for regulators using regmap, providing generic
implementations of the enable and voltage selection operations which
just need data to describe them in the drivers.
- Split out voltage mapping and voltage setting, allowing many more
drivers to take advantage of the infrastructure for selectors.
- Loads and loads of cleanups from Axel Lin once again, including many
changes to take advantage of the above new framework features
- New drivers for Ricoh RC5T583, TI TPS62362, TI TPS62363, TI TPS65913,
TI TWL6035 and TI TWL6037.
Some of the registration changes to support the core refactoring caused
so many conflicts that eventually topic branches were abandoned for this
release.
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Merge tag 'regulator-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator
Pull regulator updates from Mark Brown:
"The major thing here is the addition of some helpers to factor code
out of drivers, making a fair proportion of regulators much more just
data rather than code which is nice.
- Helpers in the core for regulators using regmap, providing generic
implementations of the enable and voltage selection operations which
just need data to describe them in the drivers.
- Split out voltage mapping and voltage setting, allowing many more
drivers to take advantage of the infrastructure for selectors.
- Loads and loads of cleanups from Axel Lin once again, including many
changes to take advantage of the above new framework features
- New drivers for Ricoh RC5T583, TI TPS62362, TI TPS62363, TI
TPS65913, TI TWL6035 and TI TWL6037.
Some of the registration changes to support the core refactoring
caused so many conflicts that eventually topic branches were abandoned
for this release."
* tag 'regulator-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator: (227 commits)
regulator: tps65910: use of_node of matched regulator being register
regulator: tps65910: dt: support when "regulators" node found
regulator: tps65910: add error message in case of failure
regulator: tps62360: dt: initialize of_node param for regulator register.
regulator: tps65910: use devm_* for memory allocation
regulator: tps65910: use small letter for regulator names
mfd: tpx6586x: Depend on regulator
regulator: regulator for Palmas Kconfig
regulator: regulator driver for Palmas series chips
regulator: Enable Device Tree for the db8500-prcmu regulator driver
regulator: db8500-prcmu: Separate regulator registration from probe
regulator: ab3100: Use regulator_map_voltage_iterate()
regulator: tps65217: Convert to set_voltage_sel and map_voltage
regulator: Enable the ab8500 for Device Tree
regulator: ab8500: Split up probe() into manageable pieces
regulator: max8925: Remove check_range function and max_uV from struct rc5t583_regulator_info
regulator: max8649: Remove unused check_range() function
regulator: rc5t583: Remove max_uV from struct rc5t583_regulator_info
regulator: da9052: Convert to set_voltage_sel and map_voltage
regulator: max8952: Use devm_kzalloc
...
Introduce codec for HDMI. At the moment, this is a dummy codec. In the
future it will parse the EDID to modify the supported parameters, such
as the number of channels and the sample rates. At the moment, it blindly
supports all the sample rates and audio channels described in the HDMI
1.4a specification.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few last-minute regression fixes for 3.4 final kernel.
All trivial, and Cc'ed to stable kernel.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few last-minute regression fixes for 3.4 final kernel. All trivial,
and Cc'ed to stable kernel."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Fix AIF2ADC power down
ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops
ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
aic3x_set_headset_detection() isn't made available outside the driver or
referenced within the driver which sparse notices and complains about.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for irq_domain support change the code to the not switch
based on the irq number. This actually makes things simpler, if slightly
repetitive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the devm_ versions of the regmap and memory allocation functions,
saving some error handling code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Makes sparse happy and avoids polluting the global namespace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
We need to read the real register values
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're trying to remove all usage of the ASoc level cache and I/O code and
for a device like this with a pretty sparse register map the rbtree cache
is a better idea anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the machines uses the gain ramp possibility for HS/HF.
This code path is mostly unused and it does not reduces the pop
noise on the output (it alters it to sound a bit different).
The preferred method to reduce pop noise is to use ABE.
Remove the gain ramp, and related features form the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As good as nothing exciting here; just a few trivial fixes for
various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
This patch improves playback quality for few sample rates like 8000 and
11025 Hz.
This also fixes an issue observed during testing of pll slave mode. Due
to the issue, on some rare occasions there was no sound output for first
time playback after system boot, though all subsequent playbacks were
fine. It was mainly because of the sequence in which SRM bit was
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the conversion to module_init_i2c() the original open coded module
exit function was left. Remove it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.
In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.
Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A workaround for an ASUS laptop and a few ASoC changes;
most of the commits are tagged for stable, too.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A workaround for an ASUS laptop and a few ASoC changes; most of the
commits are tagged for stable, too."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Improve sequencing of AIF channel enables
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
ASoC: core: Fix card RTD count for deferred probe.
ASoC: cs42l73: don't use negative array index
ASoC: dapm: Ensure power gets managed for line widgets
If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have never really updated that version number and probably never will, so
just remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.
Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
Complete the separation of the twl6040 from the twl core since
it is a separate chip, not part of the twl6030 PMIC.
Make the needed Kconfig changes for the depending drivers at the
same time to avoid breaking the kernel build (vibra, ASoC components).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in
sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using
the (negative) return value as array index on the very next line of
code - that's bad.
Catch the negative return value and propagate it to the caller (which
checks for it) and things are a bit more sane :-)
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/wm9705.c: In function 'ac97_prepare':
sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/ac97.c: In function 'ac97_prepare':
sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function)
sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function)
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Later WM8994 class devices can bypass the FLL from BCLK. Do this
automatically when the FLL input and output frequencies match up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
release_firmware() does its own NULL ptr testing, it's redundant to
also test before calling it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than adding new arguments to regulator_register() every time we
want to add a new bit of dynamic information at runtime change the function
to take these via a struct. By doing this we avoid needing to do further
changes like the recent addition of device tree support which required each
regulator driver to be updated to take an additional parameter.
The regulator_desc which should (mostly) be static data is still passed
separately as most drivers are able to configure this statically at build
time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt':
sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit
constant conversion [-Woverflow]
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following warning during kernel boot:
0-000a: 850 <--> 1600 mV at 1200 mV normal
0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
