This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.
At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (417 commits)
MAINTAINERS: EB110ATX is not ebsa110
MAINTAINERS: update Eric Miao's email address and status
fb: add support of LCD display controller on pxa168/910 (base layer)
[ARM] 5552/1: ep93xx get_uart_rate(): use EP93XX_SYSCON_PWRCNT and EP93XX_SYSCON_PWRCN
[ARM] pxa/sharpsl_pm: zaurus needs generic pxa suspend/resume routines
[ARM] 5544/1: Trust PrimeCell resource sizes
[ARM] pxa/sharpsl_pm: cleanup of gpio-related code.
[ARM] pxa/sharpsl_pm: drop set_irq_type calls
[ARM] pxa/sharpsl_pm: merge pxa-specific code into generic one
[ARM] pxa/sharpsl_pm: merge the two sharpsl_pm.c since it's now pxa specific
[ARM] sa1100: remove unused collie_pm.c
[ARM] pxa: fix the conflicting non-static declarations of global_gpios[]
[ARM] 5550/1: Add default configure file for w90p910 platform
[ARM] 5549/1: Add clock api for w90p910 platform.
[ARM] 5548/1: Add gpio api for w90p910 platform
[ARM] 5551/1: Add multi-function pin api for w90p910 platform.
[ARM] Make ARM_VIC_NR depend on ARM_VIC
[ARM] 5546/1: ARM PL022 SSP/SPI driver v3
ARM: OMAP4: SMP: Update defconfig for OMAP4430
ARM: OMAP4: SMP: Enable SMP support for OMAP4430
...
Follow-up fix needed since "ASoC: pxa-ssp.c fix clock/frame invert".
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the recent changes to the DAPM power checks it has become important
to explicitly instantiate all widgets but some drivers were forgetting
to do that. Since everything needs to do it add a call to instantiate
them immediately before the card registration - it does no harm when it
is called repeatedly and saves work in drivers.
Tested-by: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses. However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.
This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended. At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be enabled
below 24kHz and not enabled above that so remove the user visible
control for this and do it autonomously in the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be
enabled below 24kHz and not enabled above that so remove the
user visible control for this and do it autonomously in the
driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be
enabled below 24kHz and not enabled above that so remove the
user visible control for this and do it autonomously in the
driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Blackfin submission was done as a patch against a different tree
and contained a duplicate hunk which will cause us to loose track of the
substream pointers when shutting down. Remove one of the duplicated
hunks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for Zoom2 board. Zoom2 machine driver
connects both codec DAIs (audio and voice) to omap3
McBSP ports in the following way:
HiFi <-> McBSP2
Voice <-> McBSP3
The zoom2 driver has the following DAPM widgets:
* Ext Mic: MAINMIC, SUBMIC (with bias)
* Ext Spk: HFL, HFR
* Headset Stereophone: HSOL, HSOR
* Headset Mic: HSMIC (with bias)
* Aux In: AUXL, AUXR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These are not supported since performance can not be guaranteed
when they are in use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The WM8961 is a low power, high quality stereo CODEC designed for
portable digital applications with headphone and stereo class D speaker
drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* We don't need to write the registers twice, remove the first write.
* DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move it
out.
(This patch is for Mark's for-2.6.32 branch, I have not checked if the
code is duplicated on current 2.6.30)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 4eaa9819dc changed semantics of
private_value member of kcontrol. This resulted in inability to control
amplifier and subsequently in very low output volume.
Tested-by: Johannes Schauer <josch@pyneo.org>
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign proper errors when platform resource claims fail.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They are now only accessed within dapm_power_widgets() so can be local
to that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle gain ramping for PGAs so we can coalesce their power updates too.
This is not ideal since we can't cope properly with gain ramping for
stereo paths but that was the case without coalescing and gain ramping
is relatively infrequently used so the effects are limited.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The more flexible value muxes and named mixers don't need to be sorted
differently from a power management point of view, they are different
only in terms of the control interface and not in terms of seqencing
behaviour.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce the number of register writes we need to set the power state for
a CODEC by coalescing updates to widgets with the same sequence order and
same register into a single write.
This can be a noticable performance improvement with slow or heavily
contended control buses, such as I2C controllers with a low clock
frequency, and is particularly noticable when resuming. It can also
reduce the noticability of and pops and clicks by ensuring that left
and right channels are powered simultaneously if they are in the same
register.
Currently widgets that have events are not coalesced, including PGAs
which may use the volume ramping control.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch does the following:
(1) Add support for the DM646x machine
(2) Modifications required to introduce the platform driver model to get
platform data for all the machines including dm355 and dm644x.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds driver support for the two instances of McASP on TI's DM646x.
The multichannel audio serial port (McASP) functions as a general-purpose audio
serial port optimized for the needs of multichannel audio application.
(http://www.ti.com/litv/pdf/spruer1b).
There are two instances of McASP on DM646x. The McASP0 module includes up to 4
serializers that can be individually enabled to either transmit or receive
in different modes. The McASP1 module is limited with only 1 pinned-out
serializer that can be enabled to only transmit in DIT mode (neither receiving
in any mode nor transmitting in either Burst or TDM mode is supported).
McASP0 consists of transmit and receive sections that may operate
synchronized, or completely independently with separate master clocks, bit
clocks, and frame syncs, and using different transmit modes with different
bit-stream formats.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce the platform driver model to get platform data for dm355 and dm644x.
Register platform driver and acquire the resources in the probe function Since
the platform specific code had been moved from machine driver to dm<soc>.c
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Lump the list walk into a single function, and pull in the power
application too so we can do some further refactoring. Pure code
motion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the past the DAPM power sequencing was done by iterating over the list
of widgets once for each widget type and powering widgets of that type.
Instead of doing that do the sorting at the time we insert the widgets
into the lists of widgets to apply power changes to. This reduces the
amount of computation required for seqencing still further, though the
costs are generally dwarfed by the costs of the register writes
implementing them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some code analyzer software mistakenly gives
divide by 0 error messages for these lines.
This patch will end its confusion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case the initalization of an soc_device failed, there is no codec
associated with it. soc_suspend() will still dereference the pointer
and cause an Ooops when entering the sleep mode.
This happens on our board with a multi-target kernel image when booted
on a machine without audio circuits.
This patch makes the code bail out very early in this special case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the (likely cut-n-paste) error by commit
16a30fbb0d, which causes the error below:
sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache':
sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes crash when shutting down.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In addition to the operating mode check, also check the
codec's interface format in case of four channel mode.
If the codec is not in TDM (DSP_A) mode, return with error.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the codec->reg_cache instead of the array directly
in twl4030_init_chip for setting the default values.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed.
This patch provides stub codec that can be used in these configurations.
On DM646x EVM the McASP1 is connected to the S/PDIF out.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unsigned variables should use `%u' rather than `%d'.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The function signature for spin_event_timeout() has changed in version V9.
Adjust the mpc5200 AC97 driver to use the new function.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The effect of symmetric_constraints should provide a standard way to
enforce the use of the same sample rate for both directions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
These drivers use spin_event_timeout() which is only present in the
PowerPC tree at present and which is undergoing some API revisions
so temporarily mark them as BROKEN until these issues are sorted
out.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I've implemented retries for when the AC97 hardware doesn't reset on
first try. About 10% of the time both the Efika and pcm030 AC97 codecs
don't reset on first try and need to be poked multiple times. Failure
is indicated by not having the link clock start ticking. Every once in
a while even five pokes won't get the link started and I have to power
cycle.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rewrite the mpc5200 audio DMA code to support both I2S and AC97.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DAPM switch for HeadsetL/R mute. Since all bits are are needed
for the HFL/R pop removal to work the switch is using the SW_SHADOW
no HW register for the HandsfreeL/R mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shadow, non HW register for dealing with the HandsfreeL/R
muting.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E
to a more appropriate DAPM_PGA_E widget.
Also fix the power-up sequence to match with the TRM.
The power-down sequence is not described in the TRM, so do it
in a way, which seams like the correct sequence.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename the functions in the mpc5200 DMA file from i2s based names to dma
ones to reflect the file they are in.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Basic split of mpc5200 DMA code out from i2s into a standalone file.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fairly minor issues:
- Don't register the DAIs, it's not required for AC97 devices.
- Make unexported functions static.
- Wrap some excessively long lines.
- Undo tab/space breakage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8974 is a low power, high quality mono CODEC designed for portable
applications such as digital still cameras or digital voice recorders.
This driver was originally written by Graeme Gregory and Liam Girdwood
and has since been maintained by myself with some updates contributed by
Brett Saunders and Javier Martin.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver supports the audio subsystem on the Openmoko Neo FreeRunner
smartphone, often known by its codename GTA02. The system has a WM8753
connected to a Samsung S3C2442 with an external GPIO controlled speaker
amplifier.
The driver was originally written by Graeme Gregory and has recieved
contributions from Openmoko, myself and members of the Openmoko
community. For much of this time the primary Openmoko kernel maintainer
was Andy Green.
Signed-off-by: Graeme Gregory <graeme@openmoko.com>
Signed-off-by: Andy Green <andy@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that we always set a new sysclk when using the FLL in master mode
and pick out the correct value for the sample rate in hw_params().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Give unique stream names for the two playback streams so
DAPM can figure out which codec_dai is in use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this patch the initial headset pop-removal related values are
configured for the twl4030 codec (ramp delay and sysclk).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
twl4030_setup_data structure can be passed from platform drivers to
the codec via the snd_soc_device->codec_data pointer.
Currently the setup data has support for the Headset pop-removal
related configuration, which differs from board to board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles
the headset ramp up and down sequences needed for the pop noise
removal.
With this patch the order of the internal components in the twl4030
codec is turned on and off in a correct order.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Restructuring the twl4030 codec's DAPM routing to be able to handle the power
sequences correctly.
The twl4030 codec internal implementation have this order:
DAC -> Analog PGA -> Mixer/Mux
While the ASoC framework expects the following order:
DAC -> Mixer -> Analog PGA
This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and
adds two levels of mixer to handle the digital and analog loopback
functionality.
Now the analog loopback does not powers on any of the DACs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the integrated ACLC of the TXx9 family.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver.
Voice DAI init function enables corresponding interface by
writting directly to VOICE_IF codec register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a control for selecting the codec operation mode. TWL4030 codec
has two modes:
- Option 1. Audio only (4 audio DACs)
- Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC)
Control is restricted when a stream is ongoing, since codec's
operation mode cannot be changed on-the-fly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move all the gpio functions out of <mach/hardware.h> as
this file is for defining the generic IO base addresses
for the kernel IO calls.
Make a new header <mach/gpio-fns.h> to take this and
include it via the chain from <linux/gpio.h> which is
what most of these files should be using (and will be
changed as soon as possible).
Note, this does make minor changes to some drivers but
should not mess up any pending merges.
CC: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink
and downlink paths, respectively.
This patch also corrects voice 8/16kHz mode selection bit
(SEL_16K) of CODEC_MODE register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eliminate direct accesses to the driver_data field.
cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd015
The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
struct device *dev;
expression E;
type T;
@@
- dev->driver_data = (T)E
+ dev_set_drvdata(dev, E)
@@
struct device *dev;
type T;
@@
- (T)dev->driver_data
+ dev_get_drvdata(dev)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use default to optimize the switch/case in magicial_playback_hw_params(),
which also fixes the compile warnings below:
sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this function
sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a build fix, resyncing the DaVinci EVM ASoC board code
with the version in the DaVinci tree. That resync includes
support for the DM355 EVM, although that board isn't yet in
mainline.
(NOTE: also includes a bugfix to the platform_add_resources
call, recently sent by Chaithrika U S <chaithrika@ti.com> but
not yet merged into the DaVinci tree.)
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This resyncs the DaVinci I2S code with the version in the DaVinci
tree. The behavioral change uses updated clock interfaces which
recently merged to mainline. Two other changes include adding a
comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in
order to support more boards than just the DM644x EVM.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a buildfix for the DaVinci PCM code, resyncing it with
the version in the DaVinci tree. The notable change is using
current EDMA interfaces, which recently merged to mainline.
(The older interfaces never made it into mainline.)
NOTE: open issue, the DMA should be to/from SRAM; see chip
errata for more info. The artifacts are extremely easy to
hear on DM355 hardware (not yet supported in mainline), but
don't seem as audible on DM6446 hardwaare (which does have
mainline support).
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9712 can be configured by resistor strapping GPIO4 to behave like
the WM9713 and default to leaving the AC97 link disabled after cold
reset until a warm reset occurs. In this configuration we need to issue
a warm reset after cold to bring the link up so do so. The warm reset
will be harmless on systems that don't need it.
[Changelog rewritten to document the reasoning. -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa2xx_i2s_resume is :
- unconditionnaly setting SACR0_ENB
- unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0
fix these.
In pxa2xx_i2s_{resume,suspend}, save/restore registers even
when !dai->active.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown.
Now we 'get' i2s_clk at probe and 'put' it at driver removal or when
probe fails.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- hw_params enables both RPL and REC functions each time : Enable the
appropriate function in pxa2xx_i2s_trigger.
- pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is
off : Turn it off only when both functions are off.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we are in a know good state at end of probe :
Reset FIFO logic and registers, and make sure REC and RPL functions
along with FIFO service are disabled (SACR0_RST enables REC and RPL).
Resetting loses current settings so remove reset from stream startup.
Now reset occurs only at probe.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Resending the patch after fixing the minor issues.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset FIFO logic and registers, and make sure REC and RPL functions along
with FIFO service are disabled at probe.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The inputs of the twl4030 codec can be mixed, so we will use the mixer
DAPM for the analog microphone registers(0x05, 0x06), but if we enable
more than one input at the same time, the input impedance of the input
amplifier will be reduced.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the card was not instantiated in snd_soc_instantiate_card, calling
soc-remove will crash because some of codec, cpu_dai and card .remove
methods are called twice.
Fix this by returning from soc_remove immediately.
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC uses the standard ALSA data format definitions to specify the wire
format used between the CPU and CODEC. Since the ALSA data formats all
include the endianess of the data but this information is not relevant
by the time the data has been encoded onto the serial link to the CODEC
this means that either all the CODEC drivers need to declare both big and
little endian variants or the core needs to fix up the format constraints
specified by CODEC drivers.
For now take the latter approach - this will need to be revisited if any
CODECs are endianness dependant.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC driver for AT91SAM9260-based AFEB9260 board
Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN
It has not caused problems, since
TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the magic 0x80 value with a suitable macro definition.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the VIBRA output on TWL4030 codec.
The VIBRA output can be driven with audio data or with
local vibrator driver.
Add the needed DAPM elements and routes for the VIBRA output and
controls for the VIBRA driver configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the ASoC side of the board support for the Crossbow
IMB400 daughter board.
Thanks to Crossbow for considerable assistance.
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add voice digital loopback (sidetone) to the twl4030
driver. It mixes voice uplink attenuated (by sidetone gain) with
voice downlink when the codec is working in option2 (voice/audio
mode).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds voice downlink analog bypass switch. It follows
the same approach as in other analog bypass switches.
DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer',
that will also allow voice DAC to be powered in digital voice
loopback (sidetone).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mis-typing exist in dapm controller definitions and dapm route definitions,
so happen mis-matched error when snd_soc_dapm_add_routes().
Cc: stable@kernel.org
Signed-off-by: Jinyoung Park <parkjy@mtekvision.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
The AC97 wire format is completely fixed so CODECs don't have any choice
about the formats they accept but controllers accept a variety of data
formats and render them down onto the bus. Have a shared define so all
the CODEC drivers will interoperate with any of our controller drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
calls
The S3C24XX DMA API channel configuration registers are being passed
values comprised of register values which makes it hard to move the
API to cover both the S3C24XX and S3C64XX.
These values can be calculated from knowing which device the channel
is connected to, so remove them from the two calls s3c2410_dma_config
and s3c2410_dma_devconfig.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
They aren't used by anything external and aren't prototyped; if any
users appear they can be exported again for them.
Also report what modes we have a problem with when we encounter invalid
mode configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On startup we try to make sure that the port is quiesced but if the
port is already stopped then this will generate a warning about the
RX/TX mode configuration. Configure the mode before doing the teardown
to suppress these warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The data format configuration for S3C64xx IISv2 is completely different
to that for S3C24xx. Instead of a single bit configuration in bit 0 of
IISMOD we have format selection in bits 13 and 14 and bit clock rate
selection in bits 1 and 2. While we're here add support for 24 bit
samples in S3C64xx.
At some point it may be desirable to expose the bit clock rate selection
to users but given the limited configuration options that may not be
required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This makes the interface usable with the s3c-iis-v2 rate calculator
and consistent with S3C2412.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The gain control for earpiece amplifier uses 0dB ~ 12dB according to the
TRM, but the present code is implemented to -6dB ~ 6dB.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers that
really don't give the precise pointer value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to check only if the WM8350 is master and only when starting
the stream so if either is not true then we can skip the check.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds a new control named 'Master Playback Switch' for cs4270
codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch
the put function and store the information about manually set mute
controls from userspace. When a manual mute is set, we don't want the
soc core to un-mute the outputs.
Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control modifies the MUTE register, hence the polarity must be
inverted.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-By: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Say what invalid values we're seeing when we see an invalid value and
ensure that errors are displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's expected behaviour for the CODEC header to provide them but the
WM8350 doesn't due to having all the registers together under drivers/mfd.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/asoc:
ASoC: s3c-i2s-v2 needs to declare a license for modular builds
ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991
ASoC: Fix WM8580 volume update handling for large register changes
ASoC: Fix offset of freqmode in WM8580 PLL configuration
The S3C64xx IIS code had a number of problems with device registration.
The hardware has two IIS ports of which the driver supported only one
at once via a single exported DAI, attempting to identify the DAI to
use based on the dev->id of the ASoC platform device. As well as
limiting the driver to only supporting one IIS port at once this also
meant that the ID of the soc-audio device (or in future the card device)
had to match the IIS ID.
Fix both problems by converting the driver to register the DAIs based on
probing of platform devices registered by the arch/arm code, using those
platform devices to interact with the clock API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the four channel TDM mode
on Beagle board.
Depending on the channel count, the interface needs to be
configured differently (I2S for stereo DSP_A for four channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add 4 channel support to omap-mcbsp.
This mode is going to be used by the twl4030 codec, when it
is configured in Option1 mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The original idea came from pHilipp, and this makes the code looks
more consistent.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSP DMA parameters can actually be easily generated at run-time since
they are almost similar except for the FIFO width and direction. Another
benefit is the re-use of information from 'struct ssp_device', like SSDR
physical FIFO address and DRCMR register index for both directions.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: pHilipp Zabel <philipp.zabel@gmail.com>
Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have
to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset
Left/Right, Carkit Left/Right) from mux to mixer.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Save a little extra power by enabling the DC servo offset correction
for the output channels only when the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the default startup sequence in the chip to set the DC servo
dither level for optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CLK_DSP provides a master clock for the DAC and ADC related functionality
on the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT
<--> codec).
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This also switches us to using a switch statement for the widget type
in dapm_power_widget().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will form a basis for further power check refactoring: the overall
goal of these changes is to allow us to check power separately to
applying it, allowing improvements in the power sequencing algorithms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.
The PCM voice interface has two modes
- PCM mode1 : This uses the normal FS polarity and the rising edge of
the clock signal.
- PCM mode2 : This uses the FS polarity inverted and the falling edge
of the clock signal.
If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Firstly, this patch makes the palm27x asoc driver a little more sane. Also,
since all affected devices use GPIO95 as AC97_nRESET, this patch sets that
properly. Affected are PalmT5, TX and LifeDrive.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
I notice that the fixes were merged, minus one:
sound/soc/codecs/wm9705.c: At top level:
sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type
so you might find this trivial patch useful.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.
But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...
The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false
This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My email address is going to expire soon so update it. Adding also
Peter Ujfalusi <peter.ujfalusi@nokia.com> as a second contact to OMAP core
drivers since I won't have anymore access to non-public OMAP documentation
in the future and Peter is working with these drivers as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those macros are just screwed as soon as CONFIG_PXA25x is enabled.
This patch
- changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device
- adds a corresponding ssp_get_scr function.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.
The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a6c.
Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with FWID(0).
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix accidental change of <mach/regs-gpio.h> to
<plat/regs-gpio.h> in s3c2412-i2s.c
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the build error in s3c-i2s-v2.c caused by
a change to the snd_soc_dai ops field.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The definition of s3c_i2sv2_iis_calc_rate was never
renamed from s3c2412_iis_calc_rate, so rename this
to allow the build to work.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors in sound/soc/s3c24xx/jive_wm8750.c
from changes to ASoC.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa_ssp_set_dai_fmt() currently has an early exit if the desired format
equals the current configuration. This is correct behaviour unless this
function is called with a zero value parameter for the first time.
Zero is a valid value for this function, but the early exit is bogus in
this case.
Hence, set priv->dai_fmt to -1 in the beginning so we can configure the
port.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Mark 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: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark 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: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless. Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark 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: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix for compillation error introduced by the constrain patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver should pass a device that specifies internal DMA ops, but
substream->pcm is just a logical device, and thus doesn't have arch-
specific dma callbacks, therefore following bug appears:
Freescale Synchronous Serial Interface (SSI) ASoC Driver
------------[ cut here ]------------
kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237!
Oops: Exception in kernel mode, sig: 5 [#1]
...
NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac
LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108
Call Trace:
[df02bde0] [df02be2c] 0xdf02be2c (unreliable)
[df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108
[df02be10] [c023a100] fsl_dma_new+0x68/0x124
[df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234
[df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148
[df02bed0] [c023824c] cs4270_probe+0x34/0x124
[df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4
[df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68
[df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80
[df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28
...
This patch fixes the issue by using card's device instead.
Signed-off-by: Anton Vorontsov <avorontsov@ru.mvista.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark 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: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark 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: Dan Carpenter <error27@gmail.com>
Signed-off-by: Mark 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: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a driver for the I2S interface found on Stretch s6000
family processors.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds the needed code to be able to use 96KHz playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this the WM9705 driver fails badly when resuming.
Tested-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark 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: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark 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: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark 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: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark 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: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark 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: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark 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: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark 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: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark 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: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark 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: Mark 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 14fa43f53f ("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: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark 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: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Two issues are fixed here:
- I2S transmits the left frame with the clock low but I don't seem to
get LRCLK out without SFRMDLY being set so invert SFRMP and set a
delay.
- I2S has a clock cycle prior to the first data byte in each channel
so we need to delay the data by one cycle.
Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.
Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application. This driver supports the
primary audio CODEC features, including:
- 1W speaker driver
- Fully differential headphone output
- Up to 4 differential microphone inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Buildfix:
CC sound/soc/omap/osk5912.o
sound/soc/omap/osk5912.c: In function 'osk_soc_init':
sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
make[3]: *** [sound/soc/omap/osk5912.o] Error 1
There's no such (standard) clock interface.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for true pause and unpause. Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.
Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.
Optimize the delay after starting capture. Instead of delaying 1ms, the driver
now polls the hardware. The new delay is shorter by over 90% yet still
effective.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upgrade the severity of some failure messages from debug level so
they're displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode. In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS. Asynchronous mode allows playback and capture to
use different sample sizes. It also technically allows different sample rates,
but the driver does not support that.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.
Also fix the coding style for the switch below while we're here.
Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Select the relevant DMA implementation when the
sound driver is selected.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.
As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the IIS headers to their correct place.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When setting WM8510_MCLKDIV the pll was turned off.
When setting pll frequency you got twice the expected freq, because
the code calculated with postscaler of 8, but the hardware divide by 4.
Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
