Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.
Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.
Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.
Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.
Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.
Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).
In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.
Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.
Tested on my Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.
This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for i.mx27_visstrim_sm10 board machine driver which
uses an i.mx27 processor plus a wm8974 codec.
It has been tested on a visstrim_sm10 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DAI platform for the SSI present in MXC platforms.
It currently does not support i.MX3, the only thing necessary to do
this is to export DMA data for i.MX3 interface which I haven't done
because I don't have a i.MX3 based board available.
It has been tested on i.MX27 board.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DMA platform valid for i.MX1 and i.MX2 platforms.
This is not valid for i.MX3 since it doesn't share the same DMA
interface than i.MX1 and i.MX2.
It has been tested on i.MX27 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.
That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.
Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that
s->runtime->control->appl_ptr can wrap around to the beginning of the
buffer. This change fixes this problem.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC.
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Depends on:
1) latest version of the CX20442 codec driver that exposes v253_ops
structure[1],
2) patch 2/3 form this series: TTY: Add definition of a new line
discipline required by Amstrad E3 (Delta) ASoC driver[2].
CPU DAI parameters best matching the codec DAI has been selected out
empirically for best user experience.
Board specific audio function control (with related DAPM widgets) has been
modeled after empirically discovered codec capabilities.
Unlike other ASoC machine drivers, this one makes use of a codec provided line
discipline that is required for talking to a modem chip that can control the
codec behavoiur. As the line discipline operations must call board specific
bits as well, the machine driver registers its own line discipline ops, not
the codec provided, and then calls those codec provided from inside its own
callbacks.
If some kind of a glue, like a bus over a tty, exsited that could help in
runtime detection of a modem (bus adapter) over a more generic line discipline
(bus driver)[3], the line discipline code could be probably designed in a
more generic way.
In order to work at all, this driver requires a working McBSP1. On OMAP1510
based machines (not sure if other OMAP1 variants as well), where McBSP1 is a
DSP public peripheral, that means the kernel must provide basic DSP support,
ie. omap_dsp_init(), in order to power up the DSP. This used to be included in
linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c.
Without that, the driver would not work, ie. not shift in/out any bits over
the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply
to other code that makes use of McBSP1/McBSP3 on affected machines. I provide
an extra patch (4/3) as a temporary solution.
To work correctly in playback mode, this driver requires my prevoiusly
submitted patch that corrects pcm pointer calculation for OMAP1510 based
machines[5] (already included in linux-2.6.31-rc3).
To support codec controls, this driver requires my previously submitted patch
that adds support for modem found on Amstrad Delta[6].
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html
[2] http://www.spinics.net/lists/linux-serial/msg01862.html
[3] http://www.spinics.net/lists/linux-serial/msg01856.html
[4] http://www.spinics.net/lists/linux-omap/msg15114.html
[5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html
[6] http://www.spinics.net/lists/linux-omap/msg15432.html
Credits to:
Mark Underwood - for his initial, omap-alsa based sound driver for
this machine,
Mark Brown - for his help, patience and excellent subsytem maintainer support.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.
Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html
Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The irq can fire as soon as it has been requested, thus all fields accessed
from within the irq handler must be initialized prior to requesting the irq.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps CODECs with sparse register maps work better with the
register cache display interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. fix "line over 80 characters" checkpatch warnings
2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead
3. fix typos
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the s3c platform has implementing gpiolib support the s3c_gpio api has been
deprecated.
This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The I2S DAI driver for blackfin SPORT, but works in TDM mode.
I2S is not a special case of TDM with only left and right two slots for
SPORT interface. I2S coordinates with TDM in SPORT, but not a part of
TDM. TDM require different hardware configuration with I2S, not only
different slot number. One is "Stereo Serial Operation" mode of SPORT,
the other one is "Multichannel Operation" mode. They are incompatible
at the same time.
Hardware and DMA description and data transfer flow are much different
for I2S and TDM. Merging them as a whole will be very ugly and difficult
to maintain.
So we don't define a new DAI type, but give two DAI instances for standard
I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the
I2S-family DAI type.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes the old method of jack detection from palm27x-asoc
driver and adds jack detection api. It also removes some other (now)
useless stuff from the driver and corrects pin configuration for the
codec.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a few small enhancements to the ASoC jack handling, as
suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes
for related bugs found while learning Mark's code and testing results.
Enhancements:
1. Update status of an ASoC jack while associating it with new gpios.
2. Really update DAPM pins while associating them with an ASoC jack.
3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes.
Fixes:
1. Apply mask on jack status report before using it, just for case.
2. While updating jack associated DAPM pins, use full resulting jack status,
not the status report passed as an argument.
Created and tested on linux-2.6.31-rc3
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().
If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.
This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.
The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.
Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.
Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.
[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrects a bug with ADC Inversion Switch in wm8974 codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset was failing with the original udelay(50) between the code in
psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testing
it was found that a delay of 1ms was necessary for the cold_reset code to
consistently complete successfully.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6
line, controlled by register INTBR_PMBR1. Machine driver takes care
of enabling gpio line through i2c and codec driver manipulates the
line during headset ramp up/down sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clock name strings are no longer passed on platform_data. Instead,
we rely entirely on struct device and clkdev to find the right clock.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.
The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.
The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf
[Slight edit to sort the ALL_CODECS entries -- broonie.]
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the flexibility of the WM9081 FLL this should never happen
in a real system.
Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to use the best value we picked, not the last value we
looked at.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
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>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
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>
Without MODULE_LICENCE("GPL"), when built as a module it will fail
to load because it uses other GPL symbols from kernel.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In SOC DAPM layer of SOUND subsystem, when add signal route (in the
function snd_soc_dapm_add_route() ), the original code has wrong logic
when dapm layer check each widget whether an external one.
Signed-off-by: Rongrong Cao <rrcao@ambarella.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check for rtd->params->drcmr != NULL before accessing it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide an interval after the end of DAPM sequencing so that we
can distinguish between a pop in the final step of the sequence
and a pop generated from some other source outside DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The system clock is currently fixed by the driver and this avoids
the need for us to handle errors with enabling and disabling MCLK
(which was incorrect previously so this fixes bugs in error
handling).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As shipped the board does not have inputs but it is relatively
straightforward to modify the board to hook them up so support
is provided in the driver. When these modifications have not
been made enabling the microphone stage can cause problems.
Add an ifdef to disable this by default. Don't put it into
Kconfig since users will have to get their soldering irons
out to change things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This simplifies the driver by removing the need to manually
configure dividers within the CPU and improve audio performance
by ensuring that the optimal phase relationships between the
clocks in the system are maintained.
Note that currently this means that for playback to work the
Output Mixer HiFi switch must be enabled since otherwise CODEC
will not generate the DAC clock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While the hardware is capable of some limited asynmmetric modes the
driver does not currently support those modes so tell applications
that only symmetric rates are available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the dma link with correct data as soon as
the master channel has copied it. Otherwise, the
1st period will play twice.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note the slightly tricky cache usage in the volume update function due
to the requirement for a separate write for the VU bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TLV320AIC3x driver is currently the only user of the CODEC hw_read
operation and is jumping through some hoops in order to do so. In order
to support future refactoring to make the hw_read operation more usable
unwrap the usage in this driver to avoid its use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Nothing uses it and the existing hw_read operation needs to be
refectored so it's easier to remove it rather than work with it.
Support can be re-added if the code requires volatile registers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the codec is master then prepare should call
mcbsp_start, not trigger.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Code previously just "ors" in this field without clearing
first. Fix, by never reading this register.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only start sample generator if needed, and more
cleanup on davinci_mcbsp_start.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move variable declaration closer to use.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add davinci_mcbsp_dev as argument to davinci_mcbsp_start
and davinci_mcbsp_stop.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add toggle_clock function to complete i2s reset earlier.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No functional changes. Rename variable w to something
more meaningful. Remove code obfuscating macro MOD_REG_BIT.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register cache space was not being allocated for the final register,
causing bugs when it was used. Allocate space for it.
Also ensure that the final register is displayed in sysfs.
[Commit message rewritten to document actual issue. -- broonie]
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add "set_tristate" callbacks for HiFi and Voice DAIs.
Machine drivers can enable and disable tristate for each
DAI with "snd_soc_dai_set_tristate" function.
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>
Add support for EXTMUTE in Zoom2 machine driver. This is necessary
to further reduce pop noise problem. Signal EXTMUTE is connected to
signal GPIO 153 in Zoom2 board.
In addition, change ramp delay value to 3 (218/161/109 ms). With
previous ramp delay value, pop noise was louder. With a longer value
the beep tone can be observed.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC97 bus register read/write hooks need to provide locking, but the
mpc5200-psc-ac97 driver does not. This patch adds a mutex around
the register access routines.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When doing register reads, it is possible for there to be a stale
data ready bit set which will cause subsequent reads to return
prematurely with incorrect data. This patch fixes the issues by
ensuring stale data is cleared before starting another transaction.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The wrong register cache variable was being used to provide the size for
the memcpy(), resulting in a copy of only a void * of data.
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
According to TRM, an external FET controlled by a 1.8V output signal
can be used to reduce the pop-noise heard when the audio amplifier is
switched on. It is suggested that GPIO6 of TWL4030 be used, but any
other gpio can be used instead. This is indicated in machine driver
with the following twl4030_setup_data members:
-hs_extmute. Set to 1 if board has support for EXTMUTE.
-set_hs_extmute. Set to a callback funcion to control an external gpio
line. Set to NULL if MUTE[GPIO6] pin is used.
Codec driver takes care of enabling and disabling this output during
the headset pop attenuation sequence.
Also add a delay to let VMID settle in ramp up sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8523 is a high performance stereo DAC with integral charge
pump providing 2Vrms line driver outputs using a single 3.3V power
supply rail.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The clock API can't cope with unbalanced enables and disables and
we only enable in hw_params() but try to disable in shutdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The device is a mono device but it can read two channel data and
many I2S controllers only understand 2 channels.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch tries to work around the problem of broken OMAP1510 PCM playback
pointer calculation by replacing DMA function call that incorrectly tries to
read the value form DMA hardware with a value computed locally from an
already maintained variable omap_runtime_data.period_index.
Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC
driver.
Based on linux-2.6-asoc.git v2.6.31-rc1.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The needed spin_event_timeout() macro is now merged in from the
powerpc tree, so these drivers are no longer broken. This reverts
commit 0c0e09e21a (ASoC: Mark MPC5200
AC97 as BROKEN until PowerPC merge issues are resolved)
Tested against 2.6.31-rc1.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS.
Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly
registered on the AC97 bus, which prevents thinks like the WM9712
touchscreen driver from getting probed.
Tested against 2.6.31-rc1.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls,
removing word "Switch" from HandsfreeL/HandsfreeR widget name
for avoiding to duplicate it.
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>
Use kasprintf to allocate temporary devname string instead of a
fixed size string.
This fixes "FIXME" introduced on removal of BUS_ID_SIZE.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Make jack-plug notification selectable
ALSA: ctxfi - Add PM support
sound: seq_midi_event: fix decoding of (N)RPN events
ALSA: hda - Add digital-mic support to ALC262 auto model
ALSA: hda - Fix check of input source type for realtek codecs
ALSA: hda - Add quirk for Sony VAIO Z21MN
ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
sound: fix check for return value in snd_pcm_hw_refine
ALSA: ctxfi - Allow unknown PCI SSIDs
ASoC: Blackfin: update the bf5xx_i2s_resume parameters
ASoC: Blackfin: keep better track of SPORT configuration state
Event for voice sidetone was being interpreted as an
analog HiFi bypass event because VSTPGA register offset
is less than ARXR2_APGA_CTL offset. Reordering the
register checks allows to handle voice digital bypass
event properly.
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>
AVDAC clk priority allows to determine the path ADC must
be connected when the codec is in option2 and both HiFi
and Voice paths are enabled.
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>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (49 commits)
[ARM] idle: clean up pm_idle calling, obey hlt_counter
[ARM] S3C: Fix gpio-config off-by-one bug
[ARM] S3C64XX: add to_irq() support for EINT() GPIO
[ARM] S3C64XX: clock.c: fix typo in usb-host clock ctrlbit
[ARM] S3C64XX: fix HCLK gate defines
[ARM] Update mach-types
[ARM] wire up rt_tgsigqueueinfo and perf_counter_open
OMAP2 clock/powerdomain: off by 1 error in loop timeout comparisons
OMAP3 SDRC: set FIXEDDELAY when disabling SDRC DLL
OMAP3: Add support for DPLL3 divisor values higher than 2
OMAP3 SRAM: convert SRAM code to use macros rather than magic numbers
OMAP3 SRAM: add more comments on the SRAM code
OMAP3 clock/SDRC: program SDRC_MR register during SDRC clock change
OMAP3 clock: add a short delay when lowering CORE clk rate
OMAP3 clock: initialize SDRC timings at kernel start
OMAP3 clock: remove wait for DPLL3 M2 clock to stabilize
[ARM] Add old Feroceon support to compressed/head.S
[ARM] 5559/1: Limit the stack unwinding caused by a kthread exit
[ARM] 5558/1: Add extra checks to ARM unwinder to avoid tracing corrupt stacks
[ARM] 5557/1: Discard some ARM.ex*.*exit.text sections when !HOTPLUG or !HOTPLUG_CPU
...
* topic/hda:
ALSA: hda - Make jack-plug notification selectable
ALSA: hda - Add digital-mic support to ALC262 auto model
ALSA: hda - Fix check of input source type for realtek codecs
ALSA: hda - Add quirk for Sony VAIO Z21MN
ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
Make the jack-plug notification via input layer selectable via Kconfig.
This is often unnecessary, and the similr function will be provided
using the ALSA control API in near future anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the suspend/resume support to ctxfi driver.
The team tested on the following seems ok:
AMD Athlon 64 3500+ / ASUS A8N-E / 512MB DDR ATI / Radeon X1300
20k1 & 20k2 cards
Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com>
Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.
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>
When decoding (N)RPN sequencer events into raw MIDI commands, the
extra_decode_xrpn() function had accidentally swapped the MSB and LSB
controller values of both the parameter number and the data value.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the digital-mic support with ALC262 auto model.
The new ALC262 models have the digital mic at NID 0x12. This widget
isn't checked in the current alc262_auto_create_analog_input_ctls()
since it's under 0x18. So, just reuse the routine for alc269 to fix
the behavior.
But, it doesn't suffice: the digital mic is supported only with the
ADC0, we have to exclude other ADCs when d-mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the input-source type by checking the widget type of
each capture-source item. Since some codecs can have both the mixer
and selector types depending on the ADC, alc_mux_enum_put() needs to
check each widget.
With this change, spec->capture_style gets unneeded, so it's removed,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit f9e336f65b
ALSA: hda - Unify capture mixer creation in realtek codes
removed the "Input Source" mixer element creation for toshiba-s06 model
because it contains a digital-mic input.
This patch take the control back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Fix the comparison of unsigned int that causes a compile warning below
by changing to the right signed type:
patch_sigmatel.c: In function ‘stac92xx_vref_set’:
patch_sigmatel.c:658: warning: comparison of unsigned expression < 0 is always false
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's a large 500ms delay in snd_via82xx_codec_wait() that, at least
on my hardware, appears to be unnecessary. The rest of the init of
the card works without logging any warnings or errors and both audio
and mixer settings work.
This adds an "nodelay" parameter to disable this (undocumented in the
code) large delay improving bootup time by 489-500ms.
[ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs
vs.
[ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs
Signed-off-by: Simon Arlott <simon@fire.lp0.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'params' is a pointer and looking at the code this probably should be a check
for ioctl return value.
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow unknown PCI SSIDs for emu20k1 and emu20k2 as "unknown" model.
Also, add a black-list check in case any device has to be listed
as "unsupported". It has a negative value in the pci quirk entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The selected 4930G model seemed to keep the subwoofer 'tuba'
function from operating correctly. Removing the existing PCI
ID match made this work again, but it was mapped to 'Side'
instead of to LFE as one would expect.
This attempts to enable all functionality and keep the amount
of available mixer sliders low. Any slider that had no audible
effect on the output audio has been removed, and as such EAPD
is not currently enabled.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These drivers aren't BF52x specific, so don't use bf52x in the names.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Latest ASoC only passes snd_soc_dai to the resume function.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not let the SPORT be reconfigured until there are no more active
streams. Then we can let the system reprogram the SPORT state.
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>
No need to define second copies of mode and format, they're declared
with exactly the same type at the head of the function and there's no
conflict in their use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
