Eduardo Valentin [Thu, 20 Aug 2009 13:18:08 +0000 (16:18 +0300)]
OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers.
Adding McBSP register definition for IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Thu, 20 Aug 2009 13:18:07 +0000 (16:18 +0300)]
OMAP: McBSP: Provide functions for ASoC frame syncronization
ASoC has an annoying bug letting either L or R channel to be
played on L channel. In other words, L and R channels can
switch at random. This provides McBSP funtionality that may
be used to fix this feature.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shine Liu [Thu, 20 Aug 2009 15:02:23 +0000 (23:02 +0800)]
ASoC: S3C24XX : Align the peroid size to the buffer size
> Then it's a driver bug. If unaligned period size is allowed, it means
> that the irq is really generated in that period, not at the buffer
> boundary. Otherwise, it must have a proper hw-constraint to align the
> period size to the buffer size.
This patch will fix the bug metioned in the above mail. Force the peroid
size to be aligned with the buffer size.
Based and tested on linux-2.6.31-rc6.
Signed-off-by: Shine Liu <shinel@foxmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 19 Aug 2009 18:31:46 +0000 (19:31 +0100)]
ALSA: Restore support for DMAless DAIs on PXA
Used for applications such as direct bluetooth connections on
smartphones which don't go via the CPU. This used to be supported
before the refactoring to share code but this check was removed
during that move.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 19 Aug 2009 13:18:53 +0000 (14:18 +0100)]
ASoC: Provide default set_bias_level() implementation
If the CODEC does not provide a set_bias_level() then update the
bias_level variable for it since other parts of the system expect
that to be maintained.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shine Liu [Mon, 17 Aug 2009 10:52:01 +0000 (18:52 +0800)]
ASoC: UDA134X: Fix mistaken mute/unmute code
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.
Signed-off-by: Shine Liu <shinel@foxmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.
The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.
In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Chaithrika U S [Tue, 11 Aug 2009 20:59:21 +0000 (16:59 -0400)]
ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.
Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Chaithrika U S [Tue, 11 Aug 2009 20:58:52 +0000 (16:58 -0400)]
ASoC: DaVinci: McASP driver enhacements
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf
This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.
Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 29 Jul 2009 20:21:49 +0000 (21:21 +0100)]
ASoC: Factor out shared code from WM8993
The WM8993 analogue control is shared with other devices in the same
product line. Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Barry Song [Thu, 13 Aug 2009 03:59:32 +0000 (11:59 +0800)]
new ad1836 codec driver based on asoc
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.
Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 13 Aug 2009 12:59:34 +0000 (15:59 +0300)]
ASoC: TWL4030: Introduce PGAs for outputs
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Barry Song [Wed, 12 Aug 2009 03:34:25 +0000 (11:34 +0800)]
ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.
Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 11 Aug 2009 15:28:39 +0000 (16:28 +0100)]
ASoC: Update WM9081 for tdm_slot() API change
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.
Also allow 0 TDM slots to be configure (for use when disabling TDM).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):
CC [M] sound/soc/s3c24xx/s3c2443-ac97.o
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 7 Aug 2009 06:59:47 +0000 (09:59 +0300)]
ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
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>
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>
Mark Brown [Tue, 4 Aug 2009 22:50:16 +0000 (23:50 +0100)]
ASoC: Add WM8776 CODEC driver
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>
javier Martin [Wed, 5 Aug 2009 06:47:48 +0000 (08:47 +0200)]
ASoC: add DAI platform ssi driver for MXC
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>
Daniel Mack [Wed, 5 Aug 2009 18:50:43 +0000 (20:50 +0200)]
ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
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>
John Bonesio [Wed, 29 Jul 2009 15:38:55 +0000 (08:38 -0700)]
ASoC: MPC5200: Support for buffer wrap around
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>
Mark Brown [Fri, 10 Jul 2009 21:24:27 +0000 (22:24 +0100)]
ASoC: Add I/O control bus information to factored out cache setup
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>
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].
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>
ASoC: CX20442: push down machine independent line discipline bits
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>
TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver
This patch adds new line discipline name an number to include/linux/tty.h. The
line discipline will be used by the Amstrad E3 (Delta) sound driver that will
come next in this series of patches.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Alan Cox <alan@lxorguk.ukuu.org.uk> 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>
ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpiolib
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>
Barry Song [Mon, 27 Jul 2009 10:06:39 +0000 (18:06 +0800)]
ASoC: blackfin I2S(TDM mode) CPU DAI driver
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>
ASoC: CX20442: fix issues pointed out by subsystem maintainer
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>
Marek Vasut [Thu, 23 Jul 2009 20:16:56 +0000 (22:16 +0200)]
ASoC: Switch palm27x-asoc to jack detection api
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>
ASoC: Jack handling enhancements as suggested by subsystem maintainer
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>
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>
ASoC: Add support for Conexant CX20442-11 voice modem codec
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.
Chaithrika U S [Wed, 22 Jul 2009 11:45:04 +0000 (07:45 -0400)]
ASoC: tlv320aic3x: Enable PLL when not bypassed
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>
John Bonesio [Tue, 21 Jul 2009 20:15:40 +0000 (13:15 -0700)]
ASoC: MPC5200: Increase the delay time between resets
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>
ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
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>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for
double controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double
controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cliff Cai [Tue, 14 Jul 2009 14:01:40 +0000 (10:01 -0400)]
ASoC: Blackfin I2S: fix resume handling
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>
Cliff Cai [Tue, 14 Jul 2009 14:01:39 +0000 (10:01 -0400)]
ASoC: Blackfin AC97: fix resume handling
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>
projects / mirror_ubuntu-eoan-kernel.git / log