ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
Implement DMA channel self linking on OMAP1510 using AUTO_INIT and REPEAT
flags of the DMA CCR register.
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 21 Aug 2009 15:38:13 +0000 (16:38 +0100)]
ASoC: Add DAPM widget power decision debugfs files
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ben Dooks [Thu, 20 Aug 2009 21:50:42 +0000 (22:50 +0100)]
ASoC: tlv320aic3x: fixup board device changes
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.
Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ben Dooks [Thu, 20 Aug 2009 21:50:41 +0000 (22:50 +0100)]
ASoC: tlv320aic3x: Change to use device model
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ben Dooks [Thu, 20 Aug 2009 21:50:40 +0000 (22:50 +0100)]
ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec boards
Add core support for the range of S3C24XX Simtec boards with TLV320AIC23
CODECs on them. Since there are also boards with similar IIS routing the
AMP and the configuration code is placed in a core file for re-use with
other CODEC bindings.
Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:26 +0000 (16:18 +0300)]
ASoC: OMAP: Use DMA operating mode of McBSP
Configures DMA sync mode depending on McBSP operating mode value.
The value is configurable by McBSP instance. So, depending
on McBSP operating mode, the DMA sync mode is passed from
omap-mcbsp to omap-pcm. Besides that, it also configures
McBSP threshold value depending on which McBSP mode is activated.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:25 +0000 (16:18 +0300)]
ASoC: OMAP: Use McBSP threshold to playback and capture
This patch changes the way DMA is done in omap-pcm.c
in order to reduce power consumption. There is no need
to have so much SW control in order to have DMA in idle
state during audio streaming. Configuring McBSP threshold value
and DMA to FRAME_SYNC are sufficient.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Thu, 20 Aug 2009 13:18:24 +0000 (16:18 +0300)]
ASoC: Always syncronize audio transfers on frames
All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Thu, 20 Aug 2009 13:18:23 +0000 (16:18 +0300)]
ASoC: Add runtime check for RFIG and XFIG
This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:22 +0000 (16:18 +0300)]
ASoC: OMAP: Make DMA 64 aligned
Align DMA address to DMA burst transaction sizes.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:21 +0000 (16:18 +0300)]
ASoC: OMAP: Enable DMA burst mode
Improve DMA transfers by enabling Burst transaction.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Thu, 20 Aug 2009 13:18:20 +0000 (16:18 +0300)]
OMAP: McBSP: Let element DMA mode hit retention also
The device no longer hits retention if element DMA
mode is taken for at least the duration of the
serial console timeout. Force element DMA mode to
shut down through smartidle.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-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 13:18:19 +0000 (16:18 +0300)]
OMAP: McBSP: Do not enable wakeups for no-idle mode
When no-idle mode is taken, wakeups need not to be enabled.
Moreover, CLOCKACTIVITY bits are unnecessary with this mode
also.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:18 +0000 (16:18 +0300)]
OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold
Use dma mode property to configure NO IDLE or SMART IDLE of McBSPs.
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 13:18:17 +0000 (16:18 +0300)]
OMAP: McBSP: Retain McBSP FCLK clockactivity
FCLK may get autogated so that it prevents the McBSP
to work properly. It is the bit 9 that must be set
for maintaining the McBSP FCLK.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:16 +0000 (16:18 +0300)]
OMAP: McBSP: Change wakeup signals
Configure only XRDYEN and RRDYEN wakeup signals
in order to get better power consumption.
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 13:18:15 +0000 (16:18 +0300)]
OMAP: McBSP: Wakeups utilized
This patch enables the smart idle mode while
McBPS is being utilized. Once it's done,
force idle mode is taken instead. Apart of it,
it also configures what signals will wake mcbsp up.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> 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>
Peter Ujfalusi [Thu, 20 Aug 2009 13:18:14 +0000 (16:18 +0300)]
OMAP: McBSP: Add link DMA mode selection
It adds a new sysfs file, where the user can configure the mcbsp mode to use.
If the mcbsp channel is in use, it does not allow the change.
Than in omap_pcm_open we can call the omap_mcbsp_get_opmode to get the mode,
store it, than use it to implement the different modes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> 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>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:13 +0000 (16:18 +0300)]
OMAP: McBSP: Rename thres sysfs symbols
This patch renames the symbols that handles threshold
sysfs properties. This way we can add more sysfs properties
to them.
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>
Peter Ujfalusi [Thu, 20 Aug 2009 13:18:12 +0000 (16:18 +0300)]
OMAP3: McBSP: Lower the maximum buffersize for McBSP1, 3, 4, 5
Do not allow applications to use the full buffer found on
McBSP1,3,4,5. Using the full buffer in threshold mode causes
the McBSP buffer to run dry, which can be observed as channels
are switching (in reality the channels are shifting).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:11 +0000 (16:18 +0300)]
OMAP: McBSP: Create and export max_(r|t)x_thres property
This patch export through sysfs two properties to configure
maximum threshold for transmission and reception on each
mcbsp instance. Also, it exports two helper functions to
allow mcbsp users to read this values.
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>
This patch adds a way to handle transmit/receive threshold.
It export to mcbsp users a callback registration procedure.
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>
Eduardo Valentin [Thu, 20 Aug 2009 13:18:09 +0000 (16:18 +0300)]
OMAP: McBSP: Use appropriate value for startup delay
Increasing startup delay value as worst case:
CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec
Although, 100us may give enough time for two CLKSRG,
due to some unknown PM related, clock gating etc. reason,
this patch increases it to 500us.
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>
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>
Mark Brown [Mon, 17 Aug 2009 16:39:22 +0000 (17:39 +0100)]
ASoC: Add input and output AIF widgets
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.
To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.
A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.
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>
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