The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.

Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.

Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix the write to PMBR1 register through I2C. Also, the constant which
holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This
name is based on TRM to avoid confusion.
Signed-off-by: NJorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable
can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of
those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing
to start and stop individually the transmitter and receiver.

This cleans up the code in arch/arm/plat-omap/mcbsp.c and in
sound/soc/omap/omap-mcbsp.c which was the only user for those removed
functions.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Commit ca6e2ce0 is setting up few XCCR and
RCCR bits for I2S and DPS_A formats. Part of the bits are already set
for all formats and I believe that XDISABLE and RDISABLE bits are
format independent.

As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup
of XDISABLE and RDISABLE to where those cpu's are tested and remove format
dependent part for simplicity.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NBen Dooks <ben@simtec.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NBen Dooks <ben@simtec.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NEduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NEduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NEero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.
Signed-off-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Align DMA address to DMA burst transaction sizes.
Signed-off-by: NEduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Improve DMA transfers by enabling Burst transaction.
Signed-off-by: NEduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMark 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: NJorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: NMark 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: NJorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Acked-By: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add support for Zoom2 board. Zoom2 machine driver
connects both codec DAIs (audio and voice) to omap3
McBSP ports in the following way:

  HiFi <-> McBSP2
 Voice <-> McBSP3

The zoom2 driver has the following DAPM widgets:
* Ext Mic: MAINMIC, SUBMIC (with bias)
* Ext Spk: HFL, HFR
* Headset Stereophone: HSOL, HSOR
* Headset Mic: HSMIC (with bias)
* Aux In: AUXL, AUXR
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

With this patch the initial headset pop-removal related values are
configured for the twl4030 codec (ramp delay and sysclk).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver.
Voice DAI init function enables corresponding interface by
writting directly to VOICE_IF codec register.
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Acked-by: NPeter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Resending the patch after fixing the minor issues.
Signed-off-by: NAnuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds support for the four channel TDM mode
on Beagle board.

Depending on the channel count, the interface needs to be
configured differently (I2S for stereo DSP_A for four channels)
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add 4 channel support to omap-mcbsp.
This mode is going to be used by the twl4030 codec, when it
is configured in Option1 mode.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT
<--> codec).
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add Voice DAI to support the PCM voice interface of the twl4030 codec.

The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.

The PCM voice interface has two modes
 - PCM mode1 : This uses the normal FS polarity and the rising edge of
               the clock signal.
 - PCM mode2 : This uses the FS polarity inverted and the falling edge
               of the clock signal.

If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

My email address is going to expire soon so update it. Adding also
Peter Ujfalusi <peter.ujfalusi@nokia.com> as a second contact to OMAP core
drivers since I won't have anymore access to non-public OMAP documentation
in the future and Peter is working with these drivers as well.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.

The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: NArun KS <arunks@mistralsolutions.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a.

Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with  FWID(0).
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NAlexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During
initial playback startup, this FIFO is keeping the DMA request active
until the FIFO is full.

So now if ALSA buffer size is smaller, DMA is looping around it while
filling up the HW FIFO, generating burst of interrupts as well and SW
doesn't have any change to fill enough data.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.

Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:

- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.

Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>