More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: NJassi <jassi.brar@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: NChaithrika U S <chaithrika@ti.com>

Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Free socdev if snd_soc_init_card() fails.
Signed-off-by: NRoel Kluin <roel.kluin@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se
Signed-off-by: NKuninori Morimoto <morimoto.kuninori@renesas.com>
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>

The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

These need to be in the CODEC since the DAIs supported by the CODECs
aren't static.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NShine Liu <shinel@foxmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NBarry Song <21cnbao@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NMark Brown <broonie@opensource.wolfsonmicro.com>

- Build in SND_SOC_ALL_CODECS.
- Remove null suspend/resume stuff.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NBarry Song <21cnbao@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NBarry Song <21cnbao@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

It's only actually paying attention to the slot count anyway.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark 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: NDaniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark 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: NMark 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: NDaniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: NMark 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: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark 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: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark 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: NBarry Song <21cnbao@gmail.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark 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: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NMark 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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NMark 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: NChaithrika U S <chaithrika@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>