This commit adds a bare bones driver support for TLV320AIC31XX family
audio codecs. The driver adds basic stereo playback trough headphone
and speaker outputs and mono capture trough microphone inputs.

The driver is currently missing support at least for mini DSP features
and jack detection. I have tested the driver only on TLV320AIC3111,
but based on the data sheets TLV320AIC3100, TLV320AIC3110, and
TLV320AIC3120 should work Ok too.

The base for the implementation was taken from:
git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx
-branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13.
Signed-off-by: NJyri Sarha <jsarha@ti.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The tlv320aic32x4 related lines were wrongly placed after tlv320aic3x lines.
Signed-off-by: NJyri Sarha <jsarha@ti.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Rearrange clock tree shutdown to disable them in the reversed order of
startup. First disable all dividers, then PLL followed by master clock.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Support regulators to power up the codec. This patch also enables the
AVDD LDO if no AV regulator was found.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add support for a master clock passed through DT. The master clock of
the codec is only active when the codec is in use.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add DT support for this codec. The bindings differ a bit from the aic3x
codec bindings, so I created a new binding documentation.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

There are a number of mixer controls that support negative values. They
use signed values for this with different number of bits for the values.
Currently they only support the positive range.

This patch replaces the unsigned mixers with signed mixers to support
the full range.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Use function gpio_is_valid to check for gpio ports.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Some codecs use signed volume control representation with non standard
register sizes, e.g. 6 or 7 bit signed integers.

This patch adds generic signed register volume control logic to
soc-core. Instead of a fixed width signed register control, this
implementation uses a 'min' value and the signed bit location to translate
it to an absolute volume. Using the 'sign_bit' we can calculate a
correct mask for the register values and translate it back into signed
integers of standard size.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Commit 384a48d7 "ALSA: hda: HDMI: Support codecs with fewer cvts
than pins" dynamically enabled each pin widget's PIN_OUT only when the
pin was actively in use. This was required on certain NVIDIA CODECs for
correct operation. Specifically, if multiple pin widgets each had their
mux input select the same audio converter widget and each pin widget had
PIN_OUT enabled, then only one of the pin widgets would actually receive
the audio, and often not the one the user wanted!

However, this apparently broke some Intel systems, and commit
6169b673 "ALSA: hda - Always turn on pins for HDMI/DP" reverted the
dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA
CODECs.

This change supports either dynamic or static handling of PIN_OUT,
selected by a flag set up during CODEC initialization. This flag is
enabled for all recent NVIDIA GPUs.
Reported-by: NUosis <uosisl@gmail.com>
Cc: <stable@vger.kernel.org> # v3.13
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When we plug a 3-ring headset on the Dell machine (Vendor ID:
0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be
detected, after apply this patch, the headset mic can work well.

BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: NDoro Wu <fan-cheng.wu@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Remove old SPI control functions, change anti-pop init
sequence, remove some garbage from structures. The 'Apply' functions
must be called at the mixer initialization, otherwise
mixer settings sometimes will not be applied at startup.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

First of all, we should not touch the GPIOs. They are not
for selecting the capture source, but they seems just enable
the whole audio input curcuit. The 'put' function calls the
'apply' functions to change register values. Change the order
of capture sources.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Modify the input_vol_* functions to use the new SPI routines,
There is a new applying function that will be called when
the capture source changed.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

I tried both variants: volume control and impedance selector.
In the first case one minus is that we can't change the
volume of multichannel output without additional software
volume control. However, I am using this variant for the
last three months and this seems good. All multichannel
speaker systems have internal amplifier with the
volume control included, but not all headphones have
this regulator. In the second case, my software volume
control does not save the value after reboot.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change the order of elements in the output select control. This will
reduce the number of relay switches. Change 'put' function to call the
oxygen_update_dac_routing() function. Otherwise multichannel playback
does not work. Also there is a new function to apply settings, this
prevents from duplicating the code.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Actually CS4245 connected to the I2S channel 1 for
capture, not channel 2. Otherwise capturing and
playback does not work for CS4245.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Moving the mixer code away makes things easier. The mixer
will control the driver, so the functions of the
driver need to be non-static.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change the function to read the data from the new shadow buffer.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

When selecting the audio output destinations (headphones,
FP headphones, multichannel output), the channel routing
should be changed depending on what destination selected.
Also unnecessary I2S channels are digitally muted. This
function called when the user selects the destination
in the ALSA mixer.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Change CS4245 initialization: different sequence and GPIO values,
according to datasheets and reverse-engineering information.
Change cleanup/resume/suspend functions, since they use
initialization.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Add the new SPI write and read functions. The SPI read function
is used for creating initial registers dump and may be used for
debugging purposes. SPI operations are cached, so there is a new
function to manage the cache (shadow). I have to remove
the shift from the CS4245_SPI_* constants, since when
we are performing the reading, we need to shift by 8 instead
of 16.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Add additional constants to the xonar_dg.h file:
capture and playback sources. Move GPIO_* constants and the
dg struct to the header file from the xonar_dg.c file.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Add some additional information in comments and my copyright.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

When the user switches the output from stereo to multichannel
or vice versa, the driver needs to update the channel routing.
Instead of creating additional subroutines, I better export existing
oxygen_update_dac_routing symbol from the oxygen mixer
and call this function. It calls model.adjust_dac_routing()
and my function does the work.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Modify the oxygen_write_spi() function to use the newly
introduced oxygen_wait_spi() function. Change return value
from void to int, so it can return error codes. Older
drivers just ignore that return value, new drivers can
check this value. We need to wait AFTER
initiating the SPI transaction, otherwise read
operation will not work.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

The oxygen_wait_spi() function now performs waiting when the
SPI bus completes a transaction. Introduce the timeout error
checking and increase timeout to 200 from 40.
Signed-off-by: NRoman Volkov <v1ron@mail.ru>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Processing coefficients are often a vital part of the codec's configuration,
so dumping them can be important. However, because they are undocumented and
secret, we do not want to enable this for all codecs by default.

Therefore instead add this as a debugging parameter.

I have prepared for codecs that want to enable this by default by the extra
dump_coef bitfield, but unsure if we want to do that as long as the
(unlikely, but still) race remains.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Currently the Negative Terminal Input Routing Configuration is only set
when there is a special routing configuration. If we don't use one of
the inputs IN1 or IN2 as negative terminal input, the PGA and recording
does not work.

This patch adds a route from CM1L/CM1R to the PGA as negative input by
default. With this configuration the PGA can amplify all input signals
and line-in/mic works again.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Playback of a mono stream should output the same stream on both
channels. At the moment only the left analog signal is valid, the right
one is just noise.

This patch maps the left digital channel onto both DACs when receiving a
mono stream.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Similarly to other Apple products, MBA 1,1 needs a specific quirk.
Pin 0x18 must be set to VREF_50 to have sound output.  This was no
longer done since commit 1a97b7f2, resulting in a mute built-in speaker.

This patch corrects the regression by creating a fixup for the MBA 1,1.

Fixes: 1a97b7f2 ("ALSA: hda/realtek - Remove the last static quirks for ALC882")
Cc: <stable@vger.kernel.org> [v3.4+]
Tested-by: NAdrien Vergé <adrienverge@gmail.com>
Signed-off-by: NAdrien Vergé <adrienverge@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

mach/dma.h is not referenced by this file. Remove it.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NMark Brown <broonie@linaro.org>

'res' could be NULL from one of the operations above (line 1243). Thus
check 'res' for NULL before releasing the region to avoid null pointer
dereference.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NMark Brown <broonie@linaro.org>

Depend on MFD_ARIZONA to avoid the following build errors:

sound/soc/codecs/arizona.c:218: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:226: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:1719: undefined reference to `arizona_request_irq'
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NMark Brown <broonie@linaro.org>

Select S3C24XX_DMA instead of S3C2410_DMA to avoid following dependency issues
and build errors:

warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))
warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))

arch/arm/mach-s3c24xx/built-in.o: In function `s3c2410_dma_add':
arch/arm/mach-s3c24xx/dma-s3c2410.c:134: undefined reference to `s3c2410_dma_init'
arch/arm/mach-s3c24xx/dma-s3c2410.c:135: undefined reference to `s3c24xx_dma_order_set'
arch/arm/mach-s3c24xx/dma-s3c2410.c:136: undefined reference to `s3c24xx_dma_init_map'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/soc/samsung/s3c24xx-i2s.c:293: undefined reference to `s3c2410_dma_ctrl'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/built-in.o: In function `s3c2412_i2s_trigger':
sound/soc/samsung/s3c-i2s-v2.c:432: undefined reference to `s3c2410_dma_ctrl'
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NMark Brown <broonie@linaro.org>

Export the symbol so that it is accessible to modules. Fixes the
following error:

ERROR: "wm5100_detect" [sound/soc/samsung/snd-soc-lowland.ko] undefined!
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Acked-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

There are two loops that are almost identical but only with different
checks.  Refactor them with a simple helper, and give a bit more
comments what's doing there.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>