Basically, soc_bind_dai_link() checks
cpu_dai->dev->of_node and dai_link->cpu_of_node in DT case.
But after that it will check
cpu_dai->name and dai_link->cpu_dai_name too.

On the other hand, snd_soc_dai :: name is created by
fmt_single_name() or fmt_multiple_name().

There is no confusion if dai name is created by fmt_multiple_name(),
since cpu_dai->name is same as dai_link->cpu_dai_name.
but, if dai name is created by fmt_single_name(), CPU DAI never match.

Thus, simple-card not set dai_link->cpu_dai_name if DT case
to skip naming match on soc_bind_dai_link()
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

For some CPU/CODEC DAI devices the slot information maybe needed. This
patch adds the slot information parsing for simple-card driver.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This
patch adds the slot parsing from DT supports.

TDM slot properties:
    dai-tdm-slot-num : Number of slots in use.
    dai-tdm-slot-width :  Width in bits for each slot.

For instance:
    dai-tdm-slot-num = <2>;
    dai-tdm-slot-width = <8>;

And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
to specify a explicit mapping of the channels and the slots. If it's absent
the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
tx and rx masks.

For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
for an active slot as default, and the default active bits are at the LSB of
the masks.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

This patch adds snd_soc_of_parse_audio_simple_widgets() and supports
below style of widgets name on DT:

	"template-wname", "user supplied wname"

For instance:
	simple-audio-widgets =
		"Microphone", "Microphone Jack",
		"Line", "Line In Jack",
		"Line", "Line Out Jack",
		"Headphone", "Headphone Jack",
		"Speaker", "Speaker External";

The "template-wname" currently includes: "Microphone", "Line", "Headphone"
and "Speaker".
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

If the DT is used and the CPU DAI device has only one DAI, the card
name will be like :

ALSA device list:
0: 40031000.sai-sgtl5000

And this name maybe a little ugly to some customers, so here the
card name parsing from DT node is supported.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

In the asoc_simple_card_parse_of() will parse the device node's CPU/CODEC
DAI commone fmts, and then in asoc_simple_card_sub_parse_of() will parse
the CPU/CODEC DAI's sub-node fmts, so we can combine the info->daifmt and
info->set.fmt in asoc_simple_card_sub_parse_of() not while just before
_set_fmt().

And this will be more easy to add new functions, such as supporting
_set_tdm_slot(), etc.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

If the CPU/CODEC DAI set_sysclk() is not support, the -ENOTSUPP will returnd.
Here do the check like set_fmt().
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Rename the pointer to the private data structure to 'priv' to avoid confusion with the platform data pointer.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

The platform data structure contains information which is used only by
the driver, and the driver allocates platform information fields which
are of no use.
Move the driver specific data to a new private structure and cleanup
the platform data structure.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

In the non-DT sequence, the platform data is copied as a whole to the
dynamic card info and the same pointer 'cinfo' is used to copy the
platform information to the card.

Use 'priv' as the pointer to the dynamic card info and copy only the
useful information from the platform data.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

The CPU and CODEC DAI names are still copied to the user info structure.
Put them directly in the DAI links.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

Have a cleaner code using a DAI link pointer.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

The OF pointers are put in the stack and then copied to the card
descriptor.
Put them directly at their right place.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

The DT values are copied to the non-DT structure before being moved to
the card structure.
Set directly the DT values in the card and move the non-DT copy to the
non-DT sequence.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

The check of the mandatory fields is done for DT in its specific sequence.
Move the global check to the non-DT sequence.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
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>