snd_soc_register_dais() looks up the CODEC that is registering the DAIs by
looping over all registered CODECs. This patch updates the code to
simply pass the CODEC that registers the DAIs to snd_soc_register_dais() thus
avoiding the lookup.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(),
but running in a loop. The only difference is that snd_soc_register_dai() calls
fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls
fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component()
which decides whether to call snd_soc_register_dai() or snd_soc_register_dais()
to snd_soc_register_dais() to decide which naming scheme to use. This allows us
to remove snd_soc_register_dai().

The patch also updates snd_soc_register_dais() to unregister every DAI it finds
for the component rather than trying to unregister one DAI for each DAI that was
registered. Both have the same result since there won't be more DAIs than what
have been registered, but the former is easier to implement.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

    In snd_soc_bytes_put function, it forces cpu to do cpu_to_be translation,
but for mmio bus which uses REGMAP_ENDIAN_NATIVE, it doesn't need to do
endian translation. So it is better to use regmap's api which can decide
if this translation is needed according to bus configuration.
Signed-off-by: NNenghua Cao <nhcao@marvell.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

In preparation for componentization move the ignore_pmdown_time field from the
snd_soc_codec struct to the snd_soc_component struct. Set it to true for non
CODEC components for now.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

There is no reason why active count tracking should only be done for CODECs but
not for other components. Moving the active count from the snd_soc_codec struct
to the snd_soc_component struct reduces the differences between CODECs and other
components and will eventually allow component to component DAI links (Which is
a prerequisite for converting CODECs to components).
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Keep track of which component registered a DAI. We'll need this as
componentization progresses.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

Instead of directly checking the 'active' field of the CODEC struct add a new
helper function that will return either true or false depending on whether the
CODEC is active. This will make the migration to the component level easier.

The patch also updates all CODEC drivers that check the active attribute to use
the new helper function.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

For CODEC to CODEC links we need to make sure to also manage the 'active' field
of the cpu_dai CODEC.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

We have the same code that increments and decrements the active field of the
various PCM runtime components (all with the same bugs). Factor this out into
common helper functions.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

For CODEC to CODEC links we should only immediately power down if both CODECs
are configured to ignore the power down delay. Factor the logic for this
into a helper function that can be used for both compressed and normal PCMs.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

For the snd_soc_cache_init(), the reg_size maybe zero and then the value
of codec->reg_cache, which is alloced via kzalloc, maybe equal to
ZERO_SIZE_PTR. If the reg parameter of snd_soc_cache_write() is large enough,
the cache[idx] = val maybe cause the kernel crash...

So this patch fix this via doing the zero pionter check of it.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

sound/soc/soc-core.c:2708:6-13: WARNING: Assignment of
bool to 0/1
sound/soc/soc-core.c:2726:3-10: WARNING: Assignment of
bool to 0/1

    More information about semantic patching is available at
http://coccinelle.lip6.fr/Signed-off-by: NNenghua Cao <nhcao@marvell.com>
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>