Sparse gives us the following warnings on sst-firmware.c

  CHECK   sound/soc/intel/sst-firmware.c
sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces)
sound/soc/intel/sst-firmware.c:39:34:    expected void volatile [noderef] <asn:2>*dst
sound/soc/intel/sst-firmware.c:39:34:    got void *
sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces)
sound/soc/intel/sst-firmware.c:417:36:    expected void *dest
sound/soc/intel/sst-firmware.c:417:36:    got void [noderef] <asn:2>*
sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static?

and

  CC [M]  sound/soc/intel/sst-dsp.o
sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces)
sound/soc/intel/sst-dsp-priv.h:271:53:    expected void *src
sound/soc/intel/sst-dsp-priv.h:271:53:    got void [noderef] <asn:2>*
sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces)
sound/soc/intel/sst-dsp-priv.h:271:53:    expected void *src
sound/soc/intel/sst-dsp-priv.h:271:53:    got void [noderef] <asn:2>*
sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces)
sound/soc/intel/sst-dsp-priv.h:271:53:    expected void *src
sound/soc/intel/sst-dsp-priv.h:271:53:    got void [noderef] <asn:2>*

This patch removes these warnings
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

We originally thought to request SST audio DSP firmware during the SST
platform driver initialization. However plain request_firmware doesn't
work in driver probe paths if userspace is not ready to handle it. For
instance when drivers are built-in.

Implementing asynchronous firmware request in SST platform driver
initialization complicates code needlessly since it anyway will fail if
firmware is missing.

This is more simple to handle by requesting firmware asynchronously in
sst_acpi_probe() and register SST platform only after firmware is loaded.
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Move fw_base and fw_size fields in struct sst_pdata under ACPI data for
clarifying that these are not related to firmware file but for platform
specific extended firmware area reserved by the BIOS.
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The Intel audio DSP SST trace event header has been renamed from sst.h
to intel-sst.h in order to avoid any confusion with any future
Samoa Standard Time drivers ;)
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add GFP_KERNEL when allocating firmware DMA buffer.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

This adds kernel build support for Intel SST core audio.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Provide services for Intel SST drivers to load SST modular firmware.

SST Firmware can be made up of several modules. These modules can exist
within any of the compatible SST memory blocks. Provide a generic memory
block and firmware module manager that can be used with any SST firmware
and core.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Most of the SST devices will be exposed as ACPI devices. It makes sense to
avoid duplication of the driver enumeration logic and concentrate the functionality
into a single ACPI SST enumeration file.

Idea of this loader is to parse data we get from ACPI and to be able to load
needed other SST drivers and ASoC machine driver runtime based on single
ACPI ID what BIOS gives to us.
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add support for Intel Smart Sound Technology (SST) audio DSPs.
This driver provides the low level IO, reset, boot and IRQ management
for Intel audio DSPs. These files make up  the low level part of the SST
audio driver stack and will be used by many Intel SST cores like
Haswell, Broadwell and Baytrail.

SST DSPs expose a memory mapped region (shim) for config and control.
The shim layout is mostly shared without much modification across cores
and this driver provides a uniform API to access the shim and to enable
basic shim functions. It also provides functionality to abstract some shim
functions for cores with different shim features.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Resent with correct email for Mark.

In order to differentiate the different Intel SST audio core drivers we
need to rename the current drivers with a mfld prefix. This also includes
renaming in the Makefile and Kconfig
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.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>

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>