| Commit message (Collapse) | Author | Age |
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A governor may need to store its current state between calls to
throttle(). That state depends on the thermal zone, so store it as
private data in struct thermal_zone_device.
The governors may have two new ops: bind_to_tz() and unbind_from_tz().
When provided, these functions let governors do some initialization
and teardown when they are bound/unbound to a tz and possibly store that
information in the governor_data field of the struct
thermal_zone_device.
Cc: Zhang Rui <rui.zhang@intel.com>
Cc: Eduardo Valentin <edubezval@gmail.com>
Signed-off-by: Javi Merino <javi.merino@arm.com>
Signed-off-by: Eduardo Valentin <edubezval@gmail.com>
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The fair share governor has the concept of weights, which is the
influence of each cooling device in a thermal zone. The current
implementation forces the weights of all cooling devices in a thermal
zone to add up to a 100. This complicates setups, as you need to know
in advance how many cooling devices you are going to have. If you bind a
new cooling device, you have to modify all the other cooling devices
weights, which is error prone. Furthermore, you can't specify a
"default" weight for platforms since that default value depends on the
number of cooling devices in the platform.
This patch generalizes the concept of weight by allowing any number to
be a "weight". Weights are now relative to each other. Platforms that
don't specify weights get the same default value for all their cooling
devices, so all their cdevs are considered to be equally influential.
It's important to note that previous users of the weights don't need to
alter the code: percentages continue to work as they used to. This
patch just removes the constraint of all the weights in a thermal zone
having to add up to a 100. If they do, you get the same behavior as
before. If they don't, fair share now works for that platform.
Cc: Zhang Rui <rui.zhang@intel.com>
Cc: Eduardo Valentin <edubezval@gmail.com>
Cc: Durgadoss R <durgadoss.r@intel.com>
Acked-by: Durgadoss R <durgadoss.r@intel.com>
Signed-off-by: Javi Merino <javi.merino@arm.com>
Signed-off-by: Eduardo Valentin <edubezval@gmail.com>
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Currently you can specify the weight of the cooling device in the device
tree but that information is not populated to the
thermal_bind_params where the fair share governor expects it to
be. The of thermal zone device doesn't have a thermal_bind_params
structure and arguably it's better to pass the weight inside the
thermal_instance as it is specific to the bind of a cooling device to a
thermal zone parameter.
Core thermal code is fixed to populate the weight in the instance from
the thermal_bind_params, so platform code that was passing the weight
inside the thermal_bind_params continue to work seamlessly.
While we are at it, create a default value for the weight parameter for
those thermal zones that currently don't define it and remove the
hardcoded default in of-thermal.
Cc: Zhang Rui <rui.zhang@intel.com>
Cc: "Rafael J. Wysocki" <rjw@rjwysocki.net>
Cc: Len Brown <lenb@kernel.org>
Cc: Peter Feuerer <peter@piie.net>
Cc: Darren Hart <dvhart@infradead.org>
Cc: Eduardo Valentin <edubezval@gmail.com>
Cc: Kukjin Kim <kgene@kernel.org>
Cc: Durgadoss R <durgadoss.r@intel.com>
Signed-off-by: Kapileshwar Singh <kapileshwar.singh@arm.com>
Signed-off-by: Eduardo Valentin <edubezval@gmail.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"It was a busy development cycle at this time, as you can see a wide
range of changes in diffstat. There are no big changes but many
refactoring and improvements. Here we go some highlights:
ALSA core:
- Procfs codes were cleaned up to use seq_file
- Procfs can be opt out via Kconfig (only for EXPERT)
- Two types of jack API were unified finally; now both kctl and input
jack devs are handled via a single function call.
HD-audio:
- Continued code restructuring for the future ASoC driver; now HDA
controller driver is split to a core helper module.
- Preliminary codes for Skylake audio support in HDA core.
- Proper i915 gfx power well management for SKL & co
- Enabled runtime PM as default for Intel HDMI/DP codecs
- Newer Tegra chip supports
- More quirks for Dell headsets, Alienware (with CA0132), etc.
- A couple of DRM ELD helper API functions
ASoC:
- Support for loading ASoC topology maps from firmware, intended to
be used to allow self-describing DSP firmware images to be built
which can map controls added by the DSP to userspace without the
kernel needing to know about individual DSP firmwares
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring
- Big refactoring, cleanup and enhancement for the Wolfson ADSP
driver
- Cleanup series for TI TAS2552 and R-CAR drivers
- Fixes and improvements on RT56xx codecs
- Support for TI TAS571x power amplifiers
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
- Support for x86 systems with RT5650 and Qualcomm Storm
- Support for Mediatek AFE (Audio Front End) unit
- Other various small fixes to ASoC codec drivers
Firewire:
- Enhanced to allow non-blocking streams to use timestamp
synchronization
- Improve support for DM1500 and BeBoBv3
Misc:
- Cleanup of old pci API functions over all PCI sound drivers
- Fix long-standing regression of the old powermac i2c setup"
* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
ALSA: pcm: Fix pcm_class sysfs output
ALSA: hda-beep: Update authors dead email address
ASoC: wm_adsp: Move DSP Rate controls into the codec
ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
ALSA: hda: provide default bus io ops extended hdac
ALSA: hda: add hda link cleanup routine
ALSA: hda: add hdac_ext stream creation and cleanup routines
ASoC: rsrc-card: remove unused ret
ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
ASoC: mediatek: Add machine driver for MAX98090 codec
ASoC: mediatek: Add AFE platform driver
ASoC: rsnd: remove io from rsnd_mod
ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
...
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Further updates for v4.2
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
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Move gpio to gpio_desc and use gpiod APIs in codec driver.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
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In HDA extended bus the HDA link objects are created when multilink
capabilities are parsed. We need a routine which free up these link objects
for a bus. So add snd_hdac_link_free_all routine
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HDAC extended core should create streams for an extended bus and also free
up those on cleanup. So introduce snd_hdac_ext_stream_init_all and
snd_hdac_stream_free_all routines
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Yet another non-trivial conflicts resolution for the recent HD-audio fix.
Conflicts:
sound/pci/hda/hda_intel.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit a1b3fda6ae ALSA: hdac_ext: add hdac extended controller,
erroneously added snd_hdac_ext_bus_map_codec_to_link() function
declaration, so remove it
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now we have the bus and controller code added to find and initialize
the extended capabilities. Now we need to use them in stream code to
decouple stream, manage links etc
So this patch adds the stream handling code for extended capabilities
introduced in preceding patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The controller needs to support the new capabilities and allow
reading, parsing and initializing of these capabilities, so this patch
does it
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The new HDA controllers from Intel support new capabilities like
multilink, pipe processing, SPIB, GTS etc In order to use them we
create an extended HDA bus which embed the hdac bus and contains the
fields for extended configurations
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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'asoc/topic/wm5100', 'asoc/topic/wm8741' and 'asoc/topic/wm8960' into asoc-next
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Make sure userspace can define TLV controls for topology using the correct
type numbers and channel mappings.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ASoC topology UAPI header defines the structures
required to define any DSP firmware audio topology and control objects from
userspace.
The following objects are supported :-
o kcontrols including TLV controls.
o DAPM widgets and graph elements
o Vendor bespoke objects.
o Coefficient data
o FE PCM capabilities and config.
o BE link capabilities and config.
o Codec <-> codec link capabilities and config.
o Topology object manifest.
The file format is simple and divided into blocks for each object type and
each block has a header that defines it's size and type. Blocks can be in
any order of type and can either all be in a single file or spread across
more than one file. Blocks also have a group identifier ID so that they can
be loaded and unloaded by ID.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/tas3a227e' and 'asoc/topic/tas571x' into asoc-next
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Binding header file for clock input selection and configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/pxa', 'asoc/topic/qcom' and 'asoc/topic/rcar' into asoc-next
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This patch adds apq8016 lpass driver support. APQ8016 has 4 MI2S which
can be routed to one internal codec and 2 external codec interfaces.
Primary, Secondary, Quaternary I2S can do Rx(playback) and Tertiary and
Quaternary can do Tx(capture).
Tested-by: Kenneth Westfield <kwestfie@codeaurora.org>
Acked-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi' and 'asoc/topic/gpio-jack' into asoc-next
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Adding the jack type to the dt-bindings directory will allow for device
tree files to specify the type of audio jacks that are present for a
board.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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into asoc-next
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Current DPCM is caring only FE format. but it will be no sound
if FE/BE was below style, and user selects S24_LE format.
FE: S16_LE/S24_LE
BE: S16_LE
DPCM can rewrite the format, so basically we don't want to
constrain with the BE constraints. But sometimes it will be trouble.
This patch adds new .dpcm_merged_format on struct snd_soc_dai_link.
DPCM will use FE / BE merged format if .struct snd_soc_dai_link
has it. We can have other .dpcm_merged_xxx in the future
.dpcm_merged_foramt
.dpcm_merged_rate
.dpcm_merged_chan
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current struct snd_soc_dai_link has many members, but definition order
was random. Especially, bool / bit field are defined randomly.
This patch tidyups these definition order to calculate data alignment
easy.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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We can know if dmic is used by reading the value of dmic1_data_pin
and dmic2_data_pin. Also IRQ must be used if codec JD or button
detection function is used. So, dmic_en and en_jd_func can be remove
from platform data.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Whether residue can be reported or not is not a property of the audio
controller but of the DMA controller. The FLAG_NO_RESIDUE was initially
added when the DMAengine framework had no support for describing the residue
reporting capabilities of the controller. Support for this was added quite a
while ago and recently the DMAengine framework started to complain if a
driver does not describe its capabilities and a lot of patches have been
merged that add support for this where it was missing. So it should be safe
to assume that driver on actively used platforms properly implement the DMA
capabilities API.
This patch makes the FLAG_NO_RESIDUE internal and no longer allows audio
controller drivers to manually set the flag. If a DMA driver against
expectations does not support reporting its capabilities for now the generic
DMAengine PCM driver will now emit a warning and simply assume that residue
reporting is not supported. In the future this might be changed to aborting
with an error.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some CODECs have a significant number of DAPM routes and for each route,
when it is added to the card, the entire card widget list must be
searched. When adding routes it is very likely, however, that adjacent
routes will require adjacent widgets. For example all the routes for a
mux are likely added in a block and the sink widget will be the same
each time and it is also quite likely that the source widgets are
sequential located in the widget list.
This patch adds a cache to the DAPM context, this cache will hold the
source and sink widgets from the last call to snd_soc_dapm_add_route for
that context. A small search of the widget list will be made from those
points for both the sink and source. Currently this search only checks
both the last widget and the one adjacent to it.
On wm8280 which has approximately 500 widgets and 30000 routes (one of
the largest CODECs in mainline), the number of paths that hit the cache
is 24000, which significantly improves probe time.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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A demux is conceptually similar to a mux. Where a mux has multiple input
and one output and selects one of the inputs to be connected to the output,
the demux has one input and multiple outputs and selects one of the outputs
to which the input gets connected.
This similarity makes it straight forward to support them in DAPM using the
existing mux support, we only need to swap sinks and sources when initially
setting up the paths.
The only slightly tricky part is that there can only be one control per
path. Since mixers/muxes are at the sink of a path and a demux is at the
source and both types want a control it is not possible to directly connect
a demux output to a mixer/mux input. The patch adds some sanity checks to
make sure that this does not happen.
Drivers who want to model hardware which directly connects a demux output
to a mixer/mux input can do this by inserting a dummy widget between the
two. E.g.:
{ "Dummy", "Demux Control", "Demux" },
{ "Mixer", "Mixer Control", "Dummy" },
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit 57295073b6ac ("ASoC: dapm: Implement mixer input auto-disable")
added support for autodisable controls, controls whose values are only
written to the hardware when their respective widgets are powered up.
But it only added support for controls based on the mixer abstraction.
This patch add support for mux controls (DAPM controls based on the
enum abstraction) to be auto-disabled as well. As each mux can only have
a single control, there is no need to tie the autodisable widget to the
inputs (as is done for the mixer controls) it can be tided directly to
the mux widget itself.
Note that it is assumed that the first entry in a autodisable mux
control will always represent the off state for the mux and is what the
mux will be set to whilst it is disabled.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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xnitmes is clearly intended to be xnitems, but all other macros just
refer to this as xitems, so change it to that.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently drivers are responsible for managing the bias_level field of
their DAPM context. The DAPM state itself is managed by the DAPM core
though and the core has certain expectations on how and when the bias_level
field should be updated. If drivers don't adhere to these undefined
behavior can occur.
This patch adds a few helper functions for manipulating the DAPM context
state, each function with a description on when it should be used and what
its effects are. This will also help us to move more of the bias_level
management from drivers to the DAPM core.
For convenience also add snd_soc_codec_* wrappers around these helpers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DAPM context in the snd_soc_codec struct is redundant and scheduled to
be replaced by the DAPM context in the snd_soc_component struct. This patch
introduces a new helper function snd_soc_codec_get_dapm() which should be
used for getting the DAPM context for a CODEC rather then directly
accessing the dapm field. Once there are no more direct users of the dapm
field left it is possible to transparently switch all drivers to the
component DAPM context by updating snd_soc_codec_get_dapm() function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds new registers as per HD audio Spec like capability registers
for processing pipe, software position based FIFO, Multiple Links and Global
Time Synchronization.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HDA codec drivers can be matched using vendor id and revision id typically.
So provide a match function which does this and is loaded when driver hasn't
provided one (default behaviour)
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge back the latest HD-audio stuff for further development.
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Build emux_proc.o and drop the unneeded ifdefs.
Replace the left CONFIG_PROC with the new CONFIG_SND_PROC_FS.
Along with this, fix the build of emux_oss.o in Makefile, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We may disable proc fs only for sound part, to reduce ALSA
memory footprint. So add CONFIG_SND_PROC_FS and replace the
old CONFIG_PROC_FSs in alsa code.
With sound proc fs disabled, we can save about 9KB memory
size on X86_64 platform.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a helper to create the IEC958 channel status from an ALSA
snd_pcm_runtime structure, taking account of the sample rate and
sample size.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Reviwed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a helper for the EDID like data structure, which is typically passed
from a HDMI adapter to its associated audio driver. This informs the
audio driver of the capabilities of the attached HDMI sink.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The file is moved to hda core and renamed to hdac_i915.c, so can be used
by both legacy HDA driver and new Skylake audio driver.
- Add snd_hdac_ prefix to the public APIs.
- The i915 audio component is moved to core bus and dynamically allocated.
- A static pointer hdac_acomp is used to help bind/unbind callbacks to get
this component, because the sound card's private_data is used by the azx
chip pointer, which is a legacy structure. It could be removed if private
_data changes to some core structure which can be extended to find the
bus.
- snd_hdac_get_display_clk() is added to get the display core clock for
HSW/BDW.
- haswell_set_bclk() is moved to hda_intel.c because it needs to write the
controller registers EM4/EM5, and only legacy HD-A needs it for HSW/BDW.
- Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h
and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have
different layout for the extended mode registers.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add support for enabling codec wakeup override signal to allow
re-enumeration of the controller on SKL after resume from low power state.
In SKL, HDMI/DP codec and PCH HD Audio Controller are in different power
wells, so it's necessary to reset display audio codecs when power well on,
otherwise display audio codecs will disappear when resume from low power
state.
Reset steps when power on:
enable codec wakeup -> azx_init_chip() -> disable codec wakeup
v3 by Jani: Simplify to only support toggling the appropriate chicken bit.
v4 by Han: add explanation and specify the hw swquence.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Jani Nikula <jani.nikula@intel.com>
Acked-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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