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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Merge tag 'asoc-v4.2' into asoc-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.
# 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>"
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>
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>
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>
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>
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>
Yet another regression by the transition to regmap cache; for better
usability, we had the fake mute control using the zero amp value for
Conexant codecs, and this was forgotten in the new hda core code.
Since the bits 4-7 are unused for the amp registers (as we follow the
syntax of AMP_GET verb), the bit 4 is now used to indicate the fake
mute. For setting this flag, snd_hda_codec_amp_update() becomes a
function from a simple macro. The bonus is that it gained a proper
function description.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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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Merge tag 'asoc-v4.2' of 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.
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
Commit 57295073b6 ("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>
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>
A few fixes for v4.1, none earth shattering and mostly driver related
except for one change to fix !PM builds for Intel platforms which is
done by adding stubs in the core so other platforms don't run into the
same issue.
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Merge tag 'asoc-v4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.1
A few fixes for v4.1, none earth shattering and mostly driver related
except for one change to fix !PM builds for Intel platforms which is
done by adding stubs in the core so other platforms don't run into the
same issue.
A flag "link_power_control" is added to indicate whether a codec needs to
control the link power. And a new bus ops link_power() is defined for the
codec to request to enable/disable the link power.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looks like audigy emu10k2 (probably emu10k1 - sb live too) support two
modes for DMA. Second mode is useful for 64 bit os with more then 2 GB
of ram (fixes problems with big soundfont loading)
1) 32MB from 2 GB address space using 8192 pages (used now as default)
2) 16MB from 4 GB address space using 4096 pages
Mode is set using HCFG_EXPANDED_MEM flag in HCFG register.
Also format of emu10k2 page table is then different.
Signed-off-by: Peter Zubaj <pzubaj@marticonet.sk>
Tested-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
rajeev-dlh.kumar@st.com email-id doesn't exist anymore as I have left the
company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux@gmail.com>
Signed-off-by: Rajeev Kumar <rajeevkumar.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
Jack snd_kcontrols can now be created during snd_jack_new()
or by later calling snd_jack_add_new_kctls().
This patch creates the jacks during the initialisation stage
for both phantom and non phantom jacks.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dont create input devices for phantom jacks.
Here, we extend snd_jack_new() to support phantom jack creating:
pass in a bool param for [non-]phantom flag, and a bool param
initial_jack to indicate whether we need to create a kctl at
this stage.
We can also add a kctl to the jack after its created meaning we
can now integrate the HDA and ASoC jacks.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a static method get_available_index() to
allocate the index of new jack kcontrols and also adds
jack_kctl_name_gen() which is used to ensure compatibility
with jack naming by removing " Jack" from some incorrectly
passed names.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the ALSA jack core registers only input devices for each jack
registered. These jack input devices are not readable by userspace devices
that run as non root. This patch series will implement kctls inside the
core jack part, including kctls creating, status changing report, for both
HD-Audio and ASoC jack. This allows non root userspace to read jack status
and act on it.
This patch adds a new API called snd_jack_add_new_kctl(), which will create
a kcontrol, add it to the card, and also attach it to the jack kctl list.
This patch also initialises the jack kctl list after jack is newed, and
reports kctl status when jack insertion/removal events occur.
snd_jack_new() is updated in the following patches to also support creating
phantom jacks and jack kcontrols. We then remove these duplicated features
from HDA jack and have jack kctls handled by core throughout HDA and ASoC.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Modified-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Reveiwed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
On a 64-bit system there is a 32-bit hole in struct snd_pcm_constraint_list
and then 32-bit padding at the end. Reordering things slightly gets rid of
the hole and padding, reducing the size of the struct by 50% from its
original size.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On a 64-bit system there are two 32-bit holes due to the alignment of 64-bit
fields. Reordering things slightly gets rid of those holes, reducing the
size of the struct by 17% percent of its original size.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few minor cleanups:
- Move the call of snd_info_minor_register() into snd_info_init() so
that we can call all proc-related stuff in a shot
- Add missing __init prefix to snd_info_minor_register()
- Return an error properly from snd_oss_info_register()
- Drop snd_info_minor_unregister() that is superfluous now
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since each proc entry is freed automatically by the parent, we don't
have to take care of its life cycle any longer. This allows us to
reduce a few more lines of codes.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the way to manage the resource release of proc
files: namely, let snd_info_free_entry() freeing the whole children.
This makes it us possible to drop the snd_device_*() management. Then
snd_card_proc_new() becomes merely a wrapper to
snd_info_create_card_entry().
Together with this change, now you need to call snd_info_free_entry()
for a proc entry created via snd_card_proc_new(), while it was freed
via snd_device_free() beforehand.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
seq_file is _the_ standard interface for simple text proc files.
Though, we still need to support the binary proc files and the text
file write, and also we need to manage the device disconnection
gracefully. Thus this patch just replaces the text file read code
with seq_file while keeping the rest intact.
snd_iprintf() helper function is now a macro to expand itself to
seq_printf() to be compatible with the existing code. The seq_file
object is stored to the unused entry->rbuffer->buffer pointer.
When the output size is expected to be large (greater than PAGE_SIZE),
the driver should set entry->size field beforehand. Then the given
size will be preallocated and the multiple show calls can be avoided.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct small copy and paste error where autodisable was not being
enabled for the SOC_DAPM_SINGLE_TLV_AUTODISABLE control.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
When CONFIG_PM_SLEEP is not selected, calling funcs
snd_soc_suspend and _resume will generate a compiling
issue.
Here add static inline stub functions to fix it.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This will be used by hda controller driver to
setup stream params in prepare. This function will
setup the bdl and periods.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controller helper codes to hda-core library.
The I/O access ops are added to the bus ops. The CORB/RIRB, the basic
attributes like irq# and iomap address, some locks and the list of
streams are added to the bus object, together with the stream object
and its helpers.
Currently the codes are just copied from the legacy driver, so you can
find duplicated codes in both directories. Only constants are removed
from the original hda_controller.h. More integration work will follow
in the later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More updates for v4.1, pretty much all drivers:
- Lots of cleanups from Lars, mainly moving things from the CODEC level
to the card level.
- Continuing improvements to rcar from Morimoto-san, pcm512x from
Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han,
and to rt5670 from Bard.
- Support for some non-DSP Qualcomm platforms, Google's Storm
platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC.
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Merge tag 'asoc-v4.1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.1
More updates for v4.1, pretty much all drivers:
- Lots of cleanups from Lars, mainly moving things from the CODEC level
to the card level.
- Continuing improvements to rcar from Morimoto-san, pcm512x from
Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han,
and to rt5670 from Bard.
- Support for some non-DSP Qualcomm platforms, Google's Storm
platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC.
Although some races in runtime PM refcount was fixed by the commit
[664c715573c2: ALSA: hda - Work around races of power up/down with
runtime PM], there is still a race in the following case:
CPU0: CPU1 :
runtime suspend:
codec->in_pm = 1
snd_hdac_power_up_pm():
pm_runtime_get_sync() skipped
suspend finished:
codec->in_pm = 0
snd_hdac_power_down_pm():
pm_runtime_put_*() is called!
For avoiding this situation, increment in_pm flag atomically when it's
non-zero, and decrement accordingly, to ensure that in_pm is set
consistently for the whole concurrent operations.
Also, since atomic_inc_not_zero() and atomic_dec_if_positive() are
lengthy inline functions, move snd_hdac_power_up_pm() and _down_pm()
to sound/hda/hdac_device.c as no inline functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Failing to register the debugfs entries is not fatal and will not affect
normal operation of the sound card. Don't abort the card registration if
soc_dpcm_debugfs_add() fails.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, snd_hdac_power_up()/down() helpers checks whether the codec
is being in pm (suspend/resume), and skips the call of runtime get/put
during it. This is needed as there are lots of power up/down
sequences called in the paths that are also used in the PM itself. An
example is found in hda_codec.c::codec_exec_verb(), where this can
power up the codec while it may be called again in its power up
sequence, too.
The above works in most cases, but sometimes we really want to wait
for the real power up. For example, the control element get/put may
want explicit power up so that the value change is assured to reach to
the hardware. Using the current snd_hdac_power_up(), however,
results in a race, e.g. when it's called during the runtime suspend is
being performed. In the worst case, as found in patch_ca0132.c, it
can even lead to the deadlock because the code assumes the power up
while it was skipped due to the check above.
For dealing with such cases, this patch makes snd_hdac_power_up() and
_down() to two variants: with and without in_pm flag check. The
version with pm flag check is named as snd_hdac_power_up_pm() while
the version without pm flag check is still kept as
snd_hdac_power_up(). (Just because the usage of the former is fewer.)
Then finally, the patch replaces each call potentially done in PM with
the new _pm() variant.
In theory, we can implement a unified version -- if we can distinguish
the current context whether it's in the pm path. But such an
implementation is cumbersome, so leave the code like this a bit messy
way for now...
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96271
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we assumed that the node attributes like amp values remain
during the power state transition of the node itself. While this is
true for IDT/STAC codecs I've tested, but some other codecs don't seem
behaving in that way.
This patch implements a partial sync mechanism specific to the given
widget node. Now we've merged the regmap support, and it can be
easily written with regcache_sync_region().
Tested-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The delayed_work field in the snd_soc_dapm_context struct is now unused and
can be removed. Removing it reduces the size of the snd_soc_dapm_context
struct by ~50% from 100 bytes to 48 bytes.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The only two users of the suspend_bias_level field were two rather old
drivers which weren't exactly doing things by the book. Those drivers have
been updated and field is now unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Having to set different formats on the CPU side and the CODEC side of a DAI
link is usually indication that something is terribly wrong and in most
cases is a result of a broken driver that implements a set_fmt() callback
which does not follow the specification. In the past this feature has been
used to work around broken drivers, rather than fixing them. We don't really
want to encourage this, so remove support for setting different formats on
both ends of the link.
Along the way switch to static DAI format setup by setting the the dai_fmt
field of the snd_soc_dai_link rather than calling snd_soc_dai_fmt().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 16bit COEF read/write is pretty standard for many codecs, and they
can be cached in most cases -- more importantly, they need to be
restored at resume. For making this easier, add the cache support to
regmap. If the codec driver wants to cache the COEF access, set
codec->cache_coef flag and issue AC_VERB_GET_PROC_COEF with the coef
index in LSB 8 bits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio spec is inconvenient regarding the handling of stereo volume
controls. It can set and get only single channel at once (although
there is a special option to set the same value to both channels).
This patch provides a fake pseudo-register via the regmap access so
that the stereo channels can be read and written by a single call.
It'd be useful, for example, for implementing DAPM widgets.
A stereo amp pseudo register consists of the encoding like the normal
amp verbs but it has both SET_LEFT (bit 13) and SET_RIGHT (bit 12)
bits set. The regmap reads and writes a 16bit value for this pseudo
register where the upper 8bit is for the right chanel and the lower
8bit for the left channel.
Note that the driver doesn't recognize conflicts when both stereo and
mono channel registers are mixed. Mixing them would certainly confuse
the operation. So, use carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codecs may have own vendor-specific verbs, and we need to allow each
driver to give such verbs for cached accesses. Here a verb can be put
into a single array and looked through it at readable and writeable
callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp hash table was used for recording the cached reads of some
capability values like pin caps or amp caps. Now all these are moved
to regmap as well.
One addition to the regmap helper is codec->caps_overwriting flag.
This is set in snd_hdac_override_parm(), and the regmap helper accepts
any register while this flag is set, so that it can overwrite even the
read-only verb like AC_VERB_PARAMETERS. The flag is cleared
immediately in snd_hdac_override_parm(), as it's a once-off flag.
Along with these changes, the no longer needed amp hash and relevant
fields are removed from hda_codec struct now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes we need the uncached reads, e.g. for refreshing the tree.
This patch provides the helper function for that and uses it for
refreshing widgets, reading subtrees and the whole proc reads.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's start converting the access functions to regmap.
The first one is the simplest, just converting the codec parameter
read helper function snd_hda_param_read().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds an infrastructure to support regmap-based verb
accesses. Because o the asymmetric nature of HD-audio verbs,
especially the amp verbs, we need to translate the verbs as a sort of
pseudo registers to be mapped uniquely in regmap.
In this patch, a pseudo register is built from the NID, the
AC_VERB_GET_* and 8bit parameters, i.e. almost in the form to be sent
to HD-audio bus but without codec address field. OTOH, for writing,
the same pseudo register is translated to AC_VERB_SET_* automatically.
The AC_VERB_SET_AMP_* verb is re-encoded from the corresponding
AC_VERB_GET_AMP_* verb and parameter at writing.
Some verbs has a single command for read but multiple for writes. A
write for such a verb is split automatically to multiple verbs.
The patch provides also a few handy helper functions. They are
designed to be accessible even without regmap. When no regmap is set
up (e.g. before the codec device instantiation), the direct hardware
access is used. Also, it tries to avoid the unnecessary power-up.
The power up/down sequence is performed only on demand.
The codec driver needs to call snd_hdac_regmap_exit() and
snd_hdac_regmap_exit() at probe and remove if it wants the regmap
access.
There is one flag added to hdac_device. When the flag lazy_cache is
set, regmap helper ignores a write for a suspended device and returns
as if it was actually written. It reduces the hardware access pretty
much, e.g. when adjusting the mixer volume while in idle. This
assumes that the driver will sync the cache later at resume properly,
so use it carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an overriding exec_verb op to struct hdac_device so that the call
via snd_hdac_exec_verb() can switch to a different route depending on
the setup. The codec driver sets this field so that it can handle the
errors or applying quirks appropriately. Furthermore, this mechanism
will be used for smooth transition for the regmap support in later
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the sysfs files assigned to the codec device on the
bus which were formerly identical with hwdep sysfs files. Now it
shows only a few core parameter, vendor_id, subsystem_id, revision_id,
afg, mfg, vendor_name and chip_name.
In addition, now a widget tree is added to the bus device sysfs
directory for showing the widget topology and attributes. It's just a
flat tree consisting of subdirectories named as the widget NID
including various attributes like widget capability bits. The AFG
(usually NID 0x01) is always found there, and it contains always
amp_in_caps, amp_out_caps and power_caps files. Each of these
attributes show a single value. The rest are the widget nodes
belonging to that AFG. Note that the child node might not start from
0x02 but from another value like 0x0a.
Each child node may contain caps, pin_caps, amp_in_caps, amp_out_caps,
power_caps and connections files. The caps (representing the widget
capability bits) always contain a value. The rest may contain
value(s) if the attribute exists on the node. Only connections file
show multiple values while other attributes have zero or one single
value.
An example of ls -R output is like below:
% ls -R /sys/bus/hdaudio/devices/hdaudioC0D0/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/:
01/ 04/ 07/ 0a/ 0d/ 10/ 13/ 16/ 19/ 1c/ 1f/ 22/
02/ 05/ 08/ 0b/ 0e/ 11/ 14/ 17/ 1a/ 1d/ 20/ 23/
03/ 06/ 09/ 0c/ 0f/ 12/ 15/ 18/ 1b/ 1e/ 21/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/01:
amp_in_caps amp_out_caps power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/02:
amp_in_caps amp_out_caps caps connections pin_caps pin_cfg
power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/03:
.....
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now some codes and functionalities of hda_codec struct are moved to
hdac_device struct. A few basic attributes like the codec address,
vendor ID number, FG numbers, etc are moved to hdac_device, and they
are accessed like codec->core.addr. The basic verb exec functions are
moved, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few basic codes for communicating over HD-audio bus are moved to
struct hdac_bus now. It has only command and get_response ops in
addition to the unsolicited event handling.
Note that the codec-side tracing support is disabled temporarily
during this transition due to the code shuffling. It will be
re-enabled later once when all pieces are settled down.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define the common hd-audio driver and device types to bind over
snd_hda_bus_type publicly. This allows to implement other type of
device and driver code over hd-audio bus.
Now both struct hda_codec and struct hda_codec_driver inherit these
new struct hdac_device and struct hdac_driver, respectively.
The bus registration is done in subsys_initcall() to assure it
before any other driver registrations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dai-link params for codec-codec links were fixed. The fixed
link between codec and another chip which may be another codec,
baseband, bluetooth codec etc may require run time configuaration
changes. This change provides an optional alsa control to select
one of the params from a list of params.
Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it
typically results in less code than necessary for calling the non-inlined
version of the function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds the IRQ function support of rt5670. We use a flag
named dev_gpio in platform data to inform codec driver if the IRQ
function is used or not. Also, we export rt5670_set_jack_detect
for machine driver to pass the jack point.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A selection of changes for v4.1 so far. The main things are:
- Move of jack registration to the card where it belongs.
- Support for DAPM routes specified by both the machine driver and DT.
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Merge tag 'asoc-v4.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Changes for v4.1
A selection of changes for v4.1 so far. The main things are:
- Move of jack registration to the card where it belongs.
- Support for DAPM routes specified by both the machine driver and DT.
There are no users of snd_soc_jack_new() left and new users should use
snd_soc_card_jack_new() instead. So remove the function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Jacks are typically card level elements, but are currently registered with a
CODEC. When it was originally introduced snd_soc_jack_new() took a
snd_soc_card as its parameter, but at that time DAPM was only implemented at
the CODEC level and there was only one CODEC per card. This made it clear
which CODEC to use for the jack DAPM operations. But the multi-component
patchset added support for having multiple CODECs per card and with it the
API was updated to register jacks with a specific CODEC instance instead.
Subsequently DAPM support at the card level has been introduced, but the
snd_soc_jack_new() API has so remained unchanged.
This leaves us with the issue that the DAPM pins that are managed by the
jack detection logic usually are part of the card DAPM context but are
accessed through a CODEC DAPM context. Currently this works fine, but might
break in the future if we take a more hierarchical approach to DAPM
contexts.
Furthermore with componentization progressing systems that do not register
a snd_soc_codec might appear, while these system may still want to able to
register a jack.
This patch addresses these issues by adding a new function called
snd_soc_card_jack_new() that can be used to register jacks with the card
rather than a CODEC.
This new function is mostly identical to snd_soc_jack_new() except that it
additionally allows to directly specify the DAPM pins associated with the
jack. This was done since most users of snd_soc_jack_new() typically call
snd_soc_jack_add_pins() right after it, which is not necessary with the new
API and allows to reduce the amount of boiler plate code.
The old snd_soc_jack_new() is re-implemented as a wrapper around
snd_soc_card_jack_new().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fix spelling typo found in alsa-driver-api.xml.
It is because this file is generated from comments in source files,
I have to fix source files.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revive snd_device_disconnect() again so that it can be called from the
individual driver. This time, HD-audio will need it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current helper functions, snd_soc_of_parse_audio_simple_widgets()
and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes
without caring if they are already set by using build-in widgets and
routes in the card driver. So there could be one of them, build-in one
or Device Tree one, overrided by the other depending on which one was
assigned later.
This patch adds an extra pair of dapm_widgets and dapm_routes for DT
use only so as to prevent unexpected overriding.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA core with commit 257f8cce5d - "ALSA: pcm: Allow nonatomic trigger
operations" allows trigger ops to implemented as nonatomic. For ASoC, we can
specify this in dailinks and is updated while snd_pcm is created
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a helper to set pcm format directly from params
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
can be removed without breaking git-bisect now
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce more generic .get_time_info to retrieve
system timestamp and audio timestamp in single routine.
Backwards compatibility is preserved with same functionality
as with .wall_clock method (to be removed in following commits
to avoid breaking git bisect)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio timestamps can be extracted from sample counters, wall clocks,
PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This
patch provides the ability to report timestamping capabilities, select
timestamp types and retrieve timestamp accuracy, if supported.
Details can be found in Documentations/sound/alsa/timestamping.txt
This functionality is introduced by reclaiming the reserved_aligned
field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a
in snd_pcm_status to provide userspace with selection/query capabilities.
Additional driver_tstamp and audio_tstamp_accuracy fields are also added.
snd_pcm_mmap_status remains a read-only structure with only
the audio timestamp value accessible from user space. The selection
of audio timestamp type is done through snd_pcm_status only
This commit does not impact ABI and does not impact the default
behavior. By default audio timestamp is aligned with hw_pointer and
reports the DMA position. Backwards compatibility is handled by using
the HDAudio wall clock for playback and the hw_ptr for all other
cases.
For timestamp selection a new STATUS_EXT ioctl is introduced with
read/write parameters. Alsa-lib will be modified to make use of
STATUS_EXT.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The autoload lock became already superfluous due to the recent rework
of autoload code. Let's drop them now. This allows us to simplify a
few codes nicely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves the driver object initialization and allocation to
each driver's module init/exit code like other normal drivers. The
snd_seq_driver struct is now published in seq_device.h, and each
driver is responsible to define it with proper driver attributes
(name, probe and remove) with snd_seq_driver specific attributes as id
and argsize fields. The helper functions snd_seq_driver_register(),
snd_seq_driver_unregister() and module_snd_seq_driver() are used for
simplifying codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use const string pointer instead of copying the id string to each
object. Also drop the status and list fields of snd_seq_device struct
that are no longer used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've used the old house-made code for binding the sequencer device
and driver. This can be far better implemented with the standard
bus nowadays.
This patch refactors the whole sequencer binding code with the bus
/sys/bus/snd_seq. The devices appear as id-card-device on this bus
and are bound with the drivers corresponding to the given id like the
former implementation. The module autoload is also kept like before.
There is no change in API functions by this patch, and almost all
transitions are kept inside seq_device.c. The proc file output will
change slightly but kept compatible as much as possible.
Further integration works will follow in later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use generic snapshot of trigger_tstamp if low-level driver or
hardware can get a more precise value for better audio/system time
synchronization.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing too exciting here yet, a small optimization for DAPM from
Lars-Peter and a few small bits and pieces for drivers but nothing
that really stands out.
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Merge tag 'asoc-v3.19-rc2' into asoc-linus
ASoC: Updates for v3.20
Nothing too exciting here yet, a small optimization for DAPM from
Lars-Peter and a few small bits and pieces for drivers but nothing
that really stands out.
# gpg: Signature made Tue 30 Dec 2014 00:15:48 HKT using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: key AF88CD16: no public key for trusted key - skipped
# gpg: key AF88CD16 marked as ultimately trusted
# gpg: key 5621E907: no public key for trusted key - skipped
# gpg: key 5621E907 marked as ultimately trusted
# 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>"
For assigning sysfs entries for a card device from the driver,
introduce a new helper function, snd_card_add_dev_attr(). In this
way, we can avoid the possible race between the device registration
and the sysfs addition / removal.
The driver can pass a new attribute group to add freely. This has to
be called before snd_card_register().
Currently, up to two extra groups can be added. More than that, it'll
return an error.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
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Merge tag 'asoc-v3.20-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.20
More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
Conflicts:
sound/soc/intel/sst-mfld-platform-pcm.c
Instead of calling device_create_file() manually, assign the static
attribute group entries at the device registration. This simplifies
the error handling and avoids the possible races.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the device is no longer hidden but embedded into each component,
we no longer need snd_get_device(). Let's drop it and relevant codes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that all callers have been replaced with
snd_device_register_for_dev(), let's drop the obsolete device
registration code and concentrate only on the code handling struct
device directly. That said,
- remove the old snd_device_register(),
- rename snd_device_register_for_dev() with snd_device_register(),
- drop superfluous arguments from snd_device_register(),
- change snd_unregister_device() to pass the device pointer directly
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this one embeds the struct device into struct
snd_compr. As the dev field wasn't used beforehand, it's reused as
the new device struct.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this changes the device management for rawmidi,
embedding the struct device into struct snd_rawmidi. The required
change is more or less same as hwdep device.
The currently unused dev field is reused as the new embedded struct
field now.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, at this time we embed the struct device into
PCM object. However, this needs a bit more caution: struct snd_pcm
doesn't own one device but two, for both playback and capture! Thus
not struct snd_pcm but struct snd_pcm_str object contains the device.
Along with this change, pcm->dev field is dropped for avoiding
confusion. It was meant to point to a non-standard parent. But,
since now we can touch each struct device directly, we can manipulate
the parent field easily there, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patch, this one embeds the device object into hwdep
object. For a proper object lifecycle, it's freed in the release
callback.
This also allows us to create sysfs entries via passing to the groups
field of the device without explicit function calls. Since each
driver can see the device and touch its groups field directly, we
don't need to delegate in hwdep core any longer. So, remove the
groups field from snd_hwdep, and let the user (in this case only
hda_hwdep.c) modify the device groups.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch embeds a struct device for the control device into the card
object and avoid the device creation at registration time.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper function snd_device_initialize() to initialize
the device object for sound devices.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of open-coding the search over the control file loop, provide
a helper function for the preferred subdevice assigned to the current
process.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the further work on embedding struct
device into each sound device instance. It changes
snd_register_device*() helpers to receive the device object directly
for skipping creating a device there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Protect the call with a mutex, as this may be called in parallel
(either from the PCM rate change and the clock change).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define snd_ak4114_suspend() and snd_ak4114_resume() functions to
handle PM properly, stopping and restarting the work at PM.
Currently only ice1712/juli.c deals with the PM and ak4114, so fix the
calls there appropriately.
The same PM functions are defined in ak4113.c, too, although they
aren't currently called yet (ice1712/quartet.c may be enhanced to
support PM later).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ak4114 work calls its callback and the callback invokes
ak4114_reinit(), it stalls due to flush_delayed_work(). For avoiding
this, control the reentrance by introducing a refcount. Also
flush_delayed_work() is replaced with cancel_delayed_work_sync().
The exactly same bug is present in ak4113.c and fixed as well.
Reported-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add helper functions to allow drivers to specify several disjoint
ranges for a variable. In particular, there is a codec (PCM512x) that
has a hole in its supported range of rates, due to PLL and divider
restrictions.
This is like snd_pcm_hw_constraint_list(), but for ranges instead of
points.
Signed-off-by: Peter Rosin <peda@axentia.se>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
make the sta32x driver usable with device tree configs. Code is heavily based
on the sta350 driver.
Signed-off-by: Thomas Niederprüm <niederp@physik.uni-kl.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
In some cases it is necessary to before additional operations after the
device has been initialized and before the device is registered. This can
for example be resetting the device.
This patch introduces a new function snd_soc_alloc_ac97_codec() which is
similar to snd_soc_new_ac97_codec() except that it does not register the
device. Any users of snd_soc_alloc_ac97_codec() are responsible for calling
device_add() manually.
Fixes: 6794f709b7 ("ASoC: ac97: Drop delayed device registration")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Queues are used both for scheduling playback events and for assigning
timestamps to recorded events, so it is easy to need quite a lot of
them, especially on a multi-user system. Additionally, the actual
queue objects are allocated dynamically, so it does not really make
sense to have a low limit. Increase it to something still sane.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no more users of this field left so it can finally be removed.
New users should use snd_soc_dapm_to_codec(w->dapm);
The reason why it is removed is because it doesn't fit to well anymore in
the componentized ASoC hierarchy, where DAPM works on the snd_soc_component
level. And the alternative of snd_soc_dapm_to_codec(w->dapm) typically
generates the same amount of code, so there is really no reason to keep it.
For automatic conversion the following coccinelle semantic patch can be used:
// <smpl>
@@
struct snd_soc_dapm_widget *w;
@@
-w->codec
+snd_soc_dapm_to_codec(w->dapm)
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Call clk_prepare_enable() and clk_disable_unprepare() for cpu dai
clock and codec dai clock in dai statup and shutdown callbacks. This
to make sure the related clock are enabled when the audio device is
used.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following compilation error:
include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’:
include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration]
if (snd_BUG_ON(!stream))
snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly,
so include it.
Signed-off-by: Qais Yousef <qais.yousef@imgtec.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
L/R channel will be switched if under/over flow error happen on
Renesas R-Car sound device by the HW bugs. Then, HW restart is required
for salvage. This patch add salvage support for SRC.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch adds the MICBIAS VDD setting in the platform data. It can be set to
1V8 or 3V3 in the MICBIAS VDD.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some setups it is necessary to change the DAI link format at runtime.
This patch factors out the code that does the initial static DAI link format
configuration into a separate helper function which can be used board
drivers as well.
This allows board drivers that have to change the DAI link format at runtime
to reuse it instead of having to manually change the format on all DAIs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove function declarations for functions that don't have a matching
implementation.
For snd_pcm_build_linear_format the implementation was removed in
64d27f96cb ("[ALSA] Support 3-bytes 24bit format in PCM OSS
emulation"). All the others never had one (as far as git history goes).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The various PCM and hwdep allocation functions in this driver take a pointer
to a pointer of a PCM/hwdep where if this parameter is provided the newly
allocated object is stored. All callers pass NULL though, so remove the
parameter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most callers of snd_wss_pcm(), snd_wss_timer() and snd_cs4236_pcm() pass
NULL as the last parameter, some callers pass a pointer but never use it
after the function has been called and only a few callers pass a pointer and
actually use it. The later is only the case for snd_wss_pcm() for
snd_cs4236_pcm() and it is possible to get the same PCM object by accessing
the pcm field of the snd_wss struct that was passed as the first parameter.
This function removes the last parameters from the functions mentioned above
and updates the callers which used it to use chip->pcm instead. This allows
us to slightly simplify the functions since they don't have to check and set
the last parameter anymore which makes the code slightly shorter and
cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All callers of snd_sb16dsp_pcm() always pass the pcm field of the first
parameter as the last parameter. Simplify the function by moving this inside
the function itself. This makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_sb8dsp_pcm() and snd_sb8dsp_midi() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_gf1_pcm_new() and snd_gf1_rawmidi_new() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_es1688_pcm() takes a pointer to a pointer of a PCM where if this
parameter is provided the newly allocated PCM is stored. This PCM is also
available from the pcm field of the snd_es1688 struct that got passed to the
same function. This patch updates all callers which passed a pointer to use
that field instead and then removes the parameter from the function. This
makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ad1816a_pcm() and snd_ad1816a_timer() take a pointer to a pointer of a
PCM/timer where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain.
Some audio devices require notification of drain events
in order to properly drain and shutdown an audio stream.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__fls has the same semantics as ld2, so there is no need to re-implement it.
Furthermore a lot of architectures have custom implementations of __fls that
are able to use special hardware instructions to compute the result. This
makes the code slightly shorter and faster.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The difference between __ffs and ffs is that ffs will return a one based
index whereas __ffs will return a zero based index. Furthermore ffs will
check if the passed value is zero and return zero in that case, whereas
__ffs behavior is undefined if the passed parameter is 0.
Since we already check if the mask is 0 before calling ffs and also subtract
1 from the result __ffs is the better choice.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hw_params struct has a parameter that contains the period size in bytes.
This can be used instead of deriving the value from other parameters. This
is similar to e.g. params_buffer_bytes()
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline functions instead of macros for the remaining params_*()
helpers that have not been converted yet. This is slightly cleaner and
offers better type safety.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both SNDRV_PCM_IOCTL1_FALSE and SNDRV_PCM_IOCTL1_TRUE are unused and have in
fact never been used (at least as far as the git history goes).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Those two functions are not used anywhere and also their name is a bit to
generic to be in a global header, so remove them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a copy and paste error in the kernel doc description for the params_*()
functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For legacy reasons the ASoC framework assumes that a CODEC INPUT or OUTPUT
widget that is not explicitly connected to a external source or sink is
potentially connected to a source or a sink and hence the framework treats
the widget itself as source (for INPUT) or sink (for OUTPUT). For this
reason a INPUT or OUTPUT widget that is really not connected needs to be
explicitly marked as so.
Setting the card's fully_routed flag will cause the ASoC core, once that all
widgets and routes have been registered, to go through the list of all
widgets and mark all INPUT and OUTPUT that are not externally connected as
non-connected. This essentially negates the default behaviour of treating
INPUT or OUTPUT widgets without external routes as sources or sinks.
This patch takes a different approach while getting the same result. Instead
of first marking INPUT and OUTPUT widgets as sinks/sources and then later
marking them as non-connected, just never mark them as a sink or a source if
the fully_routed flag is set on a card.
This requires a lot less code and also results in a slightly faster card
initialization since there is no need to iterate over all widgets and check
whether the INPUT and OUTPUT widgets are connected or not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
* ALSA core
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
* USB-audio
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
* FireWire
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
* HD-audio
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
* ASoC
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
* Others
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle
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Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
ALSA core:
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
USB-audio:
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
FireWire:
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
HD-audio:
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
ASoC:
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
Others:
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle"
* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
ALSA: pcxhr: NULL dereference on probe failure
ALSA: lola: NULL dereference on probe failure
ALSA: hda - Add "eapd" model string for AD1986A codec
ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
ALSA: oxfw: Add hwdep interface
ALSA: oxfw: Add support for capture/playback MIDI messages
ALSA: oxfw: add support for capturing PCM samples
ALSA: oxfw: Add support AMDTP in-stream
ALSA: oxfw: Add support for Behringer/Mackie devices
ALSA: oxfw: Change the way to start stream
ALSA: oxfw: Add proc interface for debugging purpose
ALSA: oxfw: Change the way to make PCM rules/constraints
ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
ALSA: oxfw: Change the way to name card
ALSA: dice: Add support for MIDI capture/playback
ALSA: dice: Add support for capturing PCM samples
ALSA: dice: Support for non SYT-Match sampling clock source mode
ALSA: dice: Add support for duplex streams with synchronization
ALSA: dice: Change the way to start stream
ALSA: jack: Add dummy snd_jack_set_key() definition
...
* support for mx6sl and mx6sx
* OMAP HDMI audio rewrite to make it finally work
* OMAP video PLL work to prepare for new DRA7xx SoCs
* simplefb DT related improvements
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Merge tag 'fbdev-3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux
Pull fbdev updates from Tomi Valkeinen:
- support for mx6sl and mx6sx
- OMAP HDMI audio rewrite to make it finally work
- OMAP video PLL work to prepare for new DRA7xx SoCs
- simplefb DT related improvements
* tag 'fbdev-3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux: (81 commits)
video: uvesafb: Deletion of an unnecessary check before the function call "platform_device_put"
video: fbdev-VIA: Deletion of an unnecessary check before the function call "framebuffer_release"
video: fbdev-MMP: Deletion of an unnecessary check before the function call "mmp_unregister_path"
video: mx3fb: Deletion of an unnecessary check before the function call "backlight_device_unregister"
video: fbdev-OMAP2: Deletion of unnecessary checks before the function call "i2c_put_adapter"
video: fbdev-SIS: Deletion of unnecessary checks before the function call "pci_dev_put"
video: smscufx: Deletion of unnecessary checks before the function call "vfree"
video: udlfb: Deletion of unnecessary checks before the function call "vfree"
video: uvesafb: Deletion of an unnecessary check before the function call "uvesafb_free"
video: fbdev-LCDC: Deletion of an unnecessary check before the function call "vfree"
video: fbdev: arkfb: suppress build warning
video: fbdev: s3fb: suppress build warning
video: fbdev: vt8623fb: suppress build warning
OMAPDSS: hdmi5: Fix bit field for IEC958_AES2_CON_SOURCE
OMAPDSS: hdmi: Remove __exit qualifier from hdmi_uninit_output()
OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback
OMAPDSS: Remove all references to obsolete HDMI audio callbacks
ASoC: omap: Remove obsolete HDMI audio code and Kconfig options
OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio
OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver
...
For fixing a build error with CONFIG_SND_JACK=n
sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of changes this time around, the usual set of driver
updates and a huge bulk of cleanups from Lars-Peter. Probably the most
interesting thing for most users is the Intel driver updates which will
(with some more machine integration work) enable support for newer x86
laptops.
- Conversion of AC'97 drivers to use regmap, bringing us closer to the
removal of the ASoC level I/O code.
- Clean up a lot of old drivers that were open coding things that have
subsequently been implemented in the core.
- Some DAPM performance improvements.
- Removal of the now seldom used CODEC mutex.
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers.
- Support for Samsung boards using rt5631 as the CODEC.
- Removal of the obsolete AFEB9260 machine driver.
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks.
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Merge tag 'asoc-v3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.19
Lots and lots of changes this time around, the usual set of driver
updates and a huge bulk of cleanups from Lars-Peter. Probably the most
interesting thing for most users is the Intel driver updates which will
(with some more machine integration work) enable support for newer x86
laptops.
- Conversion of AC'97 drivers to use regmap, bringing us closer to the
removal of the ASoC level I/O code.
- Clean up a lot of old drivers that were open coding things that have
subsequently been implemented in the core.
- Some DAPM performance improvements.
- Removal of the now seldom used CODEC mutex.
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers.
- Support for Samsung boards using rt5631 as the CODEC.
- Removal of the obsolete AFEB9260 machine driver.
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks.
Mark Brown