Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
Signed-off-by: Taylor Hutt <thutt@chromium.org>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out. Also this patch includes minor
clean-ups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers. They have no difference now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the order of the output-path list in a way from the DAC to the
target pin. Also now the list include the target pin, too.
Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.
This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
i - Mic-in
ii- Line-in
* mixer device
Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a master volume and mute control of playback for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When switch HP independent mode, mute/unmute connctions of mixer which is
connected to headphone for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"ret" is supposed to be signed here. The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.
BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the existing aa-loop list for simplifying the check for analog
low-current mode. Also fix the stream count test for playback streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.
This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of ignoring the invalid pin configuration, return the error.
This will avoid unexpected crash, anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create capture-related mixer elements dynamically from the parsed
ADCs and input-pins instead of fixed values for each codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM. Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VIA codecs, we shouldn't create a substream for independent HP mode,
when no individual HP DAC is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.
The smart51 setup doesn't work with this change, and will be fixed
in the next commits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute the outputs via pin-controls instead of amps for the auto-mute
handling. This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.
With this change, we can use vmaster for the master control easily now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.
Also, fixed typos of vt1708_jack_detect in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of giving the fixed ADC list, parse the widgets and fill in
ADCs dynamically.
Also, probe the stereo-mixer input more dynamically, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently VIA driver controls the power-state of each pin per jack
detection. But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.
For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.
Fix originally suggested by Liam on IM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others. There's a pending patch to make the directory build as standard
but it's not merged yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The via driver spews warnigs like
hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 13882a82ee (optimize iso queueing by setting
wake only after the last packet), drivers are required to call
fw_iso_context_queue_flush() after queueing a batch of packets.
The missing call would have an effect only if the controller
queue underruns, but then the DMA would stop completely.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Scott Jiang <scott.jiang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us. So drop the redundant checking/message.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a machine driver to support the ADAU1701 SigmaDSP processors on
Analog Devices BF5XX evaluation boards.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I no longer work at Bluewater Systems. Update my email address accordingly. I
have deleted my email address from C files rather than change it. This
was suggested by several people, since the commit from my new email
address will cause scripts/get_maintainer.pl to function properly. I
have not added the .mailmap entry as suggested by Joe because I think
it is no longer necessary if I touch all the files which had my name
in them.
Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Cc: Andre Renaud <andre@bluewatersys.com>
Cc: H Hartley Sweeten <hsweeten@visionengravers.com>
Cc: Russell King <linux@arm.linux.org.uk>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Andrew Victor <avictor.za@gmail.com>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Anton Vorontsov <cbou@mail.ru>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Jesper Juhl <jj@chaosbits.net>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: trivial@kernel.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
This is a reworked patch from Creative to change the PLL code to address
unreliable 44100Hz initialization.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time with soc_widget_update_bits reflecting recent soc_update_bits changes.
Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP
devices have mixer components that can alter power usage and hence require full
DAPM support.
This provides a generic widget IO operation wrapper in preparation for
future patches that implement platform driver DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kill tasklet usage in rawmidi core code. Use workq for the event callback
instead of tasklet (which is used only in core/seq/seq_midi.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a partial revert of 28f65c11f2 ("treewide: Convert uses of
struct resource to resource_size(ptr)") as the code is rewritten
in the sound tree and thus the change is obsolete.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Initialise model-specific DAC and ADC parts.
Add controls for output and mic source selection.
Rename some mixer controls according to ControlNames.txt.
Remove Playback switches for Line-in and IEC958-in - these
were controlling the input mute/unmute which affected
capture too. Use the capture switches to control the
input mute/unmute instead - it's less confusing.
Initialise the WM8775 to invert the left-right clock
to swap the left and right channels of the mic and aux
input.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to facilitate merging with the register map I/O replace the use
of control_data for the bulk writes with direct lookup of the client data
from the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.
Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For clarity and to help ongoing refactoring in this area create a new file
to contain the physical I/O functions, separating them out from the cache
operations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If given a -1 cmd parameter then make_exec_verb() returns -1 without
setting the res output value.
Prior to this change snd_hda_codec_read() assumed that make_exec_verb()
unconditionally set res regardless of the cmd value.
This change explicitly checks the make_exec_verb() return value before
consuming the potentially unset res value.
Signed-off-by: Greg Thelen <gthelen@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a whole bunch of functions which just call straight through to
do_hw_read(). Simplify this situation by removing them and using hw_read()
directly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Using static inline functions can reduce compilation messages
and macro misuse.
sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’:
sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly
during the clean-up of auto-mute function. Fixed now.
Tested-by: Borislav Petkov <bp@alien8.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the necessary details to support the PCIe version of
E-MU's 0404 card.
From comparing the PCBs it seems the PCIe version just added a PCIe
chipset and left all other components pretty much in place.
For anyone intrigued to take a look at the PCB there are pictures I took
at <http://babelmonkeys.de/~florob/E-MU%200404/>.
Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just
add this to the list of supported cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using Word Clock on RME MADI cards, AutoSync mode was alternating
betweeen MADI and WC due to a typo: AutoSync is indicated in the second
status register (status2), not the first one (status).
While the proc output was always correct, the reported WC frequency to
ALSA was unstable as mentioned in
http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the MIDI part, we need to acquire (and release) the hmidi->lock,
access to the global hdspm structure is serialized through
hmidi->hdspm->lock instead.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The name argument of request_irq() appears in /proc/interrupts, and
it's quite ugly when the name entry contains a space or special letters.
In general, it's simpler and more readable when the module name appears
there, so let's replace all entries with KBUILD_MODNAME.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/761171
The original reporter needs the model=auto quirk for his internal
speakers to be audible in the latest daily snapshot, so add an entry in
the quirk table for his PCI SSID.
A trivially different version of this patch using the model=asus quirk
should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use
the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much
improved.
Reported-and-tested-by: tomdeering7
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The convention for pci_driver.name entry in kernel drivers seem to be
the module name or equivalent ones. But, so far, almost all PCI sound
drivers use more verbose name like "ABC Xyz (12)", and these are fairly
confusing when appearing as a file name.
This patch converts the all pci_driver.name entries in sound/pci/* to
use KBUILD_MODNAME for more unified appearance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up snd_printk() helper using the %pV prefix for recursive printks.
This also automagically fixes an Oops with RO/NX-enabled modules.
Tested-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Reatlek model quirks use master_mute bool switch for controlling
the master-mute of outputs. For these cases, the initialization of HP
pins/amps were forgotten during the transition to the common automute
helper function in 3.0 development time, and resulted in the muted HP
output as default.
This patch fixes the issue by adjusting the HP output explicitly with
master_mute switch.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tag number was forgotten to be fixed after cleaning up the model
quirks for ALC262 fujitsu and lenovo-3000 models.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC register was once defined as 0x34-37 in the old Intel datasheet,
but corrected later to 0x38-3b. For fixing the register usage, a new
bit-flag is introduced for indicating the old ICH SSYNC register, and
ICH* PCI entries are added explicitly to enable this quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several fixes as well where the +1 was missing.
Done via coccinelle scripts like:
@@
struct resource *ptr;
@@
- ptr->end - ptr->start + 1
+ resource_size(ptr)
and some grep and typing.
Mostly uncompiled, no cross-compilers.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Some HP laptops with AD1981 have SPDIF connections, but currently the
driver disables it statically. Better to check the pin default config
to judge whether to enable or disable the SPDIF.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If DMA active status should be checked, I2SCON register should be referenced.
In this patch, Fix the incorrect referencing of I2SCON register.
Reported-by : Lakkyung Jung <lakkyung.jung@samsung.com>
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Make sure we follow naming convention for all PCM ops.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM support (AKA DSP support).
There will be future patches that add support to allow PCMs to be dynamically
routed to multiple DAIs at startup and also during stream runtime. This patch
moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will
in simplify the ASoC core features into distinct files.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a
DAPM context not matching the widgets context. This can lead to a wrong
prefix_len calculation, which will result in undefined behaviour. To avoid
this always use the DAPM context from the widget itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Device tree integer properties are encoded in big-endian format, but some of
the Freescale ASoC drivers were assuming that the host is in big-endian format
as well. Although this is true, it's better to use endian-safe accessors.
Also add a check for a failed ioremap() call in the SSI driver.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs
for playback and capture, so DMA buffers should be allocated only for the
initialized streams. Instead of checking for the number of active channels,
which apparently is not reliable, check to see if the actual stream object
exists.
Also provide a better name for the DMA interrupt.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have the EP93xx DMA engine driver in place, we convert the ASoC
drivers (I2S, AC97 and PCM) to take advantage of this new API. There are no
functional changes.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Don't require an audio rate SYSCLK in hw_params() in order to better
support microphone detection use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We really should be getting the interrupt - if we don't get one it's very
likely that the configuration is incorrect and audio will fail. Also
increase the timeout substantially in this case for safety.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The chip can actually support SPI so we shouldn't assume we've got an I2C
device even though that's the most common configuration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently the rbtree code will write out the entire register map when
doing a cache sync which is wasteful and will slow things down. Check
to see if the value we're about to write is the default and don't bother
restoring it if it is, either the value will have been retained or the
device will have been reset and holds the value already.
We should really store the defaults in the nodes but this resolves the
immediate issue.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of checking the azx_dev index with a fixed number (4), check
the stream direction of the assigned substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When reading from the position-buffer results in -1, handle as it's
invalid and falls back to LPIB mode as well as 0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Commit f97d0c6d5f ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
removing unnecessary if(ret) checks
This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.
Signed-off-by: Greg Dietsche <Gregory.Dietsche@cuw.edu>
Acked-by: Jonathan Cameron <jic23@cam.ac.uk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/792712
The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.
Reported-and-tested-by: rodni hipp
Cc: <stable@kernel.org> [2.6.38+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AD1836 has a PGA for its second ADC. This patch adds a control for
adjusting the the gain of the PGA.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_type field is never used, so it can be removed. The
control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver. The main difference between those devices
is the number of DACs and ADCs.
This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.
The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This allows the card driver to use the bias level variable more easily in
multi component systems.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It's redundant now thanks to the use of the generic trace infrastructure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so
If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Avoids issues if someone does a read followed by restore and doesn't mask
out only the bits being updated.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the FLL locks on the WM8915 an interrupt is generated. For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up. Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The general concept of this change is to create a PCM device for each
pin widget instead of each converter widget. Whenever a PCM is opened,
a converter is dynamically selected to drive that pin based on those
available for muxing into the pin.
The one thing this model doesn't support is a single PCM/converter
sending audio to multiple pin widgets at once.
Note that this means that a struct hda_pcm_stream's nid variable is
set to 0 except between a stream's open and cleanup calls. The dynamic
de-assignment of converters to PCMs occurs within cleanup, not close,
in order for it to co-incide with when controller stream IDs are
cleaned up from converters.
While the PCM for a pin is not open, the pin is disabled (its widget
control's PIN_OUT bit is cleared) so that if the currently routed
converter is used to drive a different PCM/pin, that audio does not
leak out over a disabled pin.
We use the recently added SPDIF virtualization feature in order to
create SPDIF controls for each pin widget instead of each converter
widget, so that state is specific to a PCM.
In order to support this, a number of more mechanical changes are made:
* s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it
clear exactly what the code is dealing with.
* We now have per_pin and per_cvt arrays in hdmi_spec to store relevant
data. In particular, we store a converter's capabilities in the per_cvt
entry, rather than relying on a combination of codec_pcm_pars and
the struct hda_pcm_stream.
* ELD-related workarounds were removed from hdmi_channel_allocation
into hdmi_instrinsic in order to simplifiy infoframe calculations and
remove HW dependencies.
* Various functions only apply to a single pin, since there is now
only 1 pin per PCM. For example, hdmi_setup_infoframe,
hdmi_setup_stream.
* hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing
and data retrieval, rather than determining which pins/converters
are to be used for creating PCMs.
This is quite a large change; it may be appropriate to simply read the
result of the patch rather than the diffs. Some small parts of the change
might be separable into different patches, but I think the bulk of the
change will probably always be one large patch. Hopefully the change
isn't too opaque!
This has been tested on:
* NVIDIA GeForce 400 series discrete graphics card. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM
audio to a PC monitor that supports audio.
* NVIDIA GeForce 520 discrete graphics card. This model is the new
1 codec n converters m pins m>n model. Tested stereo PCM audio to a
PC monitor that supports audio.
* NVIDIA GeForce 400 series laptop graphics chip. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM,
multi-channel PCM, and AC3 pass-through to an AV receiver.
* Intel Ibex Peak laptop. This model is the new 1 codec n converters m
pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass-
through to an AV receiver.
Note that I'm not familiar at all with AC3 pass-through. Hence, I may
not have covered all possible mechanisms that are applicable here. I do
know that my receiver definitely received AC3, not decoded PCM. I tested
with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a
WAV file that I believe has AC3 content rather than PCM.
I also tested:
* Play a stream
* Mute while playing
* Stop stream
* Play some other streams to re-assign the converter to a different
pin, PCM, set of SPDIF controls, ... hence hopefully triggering
cleanup for the original PCM.
* Unmute original stream while not playing
* Play a stream on the original pin/PCM.
This was to test SPDIF control virtualization.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change won't store an entire hda_pcm_stream just to represent
the capabilities of a codec; a custom data-structure will be used. To
ease that transition, modify hdmi_eld_update_pcm_info to expect the
hda_pcm_stream to be pre-initialized with the codec's capabilities, and
to update those capabilities in-place based on the ELD.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will significantly rework the generic implementation
in order to support codecs with a different number of pins and
converters. Isolate the more custom codec variants from this change by
duplicating the small portions of generic code they share. This
simplifies the later rework of that previously shared code, since we
don't have to consider the more custom codecs, and also prevents
support for those codecs from regressing.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF output controls apply to converter widgets. A future change
will create a PCM device per pin widget, and hence a set of SPDIF output
controls per pin widget, for certain HDMI codecs. To support this, we
need the ability to virtualize the SPDIF output controls. Specifically:
* Controls can be "unassigned" from real hardware when a converter is
not used for the PCM the control was created for.
* Control puts only write to hardware when they are assigned.
* Controls can be "assigned" to real hardware when a converter is picked
to support output for a particular PCM.
* When a converter is assigned, the hardware is updated to the cached
configuration.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the data that backs the kcontrols created by
snd_hda_create_spdif_out_ctls is stored directly in struct hda_codec. When
multiple sets of these controls are stored, they will all manipulate the
same data, causing confusion. Instead, store an array of this data, one
copy per converter, to isolate the controls.
This patch would cause a behavioural change in the case where
snd_hda_create_spdif_out_ctls was called multiple times for a single codec.
As best I can tell, this is never the case for any codec.
This will be relevant at least for some HDMI audio codecs, such as the
NVIDIA GeForce 520 and Intel Ibex Peak. A future change will modify the
driver's handling of those codecs to create multiple PCMs per codec. Note
that this issue isn't affected by whether one creates a PCM-per-converter
or PCM-per-pin; there are multiple of both within a single codec in both
of those codecs.
Note that those codecs don't currently create multiple PCMs for the codec
due to the default HW mux state of all pins being to point at the same
converter, hence there is only a single converter routed to any pin, and
hence only a single PCM.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's perfectly valid for an ELD to contain no SADs. This simply means that
only basic audio is supoprted.
In this case, we still want to limit a PCM's capabilities based on the ELD.
History:
* Originally, ELD application was limited solely by sad_count>0, which
was used to check that an ELD had been read.
* Later, eld_valid was added to the conditions to satisfy.
This change removes the original sad_count>0 check, which when squashed
with the above two changes ends up replacing if (sad_count) with
if (eld_valid).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: usb - turn off de-emphasis in s/pdif for cm6206
ALSA: asihpi: Use angle brackets for system includes
ALSA: fm801: add error handling if auto-detect fails
ALSA: hda - Check pin support EAPD in ad198x_power_eapd_write
ALSA: hda - Fix HP and Front pins of ad1988/ad1989 in ad198x_power_eapd()
ALSA: 6fire: Don't leak firmware in error path
ASoC: Fix wm_hubs input PGA ZC bits
ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
This reverts commit ed0bd2333c.
Since we reverted the TTY API change, we should revert the ASoC update
to it too.
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.
Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update Makefile and Kconfig to build HDMI audio support for
OMAP4 SDP and Panda boards.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add machine driver for HDMI audio on OMAP4 boards. This driver is
in charge of putting together the HDMI audio codec and the CPU DAI
and register the HDMI sound card with ALSA.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Addition of the HDMI CPU DAI driver for OMAP4. This driver is in
charge of configuring DMA settings for HDMI. Also, it finds
the HDMI video device and determines if audio playback can proceed.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
We should only call ssc_free() when ssc_request() succeeds or bad
things will happen.
Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the original code if auto detect failed and tea575x_tuner == 4
then we copy bogus information to chip->tea.card. I've changed the
autodetect code to cleanup and return -ENODEV on error instead.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for
HP and Front outputs of AD1988/AD1989. These are actually same with the
ones for AD1984 & co, port-A is 0x11 and port-D 0x12.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We only need to increase the detection rate to maximum if we're monitoring
for button presses as the response times needed for user interaction there
are much lower.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revision 2 of the Speyside platform supplies a 32kHz clock on MCLK2 rather
than MCLK1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DCVDD and MICVDD are intended to be (and almost always are) generated by
on-board LDOs which are transparently controlled by the driver so we
shouldn't really be requesting them from the regulator API. If the driver
is updated to support external supply of these then we will need to change
the way we handle this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This goto is after the call to clk_get, so it should go to the label that
includes a call to clk_put.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
expression e1,e2;
statement S;
@@
e1 = clk_get@p1(...);
... when != e1 = e2
when != clk_put(e1)
when any
if (...) { ... when != clk_put(e1)
when != if (...) { ... clk_put(e1) ... }
* return@p3 ...;
} else S
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Really this should be something the IRQ core can cope with for us but since
it doesn't currently do so (at least for threaded interrupts like this) do
so in the driver. This allows us to run with interrupt controllers that
only support edge triggered interrupts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Current implementation set max98095->sysclk/max98088->sysclk to freq twice.
Set it once is enough, this patch removes the first assignment in case
we may set invalid clock frequency to max98095->sysclk/max98088->sysclk.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (45 commits)
ARM: 6945/1: Add unwinding support for division functions
ARM: kill pmd_off()
ARM: 6944/1: mm: allow ASID 0 to be allocated to tasks
ARM: 6943/1: mm: use TTBR1 instead of reserved context ID
ARM: 6942/1: mm: make TTBR1 always point to swapper_pg_dir on ARMv6/7
ARM: 6941/1: cache: ensure MVA is cacheline aligned in flush_kern_dcache_area
ARM: add sendmmsg syscall
ARM: 6863/1: allow hotplug on msm
ARM: 6832/1: mmci: support for ST-Ericsson db8500v2
ARM: 6830/1: mach-ux500: force PrimeCell revisions
ARM: 6829/1: amba: make hardcoded periphid override hardware
ARM: 6828/1: mach-ux500: delete SSP PrimeCell ID
ARM: 6827/1: mach-netx: delete hardcoded periphid
ARM: 6940/1: fiq: Briefly document driver responsibilities for suspend/resume
ARM: 6938/1: fiq: Refactor {get,set}_fiq_regs() for Thumb-2
ARM: 6914/1: sparsemem: fix highmem detection when using SPARSEMEM
ARM: 6913/1: sparsemem: allow pfn_valid to be overridden when using SPARSEMEM
at91: drop at572d940hf support
at91rm9200: introduce at91rm9200_set_type to specficy cpu package
at91: drop boot_params and PLAT_PHYS_OFFSET
...
Commit 9477c58e33 ("ALSA: hda - Reorganize controller quriks with bit
flags") changed the driver type compares into various quirk bits.
However, the check for AZX_DCAPS_NO_TCSEL got reverted: instead of
clearing TCSEL for chipsets that have that standard capability, it
cleared then when the NO_TCSEL bit was set.
This can lead to noise and repeated sounds - a weird "echo" behavior.
As the comment just above says: "Ensuring these bits are 0 clears
playback static on some HD Audio codecs". Which is definitely true at
least on my Core i5 Westmere system.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
We want the default state of the HP_MUTE signal to be asserted, so that
the headphones are muted before the first audio playback. Without this,
the headphones are left unmuted until shortly after the first audio
playback completes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Run the data through cpu_to_be16() so it's at least clear what we're up to.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Commit af46800 ("ASoC: Implement mux control sharing") introduced
function dapm_is_shared_kcontrol.
When this function returns true, the naming of DAPM controls is derived
from the kcontrol_new. Otherwise, the name comes from the widget (and
possibly a widget's naming prefix).
A bug in the implementation of dapm_is_shared_kcontrol made it return 1
in all cases. Hence, that commit caused a change in control naming for
all controls instead of just shared controls.
Specifically, a control is always considered shared because it is always
compared against itself. Solve this by never comparing against the widget
containing the control being created.
Equally, controls should never be shared between DAPM contexts; when the
same codec is instantiated multiple times, the same kcontrol_new will be
used. However, the control should no be shared between the multiple
instances.
I tested that with the Tegra WM8903 driver:
* Shared is now mostly 0 as expected, and sometimes 1.
* The expected controls are still generated after this change.
However, I don't have any systems that have a widget/control naming
prefix, so I can't test that aspect.
Thanks for Jarkko Nikula for pointing out how to fix this.
Reported-by: Liam Girdwood <lrg@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6: (57 commits)
regulator: Fix 88pm8607.c printk format warning
input: Add support for Qualcomm PMIC8XXX power key
input: Add Qualcomm pm8xxx keypad controller driver
mfd: Add omap-usbhs runtime PM support
mfd: Fix ASIC3 SD Host Controller Configuration size
mfd: Fix omap_usbhs_alloc_children error handling
mfd: Fix omap usbhs crash when rmmoding ehci or ohci
mfd: Add ASIC3 LED support
leds: Add ASIC3 LED support
mfd: Update twl4030-code maintainer e-mail address
mfd: Correct the name and bitmask for ab8500-gpadc BTempPullUp
mfd: Add manual ab8500-gpadc batt temp activation for AB8500 3.0
mfd: Provide ab8500-core enumerators for chip cuts
mfd: Check twl4030-power remove script error condition after i2cwrite
mfd: Fix twl6030 irq definitions
mfd: Add phoenix lite (twl6025) support to twl6030
mfd: Avoid to use constraint name in 88pm860x regulator driver
mfd: Remove checking on max8925 regulator[0]
mfd: Remove unused parameter from 88pm860x API
mfd: Avoid to allocate 88pm860x static platform data
...
* 'omap-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap-2.6: (33 commits)
OMAP3: PM: Boot message is not an error, and not helpful, remove it
OMAP3: cpuidle: change the power domains modes determination logic
OMAP3: cpuidle: code rework for improved readability
OMAP3: cpuidle: re-organize the C-states data
OMAP3: clean-up mach specific cpuidle data structures
OMAP3 cpuidle: remove useless SDP specific timings
usb: otg: OMAP4430: Powerdown the internal PHY when USB is disabled
usb: otg: OMAP4430: Fixing the omap4430_phy_init function
usb: musb: am35x: fix compile error when building am35x
usb: musb: OMAP4430: Power down the PHY during board init
omap: drop board-igep0030.c
omap: igep0020: add support for IGEP3
omap: igep0020: minor refactoring
omap: igep0020: name refactoring for future merge with IGEP3
omap: Remove support for omap2evm
arm: omap2plus: GPIO cleanup
omap: musb: introduce default board config
omap: move detection of NAND CS to common-board-devices
omap: use common initialization for PMIC i2c bus
omap: consolidate touch screen initialization among different boards
...
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Matti Aaltonen <matti.j.aaltonen@nokia.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Commit 52ba67b ("ASoC: Force all DAPM contexts into the same bias state")
powers up all the DAPM contexts in a card if any DAPM context becomes
active. Unfortunately power down newer happens if per-card DAPM context
doesn't have any widgets.
Reason for this is that power state of per-card DAPM context without
widgets is never cleared and thus all the DAPM contexts remain permanently
active. Test for widgetless calling DAPM context in dapm_power_widgets()
doesn't work for per-card DAPM context since power change is never
originating from widgetless per-card DAPM context.
Fix this by pre-clearing power state flag of non-codec DAPM context at the
beginning of power sequence.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's enough to include linux/delay.h just once in
sound/soc/codecs/wm8915.c, so remove the duplicate.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
at91sam9g20 is providing master clock to wm8731: not using a crystal but an
external MCLK. We can avoid conflict and save power using WM8731_SYSCLK_MCLK as
we do not need oscillator to be powered.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The crystal oscillator is only enabled if the WM8731_SYSCLK_XTAL master clock
is specified. Fix the connected() struct snd_soc_dapm_route function to take
this into account. Oscillator is not enabled on machine that need it otherwise.
Machine drivers have to make sure that they use the proper SYSCLK value.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We do not have to free a resource that is not allocated yet.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For cards that have two or more DAIs, snd_soc_resume's loop over all
DAIs ends up calling schedule_work(deferred_resume_work) once per DAI.
Since this is the same work item each time, the 2nd and subsequent
calls return 0 (work item already queued), and trigger the dev_err
message below stating that a work item may have been lost.
Solve this by adjusting the loop to simply calculate whether to run the
resume work immediately or defer it, and then call schedule work (or not)
one time based on that.
Note: This has not been tested in mainline, but only in chromeos-2.6.38;
mainline doesn't support suspend/resume on Tegra, nor does the mainline
Tegra ASoC driver contain multiple DAIs. It has been compile-checked in
mainline.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce bit-flags indicating the necessary controller quirks, and
set them in pci driver_data field. This simplifies the checks in the
driver code and avoids the pci-id lookup in different places.
Also, this patch adds the PCI ID entry for AMD Hudson. AMD Hudson
requires a similar workaround like ATI SB while other generic ATI and
AMD controllers don't need but some ATI-HDMI quirks. So, we need a
different entry for Hudson.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the wrong usage of snd_printdd() for debug prints of input
entries. It should be snd_printd() like others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced NVIDIA GeForce GT 520 has 4 pins within a single
codec. Bump MAX_HDMI_PINS to accomodate this. Also bump MAX_HDMI_CVTS
to match it; this might be needed later too.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the PCM period size is set larger than 10 seconds, currently the
PCM core may abort the operation with DMA-error due to the fixed timeout
for 10 seconds. A similar problem is seen in the drain operation that
has a fixed timeout of 10 seconds, too.
This patch fixes the timeout length depending on the period size and
rate, also including the consideration of no_period_wakeup flag.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The comment does not reflect reality anymore since the multi-component
monster patch landed. Things are matched by names now, and not by
exporting and referencing a struct. Fix it to avoid confusion.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the previous commit 'ASoC: davinci-pcm: convert to BATCH mode', the phase
offset of 2 was mentioned in the commit message but not well commented in the
source.
Add descriptive comments of the phase offset with and without ping-pong
buffers enabled.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The davinci-pcm driver's snd_pcm_ops pointer function currently calls into
the edma controller driver to read the current positions of the edma channels
to determine pos to return to the ALSA framework. In particular,
davinci_pcm_pointer() calls edma_get_position() and the latter has a comment
indicating that "Its channel should not be active when this is called" whereas
the channel is surely active when snd_pcm_ops.pointer is called.
The operation of davinci-pcm in capture and playback appears to follow close
the other pcm drivers who export SNDRV_PCM_INFO_BATCH except that davinci-pcm
does not report it's positions from pointer() using the last transferred
chunk. Instead it peeks directly into the edma controller to determine the
current position as discussed above.
Convert the davinci-pcm driver to BATCH mode: count the periods elapsed in the
prtd->period member and use its value to report the 'pos' to the alsa
framework in the davinci_pcm_pointer function.
There is a phase offset of 2 periods between the position used by dma setup
and the position reported in the pointer function. Either +2 in the dma
setup or -2 in the pointer function (with wrapping, both) accounts for this
offset -- I opted for the latter since it makes the first-time setup clearer.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extract functions that modify the prtd->period member in preparation for
conversion to BATCH mode playback.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The release of the dma channels was being performed in prepare and there was a
edma_resume call for the asp-channel only being executed on START, RESUME and
PAUSE_RELEASE.
The mcasp on da850evm with ping-pong buffers enabled was exhibiting an audible
glitch on every playback after the first. It was determined through trial and
error that the following two changes fix this problem:
1) Move the edma_start calls from prepare to trigger and 2) reverse the order
of starting the asp and ram channels.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the registration of davinci-mcasp.1 in the davinci-evm platform
setup for da830 and dm6467, davinci-pcm can handle more than the currently
reported maximum channels of 2.
Increase the maximum channels to 384 to match the maximum reported by
davinci-mcasp.1.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the data_type test in ping_pong_dma_setup, davinci-pcm is capable of
handling data of width up to and including 32bits.
"
if ((data_type == 0) || (data_type > 4)) {
printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
return -EINVAL;
}
"
Update the .format member of the snd_pcm_hardware instances it registers to
reflect this capability.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The setup of the pong channel uses EDMA_CHAN_SLOT instead of & 0x3f as the
setup of the ping channel does.
Make the setup of ping and pong symmetric. There is no functional change
introduced by this patch.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make use of the freshly introduced methods to re-use standard mixer
handling and add some controls that are hidden but implemented in a
standard conform way on M-Audio's FastTrack devices.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Original-code-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk type will let the driver assume that there is a standard
mixer on a given interface, or that a specific mixer quirks will handle
the device.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to allow quirks functions to hook up to the standard feature
unit op tables, this patch exports a pointer to the struct that is used
internally.
That way, all the code handling the control can be kept private, and
external code can reference the symbol to re-use it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames add_control_to_empty() to snd_usb_mixer_add_control()
and exports it, so the quirks functions can make use of it.
Also, as "struct mixer_build" is private to mixer.c, rewrite the
function to take an argument of type "struct usb_mixer_interface"
instead.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This change unifies the initial handling of a pin's state with the code to
update a pin's state after a hotplug (unsolicited response) event. The
initial probing, and all updates, are now routed through hdmi_present_sense.
The stored PD and ELDV status is now always derived from GetPinSense verb
execution, and not from the data in the unsolicited response. This means:
a) The WAR for NVIDIA codec's UR.PD values ("old_pin_detect") can be
removed, since this only affected the no-longer-used unsolicited
response payload.
b) In turn, this means that most NVIDIA codecs can simply use
patch_generic_hdmi instead of having a custom variant just to set
old_pin_detect.
c) When PD && ELDV becomes true, no extra verbs are executed, because the
GetPinSense that was previously executed by snd_hdmi_get_eld (really,
hdmi_eld_valid) has simply moved into hdmi_present_sense.
d) When PD && ELDV becomes false, there is a single extra GetPinSense verb
executed for codecs where old_pin_detect wasn't set, i.e. some NVIDIA,
and all ATI/AMD and Intel codecs. I doubt this will be a performance
issue.
The new unified code in hdmi_present_sense also ensures that eld->eld_valid
is not set unless eld->monitor_present is also set. This protects against
potential invalid combinations of PD and ELDV received from HW, and
transitively from a graphics driver.
Also, print the derived PD/ELDV bits from hdmi_present_sense so the kernel
log always displays the actual state stored, which will differ from the
values in the unsolicited response for NVIDIA HW where old_pin_detect was
previously set.
Finally, a couple of small tweaks originally by Takashi:
* Clear the ELD content to zero before reading it, so that if it's not
read (i.e. when !(PD && ELDV)) it's in a known state.
* Don't show ELD fields in /proc ELD files when the ELD isn't valid.
The only possibility I can see for regression here is a codec where the
GetPinSense verb returns incorrect data. However, we're already exposed
to that, since that data is used (a) from hdmi_add_pin to set up the
initial pin state, and (b) within snd_hda_input_jack_report to query
a pin's presence value. As such, I don't believe any HW has bugs here.
Includes-changes-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The microphone input on the back panel (pink connector)
stopped operating correctly after an upgrade from
2.6.35 to 2.6.38; the actual problem manifests itself
as a lack of microphone bias voltage (VREF_HIZ) on
node 0x17.
With AD1988_6STACK_DIG the maximum bias voltage (VREF_80)
is applied and the headset operates correctly.
Signed-off-by: Tony Vroon <tony@linx.net>
Tested-by: Doug Redlich <pbrigade@nxltech.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
pcmcia: Make struct pcmcia_device_id const, sound drivers edition
staging: pcmcia: Convert pcmcia_device_id declarations to const
pcmcia: Convert pcmcia_device_id declarations to const
pcmcia: Make declaration and uses of struct pcmcia_device_id const
pcmcia/sa1100: put sa11x0_pcmcia_hw_init[] to .devinit.data
Currently CODEC and platform drivers have their module reference count
incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference
count incremented in soc_bind_dai_link().
CPU DAIs should have their reference count incremented in soc_probe_dai_link()
just like the CODEC and platform drivers.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support)
broke support for Raumfeld platforms as it didn't take into account the
different hardware features on individual devices.
In particular, Raumfeld speakers have no S/PDIF output, so the members
of the snd_soc_card struct must be set dynamically.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch is preparation of cleanup suspend/resume patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was using saved_xxx variable for suspend/resume.
OTOH, the start and stop of power/clock are controlled by
fsi_hw_startup/fsi_hw_shutdown in current FSI driver.
The all necessary registers value are set by fsi_hw_startup.
So, if fsi_hw_shutdown is called when "suspend" is generated,
and fsi_hw_startup is called at "resume",
the saved_xxx are not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSIA/B ports is enabled by default when power-on,
and current FSI is supporting RuntimePM.
In addition, current fsi_module_init/kill doesn't care
simultaneous playback/recorde.
This mean FSI port control is not needed.
This patch remove fsi_module_init/kill
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
