devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver had the property check for 'cirrus,amutec_eq_bmutec' the
wrong way around. That happens if you misspell the property in the
bindings during tests.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
enabled. Unfortunately the DMA header patch had to be redone
to avoid adding new multiplatform specific include paths, the
other patches are just trivial compile fixes.
Note that this does not yet contain the necessary Kconfig
changes as we are still waiting for some drivers to get
fixed up first.
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Merge tag 'tags/omap-for-v3.8/cleanup-multiplatform-no-clock-signed' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap into next/pm2
From Tony Lindgren:
Remaining patches to allow omap2+ to build with multiplatform
enabled. Unfortunately the DMA header patch had to be redone
to avoid adding new multiplatform specific include paths, the
other patches are just trivial compile fixes.
Note that this does not yet contain the necessary Kconfig
changes as we are still waiting for some drivers to get
fixed up first.
* tag 'tags/omap-for-v3.8/cleanup-multiplatform-no-clock-signed' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap:
ARM: OMAP: Move plat-omap/dma-omap.h to include/linux/omap-dma.h
ASoC: OMAP: mcbsp fixes for enabling ARM multiplatform support
watchdog: OMAP: fixup for ARM multiplatform support
Conflicts due to surrounding changes in:
arch/arm/mach-omap2/omap_hwmod_2420_data.c
arch/arm/mach-omap2/omap_hwmod_2430_data.c
Signed-off-by: Olof Johansson <olof@lixom.net>
The same type of code is being used in multiple places in various
codec drivers, so put it as a core library.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We cannot include any plat or mach headers for the multiplatform
support.
Fix the issue by defining local mcbsp_omap1().
cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
cc: Liam Girdwood <lrg@ti.com>
cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
cc: Jaroslav Kysela <perex@perex.cz>
cc: Takashi Iwai <tiwai@suse.de>
cc: alsa-devel@alsa-project.org
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
A lot of headsets/headphones have a "Speaker" mixer control. This confuses
PulseAudio to think it is a speaker instead of a headphone/headset.
Therfore, we rename it to "Headphone".
We determine if something is a headphone similar to how udev determines
form factor (see 78-sound-card.rules).
BugLink: https://bugs.launchpad.net/bugs/1082357
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The playback endpoint uses implicit feedback mode, similar
to the M-Audio FTU. Like with the FTU, we need to associate
the sync pipe ourselves.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ranges for various Fast Track C400 controls, as observed
while using the vendor's mixer control software (res values
are an estimation).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds the unit ID and the control as parameters to the creation of the
effect unit control for the M-Audio Fast Track Ultra. This allows the
code to be shared with other devices that use different unit ID and
control, such as the M-Audio Fast Track C400.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code mishandles the case where the device is a UAC2
and the bDescriptorSubtype is a UAC2 Effect Unit (0x07).
It tries to parse it as a Processing Unit (which is similar to two
other UAC1 units with overlapping subtypes), but since the structure
is different (See: 4.7.2.10, 4.7.2.11 in UAC2 standard), the parsing
is done incorrectly and prevents the device from initializing.
For now, just ignore the unit.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, channel IDs exceeding 31 (0x1f) cannot be used.
The channel ID is derived from the cmask. Extending cmask
to a 64-bit type would only allow it to go up to 63 (0x3f).
Some devices have channel IDs exceeding that as well.
To address that, add an offset to the mixer element which
is then accounted for in the UAC set/get functions.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For implicit feedback endpoints, the number of bytes for each packet
is matched by the corresponding synchronizing endpoint.
The size is calculated by taking the actual size and dividing it by
the stride - currently by the endpoint's stride, but we should use the
synchronization source's stride.
This is evident when the number of channels differ between the
synchronization source and the implicitly fed-back endpoint, as with
M-Audio Fast Track C400 - the synchronization source (capture)
has 4 channels, while the implicit feedback mode endpoint has 6.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed unnecessary labels probably left out of some previous cleanup.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As the I2S src clks are registered with clkdev using generic
connection id, driver can get the clk name using generic id.
So the variable representing the array of rclk src clks is
deleted.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the jack detection inforamtion as bool from devicetree.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regmap_init_i2c() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some ADSP devices can make use of DVFS to optimise power consumption
depending on the operating frequency of the DSP core. Implement
support for this in the generic ADSP code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a haptics widget for use by the haptics driver and expose the DAPM
context for it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a haptics widget for use by the haptics driver and expose the DAPM
context for it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a preliminary patch for introducing a protection to access
races of snd_array instances. Call snd_array_init() appropriately
at the initialization time and don't call it twice.
Also the allocations of codec-spec structs are cleaned up by helper
functions in patch_sigmatel.c and patch_analog.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We cannot include any plat or mach headers for the multiplatform
support.
Fix the issue by defining local mcbsp_omap1().
Signed-off-by: Tony Lindgren <tony@atomide.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Hi Mark,
thanks your insisting on a better description for the patch, I found a
more appropriate solution for the problem:
Compiling the SoC Audio driver for Freescale i.MX as a module
(CONFIG_SND_SOC_IMX_PCM=m) results in a non-functional sound driver
indicated by the error message:
| imx-sgtl5000 sound.1: platform imx-pcm-audio not registered
| imx-sgtl5000 sound.1: snd_soc_register_card failed (-517)
| platform sound.1: Driver imx-sgtl5000 requests probe deferral
instead of the message:
| imx-sgtl5000 sound.1: sgtl5000 <-> 63fcc000.ssi mapping ok
that is to be expected upon loading the snd-soc-imx-pcm.ko module.
The build log reveals, that the file imx-pcm-dma.o (or imx-pcm-fiq.o
depending on the kernel configuration), which should be linked
together with imx-pcm.o into snd-imx-pcm.ko, is not being compiled in
this case.
The make rules for these files shows that the target object imx-pcm.o
is assigned to the variable snd-soc-imx-pcm-y while
imx-pcm-{dma,fiq}.o are added to to
snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) and
snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) which resolve to
snd-soc-imx-pcm-m in this case.
According to Documentation/kbuild/modules.txt:
|When the module is built from multiple sources, an additional line is
|needed listing the files:
|
| <module_name>-y := <src1>.o <src2>.o ...
Thus the type of the config variables CONFIG_SND_SOC_IMX_PCM_DMA and
CONFIG_SND_SOC_IMX_PCM_FIQ should be 'bool' instead of 'tristate' to
resolve to 'y' when selected.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to a broken make rule, sound/soc/fsl/imx-pcm-dma.c or
sound/soc/fsl/imx-pcm-fiq.c (whatever is selected via Kconfig) will
not be compiled into imx-pcm.o when building as module, i.e.:
CONFIG_SND_SOC_IMX_PCM=m
CONFIG_SND_SOC_IMX_PCM_DMA=m
resulting in a non-functional sound driver.
This gives the error messages:
| imx-sgtl5000 sound.1: platform imx-pcm-audio not registered
| imx-sgtl5000 sound.1: snd_soc_register_card failed (-517)
| platform sound.1: Driver imx-sgtl5000 requests probe deferral
when loading the driver instead of what's to be expected:
| imx-sgtl5000 sound.1: sgtl5000 <-> 63fcc000.ssi mapping ok
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The device should not be generating interrupts when it does not have power
so ignore incoming interrupts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't rely on the kcontrol for robustness reasons, the widget mechanism
is what the framework uses.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the support for channel maps of the PCM streams on USB audio
devices. The channel map information is already found in
ChannelConfig descriptor entries, which haven't been referred until
now.
Each chmap entry is added to audioformat list entry and copied to TLV
dynamically instead of creating a whole chmap array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'v3.7-rc7' into next/cleanup
Merging in mainline back to next/cleanup since it has collected a few
conflicts between fixes going upstream and some of the cleanup patches.
Git doesn't auto-resolve some of them, and they're mostly noise so let's
take care of it locally.
Conflicts are in:
arch/arm/mach-omap2/omap_hwmod_44xx_data.c
arch/arm/plat-omap/i2c.c
drivers/video/omap2/dss/dss.c
Signed-off-by: Olof Johansson <olof@lixom.net>
I forgot this again... codec->in_pm is in #ifdef CONFIG_PM
Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is better style and facilitates implementation of device tree support
for the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This device doesn't have a pdata definition for legacy boards, and
unless anyone need to control the reset GPIO, it's not worth adding one.
So this feature is only available to DT users for now.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When a playback stream is paused, the stream isn't actually stopped,
thus we still need to take care of the in-flight data amount for the
delay calculation. Otherwise the value of subs->last_delay is no
longer reliable and can give a bogus value after resuming from pause.
This will result in "delay: estimated XX, actual YY" error messages.
Also, during pause after all in flight data are processed
(i.e. last_delay = 0), we don't have to calculate the actual delay
from the current frame. Give a short path in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
set the 'onwer' field of the registered snd_soc_card object to prevent
removal of the module when its resources are in use.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
also set MODULE_AUTHOR and MODULE_DESCRIPTION
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It doesn't make sense to calculate the delay for capture streams in
the current implementation. It's always zero, so we should skip the
computation in snd_usb_pcm_pointer() in the case of capture.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes following warning.
sound/soc/codecs/wm8753.c:1594:1-6: WARNING: invalid free of devm_ allocated data
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes following warning.
sound/soc/codecs/wm8510.c:614:1-6: WARNING: invalid free of devm_ allocated data
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM is a pseudo-device. It doesn't have any of it's own registers
or hardware. It rather acts as a layer of abstraction for DMA
transfers. Hence, instead of classifying it as a device in its own
right, we call the initialisation from the MSP driver.
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Ola LILJA2 <ola.o.lilja@stericsson.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The info record at the start of the dsp firmware file has been
expanded to incorporate additional version information. We need
to check the version to make sure we understand the layout of
the information in the record. The srec2image tool is currently
used to create this record during creation of the .dfw file.
Signed-off-by: Scott Ling <sl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The internal mic on MBP81 gives only the right channel, and the left
channel is static. Add a verb to fix the ADC2 channel mode to expand
mono right to stereo.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50781
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to apply the fixup verbs in cs_init().
But adding the fixup verbs will break mbp101 fixup that has been fixed
recently again, since the mbp101 fixup contains the wrong verbs to
override. So these bogus verbs must be removed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio
driver which causes the code to not initialize the sync endpoint from
configure_endpoint().
Reported-by: Jeffrey Barish <jeff_barish@earthlink.net>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the recent firmware code tries to reread and cache the firmware by
itself, it's even better to keep the struct firmware data instead of
keeping a local copy. Also, it makes little sense to disable the fw
loader for this driver, so added the explicit dependency, too.
Last, but not least, allocate the firmware data loaded via ioctl in
vmalloc'ed buffer instead, as the firmware size isn't that small.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new firmware code tries to re-read the formerly read firmware
files before suspend. Thus it's wiser to keep the "patch" firmware in
the driver for avoiding this unnecessary re-reading.
Of course, this will consume a bit of memory for unused stuff, but
the patch fw is supposed to be fairly small, so it's more benefit in
the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet again like previous two commits, drop the old hwdep user-space
firmware code from vx driver (snd-vxpocket and snd-vx222).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes no longer sense to keep the old hwdep user-space firmware
loading, which has been deprecated since ages ago.
Just add a hard dependency on CONFIG_FW_LOADER and drop the useless
code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we keep the pin default config values anyway internally, we
don't have to set the values in the codec. This patch removes the
code writing the pincfg values.
As a gratis bonus, we can remove also the code restoring the original
pincfg values at PM resume or module free. This will give us more
benefit, as it can reduce the unnecessary power-up of codecs.
This won't change the driver functionality. The only difference would
be that the codec proc file will show the original pincfg values
instead of the actually referred values. The actually referred values
can be determined from sysfs *_pin_configs files.
(Also hda-emu was updated to follow this change.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When lowering SYSCLK to 50kHz for accessory detection also lower the
AIFnCLK divisor to normalise the clocking configuration within the
device. This will not disrupt audio as we cannot support active audio
with such a low SYSCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The free callback is called at the state where no extra verbs are
executed, thus calling *_shutup() is useless, as it's checking the
shutdown flag. Remove such superfluous calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase timeout to be more reliable and avoid the chance of
missing interrupts during boot.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PCM hw_free and close should wait until all the pending stop
operations have been finished. Basically only PCM trigger callback
should use non-wait calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.
So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop(). (Actually there is only one
place calling this, so it was safe to change.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For further code simplification, drop the conditional call for
usb_kill_urb() with can_wait argument in deactivate_urbs(), and use
only usb_unlink_urb() and wait_clear_urbs() pairs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop(). Also replaced from int to bool.
No functional changes by this commit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The async unlink behavior has been working over years. The option was
provided only as a workaround for 2.4.x kernel. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'rt' was dereferenced before the NULL check.
Moved the code after the check.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return the value obtained from snd_pcm_hw_constraint_minmax() instead
of -EINVAL. Silences the following smatch warning:
sound/core/pcm_native.c:2003 snd_pcm_hw_constraints_complete() info:
why not propagate 'err' from snd_pcm_hw_constraint_minmax() instead of -22?
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If this array is not cleared, the jack related code later might
fail to create "Internal Speaker Phantom Jack" on Dell Inspiron 3420 and
Dell Vostro 2420.
BugLink: https://bugs.launchpad.net/bugs/1076840
Cc: stable@vger.kernel.org (3.6+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Various cleanups and enhancements are made to core Tegra code towards the
aim of including Tegra in a multi-platform ARM kernel:
RTC, timer, and TWD are configured via device tree.
SPARSE_IRQ is enabled.
Tegra's debug_ll options are simplified, and the macros brought into
line with other multi-platform implementations, and moved to the new
common location.
Two headers still need to be eliminated in order to include Tegra in a
multi-platform kernel/ <mach/{clk,powergate}.h>. A new common API needs
to be invented to replace parts of clk.h. powergate.h might be replaced
by regulators; this needs more investigation.
This pull request is based on tegra-for-3.8-dt, followed by a merge of
arm-soc's devel/debug_ll_init branch.
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Merge tag 'tegra-for-3.8-single-zimage' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra into next/multiplatform
From Stephen Warren:
ARM: tegra: single-zImage preparation work
Various cleanups and enhancements are made to core Tegra code towards the
aim of including Tegra in a multi-platform ARM kernel:
RTC, timer, and TWD are configured via device tree.
SPARSE_IRQ is enabled.
Tegra's debug_ll options are simplified, and the macros brought into
line with other multi-platform implementations, and moved to the new
common location.
Two headers still need to be eliminated in order to include Tegra in a
multi-platform kernel/ <mach/{clk,powergate}.h>. A new common API needs
to be invented to replace parts of clk.h. powergate.h might be replaced
by regulators; this needs more investigation.
This pull request is based on tegra-for-3.8-dt, followed by a merge of
arm-soc's devel/debug_ll_init branch.
* tag 'tegra-for-3.8-single-zimage' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra: (58 commits)
ARM: tegra: move debug-macro.S to include/debug
ARM: tegra: don't include iomap.h from debug-macro.S
ARM: tegra: decouple uncompress.h and debug-macro.S
ARM: tegra: simplify DEBUG_LL UART selection options
ARM: tegra: select SPARSE_IRQ
ARM: tegra: enhance timer.c to get IO address from device tree
ARM: tegra: enhance timer.c to get IRQ info from device tree
ARM: timer: fix checkpatch warnings
ARM: tegra: add TWD to device tree
ARM: tegra: define DT bindings for and instantiate RTC
ARM: tegra: define DT bindings for and instantiate timer
ARM: tegra: whistler: enable HDMI port
ARM: tegra: tec: Enable HDMI output
ARM: tegra: plutux: Enable HDMI output
ARM: tegra: tamonten: Add host1x support
ARM: tegra: trimslice: enable HDMI port
ARM: tegra: harmony: enable HDMI port
ARM: tegra: Add Tegra30 host1x support
ARM: tegra: Add Tegra20 host1x support
ARM: tegra: trimslice: enable SPI flash
...
Various trivial cleanup changes of the Tegra code for 3.8.
Many of the changes simply remove useless #include statements, which
enable those headers to be removed or moved later, as work towards
multi-platform zImage support.
<mach/{iram,io}map.h> are moved up to arch/arm/mach-tegra to prevent
any new code outside mach-tegra from using them.
Finally, the regulator definitions in all board device tree files are
updated to use the new simpler syntax that was agreed upon.
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Merge tag 'tegra-for-3.8-cleanup' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra into next/cleanup
From Stephen Warren:
ARM: tegra: cleanup for 3.8
Various trivial cleanup changes of the Tegra code for 3.8.
Many of the changes simply remove useless #include statements, which
enable those headers to be removed or moved later, as work towards
multi-platform zImage support.
<mach/{iram,io}map.h> are moved up to arch/arm/mach-tegra to prevent
any new code outside mach-tegra from using them.
Finally, the regulator definitions in all board device tree files are
updated to use the new simpler syntax that was agreed upon.
* tag 'tegra-for-3.8-cleanup' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra:
ARM: tegra: move irammap.h to mach-tegra
ARM: tegra: move iomap.h to mach-tegra
ARM: tegra: remove <mach/dma.h>
ARM: tegra: move tegra-ahb.h out of arch/arm/mach-tegra/
ARM: tegra: remove unnecessary includes of <mach/*.h>
iommu: tegra: remove include of <mach/iomap.h>
staging: nvec: remove include of <mach/iomap.h>
crypto: tegra: remove include of <mach/clk.h>
ARM: tegra: update *.dts for regulator-compatible deprecation
usb: phy: tegra remove include of <mach/iomap.h>
usb: host: tegra remove include of <mach/iomap.h>
We found a new codec ID 292, and that just a simple quirk would enable
sound output/input on this ALC292 chip.
BugLink: https://bugs.launchpad.net/bugs/1081466
Cc: stable@vger.kernel.org
Tested-by: Acelan Kao <acelan.kao@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The biggest batch of fixes here is the Kirkwood DMA fixes, plus a couple
of other small fixes.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
The biggest batch of fixes here is the Kirkwood DMA fixes, plus a couple
of other small fixes.
This patch adds the max98090 codec prototype driver.
It supports Headphone only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removed struct ak4642_priv which had
meaningless variable.
It is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
With SPDIF passthrough, we are not restricted to just two channels of
audio; we can support however many channels the non-audio stream can
itself support. In any case, kirkwood-dma is not involved in the
format selection. So yet rid of this restriction.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox, and cleaned up by me.
Some platforms provide an external clock which can be used to allow
other sample rates to be selected. Provide support for this.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
The kirkwood DMA hardware for ASoC does not impose any restrictions
on the sample rates available, so it's silly to impose an artificial
set in the DMA code. The restrictions come from the availble clocks
to the I2S module, which are already handled in the I2S part of the
driver.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simplify the cleanup paths in the driver by using the devm_* APIs,
ensuring that all error paths are correctly checked.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't even momentarily set the pause status when starting the channel;
if we do, we should check the busy bit to ensure that we comply with
the spec. In any case, it isn't necessary; we will not active on a
START event so there is no need to pause the DMA.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stress testing the driver with multiple start/stop events causes
kirkwood-dma to report underrun errors (which used to cause the kernel
to lock up solidly). This is because kirkwood-i2s is not respecting
the restrictions imposed on clearing the 'pause' bit. Follow what the
spec says; the busy bit must be read as being clear twice before the
pause bit can be released. This solves the underruns.
However, it has been noticed that the busy bit occasionally does not
clear itself, hence the waiting is bounded to 5ms maximum to avoid a
new reason for the kernel to lockup.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox, which is further attributed to Sebastian Hesselbrath.
Rather than masking the KIRKWOOD_DCO_SPCR_STATUS register contents
against the registers virtual address, let's actually use the bit
definition for the locked status, as required in the documentation.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ignoring the real cause of the interrupt is not a good idea; this
behaviour has been observed to bring Dove platforms to silently
lockup. Instead, on error fall through to the normal interrupt
processing.
This is especially important on Dove platforms as errors are
handled separately, and allows us to clear down the real cause of
the interrupt.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
You can not use virt_to_phys() on the address returned from
dma_alloc_coherent(); it may not be part of the kernel direct-mapped
memory. Fix this to use the DMA address instead.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seems WM_ADSP2("DSP1", 0) is added twice to the widgets list, remove
that and in place use ARIZONA_DSP_WIDGETS(DSP1, "DSP1").
We need to make sure that the DSP1 Aux widgets are provided otherwise
we'll see errors such as "Failed to add route DSP1 Aux 1 -> DSP1" etc.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CONFIG_HOTPLUG is going away as an option so __devexit_p is no longer
needed, remove it.
Also fix the indentation for the initialization of the
max98088_i2c_driver struct to make chkpatch happy.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch adds SND_SOC_DAIFMT_INV_xxx support,
and it is possible to independent from platform information pointer.
Old type SH_FSI_xxx_INV is still supported,
but it will be removed soon.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes stream mode format
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes spdif format
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is requesting sh_fsi_platform_info pointer from platform,
and it didn't allowed NULL pointer.
This patch fixes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch tidyup to use fsi pointer for FSIA/B settings
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CONFIG_HOTPLUG is going away as an option so __devinitconst is no
longer needed.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We've got a report that the runtime PM may make the codec the
unresponsive on AMD platforms. Since the feature has been tested only
on the recent Intel platforms, it's safer to limit the support to such
devices for now.
This patch adds a new DCAPS bit flag indicating the runtime PM
support, and mark it for Intel controllers.
Reported-and-tested-by: Julian Wollrath <jwollrath@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Aspire AO756 has an analog built-in mic on nid 0x1b and an
external mic on nid 0x19. The BIOS doesn't set this up.
The mic detect on this Acer Aspire netbook and Acer C7 ChromeBook is
only valid when the headphone is plugged. The detect circuit relies on
the tip detect switch being closed on the jack. Tell hda_jack to ignore
the mic sense unless the headphones are plugged.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the concept of a "gated" jack. The gated jack's pin sense
is
only valid when the "gating" jack is plugged. This requires checking
the gating jack when the gated jack changes and re-checking the gated
jack when the gating jack is plugged/unplugged.
This allows handling of devices where the mic jack detect floats when
the headphone jack is unplugged.
[Rewritten for fixing the possible snd_array reallocation, covering
the missing callback calls and jack sync operations, as well as some
code cleanups -- tiwai]
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Asynchronous" is misspelled in some comments. No code changes.
Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
"Whether" is misspelled in various comments across the tree; this
fixes them. No code changes.
Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Some comments misspell "registered"; this fixes them. No code changes.
Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
When the bus reset is performed during the suspend/resume (including
the power-saving too), it calls snd_hda_suspend() and
snd_hda_resume() again, and deadlocks eventually.
For avoiding the recursive call, add a new flag indicating that the PM
is being performed, and don't go to the bus reset mode when it's on.
Reported-and-tested-by: Julian Wollrath <jwollrath@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return the value obtained from get_coeff() instead of EINVAL.
Silences a smatch warning.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend) added
autosuspend code to all files making up the snd-usb-audio driver.
However, midi.c is part of snd-usb-lib and is also used by other
drivers, not all of which support autosuspend. Thus, calls to
usb_autopm_get_interface() could fail, and this unexpected error would
result in the MIDI output being completely unusable.
Make it work by ignoring the error that is expected with drivers that do
not support autosuspend.
Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com>
Reported-by: Devin Venable <venable.devin@gmail.com>
Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk>
Reported-by: Jannis Achstetter <jannis_achstetter@web.de>
Reported-by: Rui Nuno Capela <rncbc@rncbc.org>
Cc: Oliver Neukum <oliver@neukum.org>
Cc: 2.6.39+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
We set "s" before we have capped "speed" so it could be the wrong value.
This could lead to a divide by zero bug.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use bitmap_weight to count the total number of bits set in bitmap.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
convert at91sam9g20ek with wm8731 to device tree support
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove unneeded code with the new method of dai and pcm register
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
change the method for register dai and pcm
- let the atmel-ssc-dai no longer as a standalone platform device
- remap ssc and then register dai directly
- register pcm from dai directly
- modify the code which related with this change
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove remuxing GPIO1. Leave control of this up to the platform device.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize performance by providing a 512fs based CLKIN.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
It seems git has been getting confused by the very similar contexts
for the speaker DAIs and has been applying patches to the wrong places
causing all sorts of confusion. Fix this up by hand.
Reported-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again. There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:
A. snd_usb_audio_disconnect() ->
card.c::register_mutex ->
chip->shutdown_rwsem (write) ->
snd_card_disconnect() ->
pcm.c::register_mutex ->
pcm->open_mutex
B. snd_pcm_open() ->
pcm->open_mutex ->
snd_usb_pcm_open() ->
chip->shutdown_rwsem (read)
Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a precedence bug because | has higher precedence than ?:. This
code was cut and pasted and I fixed a similar bug a few days ago.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no mixer attached to the ASRC on the wm5110 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5110 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no mixer attached to the ASRC on the wm5102 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5102 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Asynchronous Sample Rate Converters on the wm5102/wm5110 have no
mixer attached to their input, but they do allow the input to be
selected from a number of sources via a multiplexer. Currently the
platform assumes the presence of 4 multiplexers and a mixer for each
block.
This patch adds support multiplexed single input blocks into the Arizona
platform.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails. This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():
usbaudio: unit 51: unexpected type 0x09
snd-usb-audio: probe of 1-8:1.0 failed with error -5
Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out. It would be nice to
avoid that.
While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces. This allows audio and MIDI interfaces to be
used without any obvious issues.
Interface 0 is the main one to ignore. It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this. Interface 3 is some sort of control interface that
I don't understand:
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 3
bAlternateSetting 0
bNumEndpoints 0
bInterfaceClass 1 Audio
bInterfaceSubClass 1 Control Device
bInterfaceProtocol 0
iInterface 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 1 (HEADER)
bcdADC 1.00
wTotalLength 9
bInCollection 1
baInterfaceNr( 0) 1
Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I don't think this works as intended. '|' higher precedence than ?: so
the bitwize OR "0 | (val & STR_MOST)" is a no-op.
I have re-written it to be more clear.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
In case of probe deferral, the allocated GPIO line is not freed, which
prevents it from being claimed and properly asserted in later attempts.
Fix this by using devm_gpio_request().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Michael Hirsch <hirsch@teufel.de>
Cc: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a MIDI driver for the Stanton FireWire DJ controllers.
Tested-by: Sean M. Pappalardo - D.J. Pegasus <spappalardo@mixxx.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Assume that unknown ICE1724-based cards are AC97-only that can suspend
without any additional card-specific code.
This fixes suspend on Gainward Hollywood@Home 7.1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another variant of iMac 9,1 with a different codec SSID.
Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com>
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_put_volsw_sx function fails to update second control
if first control is updated by snd_soc_update_bits_locked.
Signed-off-by: Mukund Navada <navada@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Move the firmware load and record parsing functionality out into
a separate function from the boot function.
Signed-off-by: Scott Ling <sl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A closer look shows that the name is not even used and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When two built-in speakers are found on the machine, we can suppose
it's rather a 2.1 speaker system with a bass output instead of
front/surround channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL
This patch fixes these places.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
lock = (status & (0x1<<idx)) ? 1 : 0;
sync = (status & (0x100<<idx)) ? 1 : 0;
The index is given in kcontrol->private_value:
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
HDSPM_SYNC_CHECK("AES SyncCheck", 1),
HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),
The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the return value of cs42l52_set_fmt() when clock inversion is
not allowed and also remove the useless variable ret.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
[We had been assigning to ret but then ignoring the value we assgined
-- broonie]
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e. This confuses
the auto-parser. Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i). When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[]. This confuses is_empty_dac() and trims the detected DAC
in later reference.
This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some of the PantherPoint HDMI machines we currently enable, we're seeing
trouble with unsol events, i e detecting monitor presence, especially when
on battery and after suspend/resume.
BugLink: https://bugs.launchpad.net/bugs/1075882
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From Michal Simek:
* 'arm-next' of git://git.xilinx.com/linux-xlnx:
zynq: move static peripheral mappings
zynq: remove use of CLKDEV_LOOKUP
zynq: use pl310 device tree bindings
zynq: use GIC device tree bindings
+ Linux 3.7-rc3
- Move all remaining headers out of arch/arm/plat-nomadik/include/plat
out to e.g. include/linux/platform_data
- Delete arch/arm/plat-nomadik
- Convert Nomadik and Ux500 to SPARSE_IRQ
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Merge tag 'kill-plat-sparse-irq' of git://git.kernel.org/pub/scm/linux/kernel/git/linusw/linux-pinctrl into next/multiplatform
From Linus Walleij:
This patchset will:
- Move all remaining headers out of arch/arm/plat-nomadik/include/plat
out to e.g. include/linux/platform_data
- Delete arch/arm/plat-nomadik
- Convert Nomadik and Ux500 to SPARSE_IRQ
* tag 'kill-plat-sparse-irq' of git://git.kernel.org/pub/scm/linux/kernel/git/linusw/linux-pinctrl:
ARM: plat-nomadik: convert platforms to SPARSE_IRQ
mfd/db8500-prcmu: use the irq_domain_add_simple()
mfd/ab8500-core: use irq_domain_add_simple()
ARM: plat-nomadik: move MTU, kill plat-nomadik
ARM: plat-nomadik: move DMA40 header to <linux/platform_data>
ARM: plat-nomadik: use DIV_ROUND_CLOSEST()
ARM: plat-nomadik: pass IRQ to timer driver
clk/ux500: explicitly include register header
pinctrl/nomadik: merge old pincfg header
pinctrl/nomadik: move the platform data header
ARM: plat-nomadik: move NMK_GPIO_PER_CHIP into gpio-nomadik.h
ARM: plat-nomadik: Introduce new DB8540 GPIO registers
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Remove the boot_done counter variable and check the wm0010 state
variable instead.
Signed-off-by: Scott Ling <scott.ling@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove includes of <mach/dma.h> from sound/soc; nothing from it is used.
Remove include of <mach/dma.h> from mach-tegra/apbio.c; since the DMA
transfers made by this file don't need flow-control with any peripheral,
there's no need to set any slave ID.
Once those changes are made, there are no remaining users of <mach/dma.h>
so remove it. Drivers should get this information from device tree. This
removal is necessary for single zImage.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for ASUS - Xonar DSX sound cards. Tested on
openSUSE 12.2 with kernel:
Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux
Works:
- play sounds
- adjust volume on master channel.
- mute .
Since Xonar DS uses the same chip, everything that works for DS should
work for DSX as well.
Signed-off-by: Sergiu Giurgiu <sgiurgiu11@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.
Original fix credit to Javeed Shaikh.
BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361
Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add generic ESS vendor ID to pm_whitelist. This should fix suspend on
all Maestro-2 and Maestro-2E based PCI cards.
Tested on Terratec DMX and SF64-PCE2.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.
This fixes the digital mic on the Macbook Pro 10,1/Retina.
Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This moves the DMA40 platform data header from <plat/ste_dma40.h>
to <linux/platform_data/dma-ste-dma40.h> where is belongs.
Cc: Dan Williams <djbw@fb.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Ola Lilja <ola.o.lilja@stericsson.com>
Cc: Herbert Xu <herbert@gondor.apana.org.au>
Cc: Andreas Westin <andreas.westin@stericsson.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.
Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Starting audio or seeking in various music players causes
setup_dig_out_stream() to be called, which resets the SPDIF stream,
which caused one DAC (but not another) to make a clicking noise every
time.
This patch ensures the reset only happens when it needs to, which is
when the format changes, and makes the code a little more readable.
Signed-off-by: Laurence Darby <ldarby@tuffmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Omap no longer needs this option, mach/gpio.h is
empty.
Also remove mach/irqs.h from gpio-omap.h and
include it directly from the related omap1
gpio init files.
Otherwise omap2+ build fails for MULTI_PLATFORM.
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: alsa-devel@alsa-project.org
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
Playing 24-bit format file leads to channel swap on mx28 and the reason is that
the current driver performs one write/read to/from the SAIF_DATA register to
trigger the transfer.
This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register
and thus is capable of storing the 16-bit left and right channels, but for the
S24_LE case it can only store one channel, so in order to not lose the FIFO sync
an extra read/write is needed.
Reported-by: Dan Winner <DWinner@tc-helicon.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Dan Winner <DWinner@tc-helicon.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we select the WM1250-EV1 as the current software system
configuration demands it.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver didn't care fsi_hw_start/stop() return value,
and it causes WARNING() call if SNDRV_PCM_TRIGGER_START failed.
This patch solved this issue
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In its previous status, the first capture didn't work properly;
nothing was actually recorded from the microphone. This
behaviour was observed using a Visstrim M10 board.
In order to solve this BUG a workaround has been added that,
during the initialization process of the codec, powers on and
off the ADC.
The issue seems related to a HW BUG or some behavior that
is not documented in the datasheet.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The rate isn't restored properly after resume since it's only set up
in hw_params, and not in prepare callback. For fixing it, put the
corresponding call to resume callback as well.
Reported-and-tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Evalation of the WM5102 has identified a number of register values which
should be written after SYSCLK is enabled on revision A in order to
improve performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.
Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.
The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
mixer.c and others; the device speed is now cached in subs->speed
instead of accessing to chip->dev
The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.
The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks. They'll be covered by the
upcoming change to rwsem.
Also the mixer codes are untouched, too. These will be fixed in
another patch, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races at PCM disconnection:
- while a PCM device is being opened or closed
- while the PCM state is being changed without lock in prepare,
hw_params, hw_free ops
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver is using fsi_set_master_clk() if it needs system clock.
But this function was called from
fsi_hw_shutdown()/fsi_dai_trigger()/fsi_resume() without a sense of unity.
Because of this, sound playback after suspend failed sometimes.
To keep consistency, fsi_master_clk() was called from
fsi_hw_start/stop() only now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many Arizona class devices contain ADSP2 cores with a standard method for
hooking them into the audio map. Define standard helpers for this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many current Wolfson devices feature DSPs based around an architecture
known as ADSP. Since there is a lot of commonality in the system
integration of these devices a common library will be used to provide
support for them.
This version provides equivalent support for ADSP1 to that currently
included in the WM2200 driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dev_<level> calls take less code than dev_printk(KERN_<LEVEL>
and reducing object size is good.
Coalesce multiline formats for easier grep.
Coalesce segmented single line formats too.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
Commit 4eeaaeaea (ALSA: core: add hooks for audio timestamps) added the
new audio_tstamp field to struct snd_pcm_status. However, struct
timespec requires 64-bit alignment, so the 64-bit compiler would insert
32 bits of padding before this field, which broke SNDRV_PCM_IOCTL_STATUS
with error messages like this:
kernel: unknown ioctl = 0x80984120
To solve this, insert the padding explicitly so that it can be taken
into account when calculating the ABI structure size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should really use "fck" when asking for the functional clock and not
"dmic_fck".
This way we can ensure that multiple dmic modules can exist in the system.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also drop the includes that are no longer needed and just
cause problems for the ARM common zImage.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
[tony@atomide.com: updated to drop unneeded headers]
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use regmap-mmio instead of open-coding caching and register accessors.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use devm_request_and_ioremap for requesting and mapping the IO region. This
makes the code a bit smaller and simpler.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add dB TLV ranges for the various volume controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The BIOS on HP dv5 doesn't have the DMI string to guide the setup of
mute led GPIO and polarity. Associate this laptop with the hp-inv-led
model.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: Vinícius Angiolucci <angiolucci@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A couple of driver fixes, one that improves the interoperability of
WM8994 with controllers that are sensitive to extra BCLK cycles and some
build break fixes for ux500.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
A couple of driver fixes, one that improves the interoperability of
WM8994 with controllers that are sensitive to extra BCLK cycles and some
build break fixes for ux500.
ARIZONA_MICB1_ENA_SHIFT was used for micbias 2 and 3. This change
correctly uses the ARIZONA_MICBX_ENA_SHIFT for each corresponding DAPM
supply. This should not have caused any problems as the micbias enables
are in the same place in each register.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The check of the return value from vortex_wtdma_bufshft() in
vortex_interrupt() is useless as it executes no code. Actually the
return value is intentionally ignored because the delta calculation
for wavetable doesn't work always correctly. For avoiding the
confusion, a comment is added and the superfluous if () is removed.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
window to remove most of the remaining plat includes to get us
closer to ARM common zImage support.
To avoid a huge amount of trivial merge conflicts with includes,
this branch is based on several small topic branches coordinated
with the driver subsystem maintainers. These branches are based on
v3.7-rc1 and can also be merged into the related driver subsystem
branches as needed:
omap-for-v3.8/cleanup-headers-prepare few trivial driver changes
omap-for-v3.8/cleanup-headers-dma move of the DMA header
omap-for-v3.8/cleanup-headers-gpmc GPMC and MTD changes
omap-for-v3.8/cleanup-headers-mmc MMC related changes
omap-for-v3.8/cleanup-headers-dss DSS related changes
omap-for-v3.8/cleanup-headers-asoc ASoC related changes
Note that for the dma-omap.h, it was decided that it should be
is completed. For the related discussion, please see:
https://patchwork.kernel.org/patch/1519591/#
After these patches we still have a few plat headers remaining
that will be handled in later pull requests.
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Merge tag 'omap-for-v3.8/cleanup-headers-signed' into omap-for-v3.8/cleanup-headers-serial-take2
This is the first set of omap cleanup patches for v3.8 merge
window to remove most of the remaining plat includes to get us
closer to ARM common zImage support.
To avoid a huge amount of trivial merge conflicts with includes,
this branch is based on several small topic branches coordinated
with the driver subsystem maintainers. These branches are based on
v3.7-rc1 and can also be merged into the related driver subsystem
branches as needed:
omap-for-v3.8/cleanup-headers-prepare few trivial driver changes
omap-for-v3.8/cleanup-headers-dma move of the DMA header
omap-for-v3.8/cleanup-headers-gpmc GPMC and MTD changes
omap-for-v3.8/cleanup-headers-mmc MMC related changes
omap-for-v3.8/cleanup-headers-dss DSS related changes
omap-for-v3.8/cleanup-headers-asoc ASoC related changes
Note that for the dma-omap.h, it was decided that it should be
is completed. For the related discussion, please see:
https://patchwork.kernel.org/patch/1519591/#
After these patches we still have a few plat headers remaining
that will be handled in later pull requests.
Rather than always assuming the maximum possible BCLK rate will be
required generate BCLKs for stereo if either one or two channels is
enabled. In order to support this we also need to ensure that only
the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When lowering SYSCLK to 50kHz for accessory detection also lower the
AIFnCLK divisor to normalise the clocking configuration within the
device. This will not disrupt audio as we cannot support active audio
with such a low SYSCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors by using correct kconfig symbol name:
sound/pci/ice1712/psc724.c:417:5: error: 'struct snd_ice1712' has no member named 'pm_resume'
sound/pci/ice1712/psc724.c:418:5: error: 'struct snd_ice1712' has no member named 'pm_suspend_enabled'
[Fixed another #ifdef CONFIG_PM in the same file, too, by tiwai]
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and
UAC_EP_CS_ATTR_SAMPLE_RATE attribute.
Signed-off-by: Didier Villevalois <ptitjes@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reuse code from clocksource to handle wall clock counter.
Since wrapparound occurs, the audio timestamp is reinitialized
to zero on a trigger. Synchronized linked devices will
start counting from same reference to avoid any drift.
Max buffer time is limited to 178 seconds to make sure
wall clock counter does not overflow
Wallclock timestamps are disabled on capture streams
until we figure out how to handle digital inputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This small reference boards has a Freescale P1022 dual-core PowerPC SOC
and a Wolfson Microelectronics WM8960 codec.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign each dai_link a unique name to avoid this run-time error.
[ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok
[ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97'
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removes the DaVinci private SRAM API and replaces it with
the genalloc API. The SRAM gen_pool is passed in pdata since
DaVinci is in the early stages of DT conversion.
[zonque@gmail.com: stub out gen_pool functions for
!CONFIG_GENERIC_ALLOCATOR]
Signed-off-by: Matt Porter <mporter@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable oldstatus is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix wm2200.c printk format warnings (seen on x86_64):
sound/soc/codecs/wm2200.c:1027:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm2200.c:1139:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int'
sound/soc/codecs/wm2200.c:1181:2: warning: format '%d' expects type 'int', but argument 7 has type 'size_t'
sound/soc/codecs/wm2200.c:1201:5: warning: format '%x' expects type 'unsigned int', but argument 3 has type 'long unsigned int'
sound/soc/codecs/wm2200.c:1264:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm2200.c:1328:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When switching to common clock driver for ux500 this clock needs to
be handled as well. Before this clock was internally managed by the
clock driver itself.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure clocks are being prepared and unprepared as well
as enabled and disabled.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable stream is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_miro_probe is a static function that is only called twice in the file
that defines it. At each call site, its argument is freed using
snd_card_free. Thus, there is no need for snd_miro_probe to call
snd_card_free on its argument on any of its error exit paths.
Because snd_card_free both reads the fields of its argument and kfrees its
argments, the results of the second snd_card_free should be unpredictable.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r@
identifier f,free,a;
parameter list[n] ps;
type T;
expression e;
@@
f(ps,T a,...) {
... when any
when != a = e
if(...) { ... free(a); ... return ...; }
... when any
}
@@
identifier r.f,r.free;
expression x,a;
expression list[r.n] xs;
@@
* x = f(xs,a,...);
if (...) { ... free(a); ... return ...; }
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this quirk the VG-99 will work in standard mode (set under
USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out
but no midi, and is not recognised when set to advanced mode.
After applying this, I can also use the VG-99 in advanced mode: 24
24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far
untested.
In standard mode, the device appears with ID 0x00b3, so the
behaviour isn't affected by this quirk.
Thanks to Clemens Ladisch for simplifying and correcting my initial
attempt!
Signed-off-by: Pete Leigh <pete.leigh@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are small races opened in the check of running bit and the timer
lock. Instead of adding yet more flag, just protect the whole racy
codes with the existing cable->lock. As a bonus, we can get rid of
timer_lock now.
Reported-and-tested-by: Omair Mohammed Abdullah <omair.m.abdullah@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable ep is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a few mixer applications do not handle deactivated controls
correctly. This patch adds such controls to snd-dummy to make
crash^H^H^H^H^Htesting these apps easier.
To make the testing of deactivated mixer controls easier (and for people
with common hardware, possible), add a control that deactivates some
other controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
By some reason, Toshiba laptop doesn't like the EAPD turned up for the
headphone pin. Add a fix up code to force to turn down EAPD for NID
0x15.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=569991
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
checkpatch.pl discourages the use of spaces at the beginning of lines.
Some of the CTL_ELEM defines were not properly indented.
This patch replaces the leading spaces by tabs. No functionality is
changed, the commit is purely cosmetic.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the documentation, AES32 cards use a different bit position
for reporting the sync_in status.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In contrast to AES32, MADI uses the first status register to report the
sync_in status.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MADI and MADIface used to report the autosync_sample_rate. This
functionality was lost in commit
0dca179306, this commit now adds it back.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Missing breaks lead to a fall-through, thus causing the wrong
autosync_sample_rate to be reported.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to missing breaks and the resulting fall-through, card subtype
selection was effectively missing, thus causing the wrong sync check
functions to be called.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a follow-up to a97bda7d29, report the
external sample rate as system_sample_rate when in slave mode.
For PCIe MADI cards, the DDS value automatically contains the external
sample rate, but the PCI version needs this manual workaround.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DDS value is the actual physical sample rate. We set it indirectly
when selecting 44100, 48000 and so on via snd_hdspm_hw_params or
hdspm_set_clock_source.
This commit now allows the DDS value to be altered at runtime, thus
speeding up or slowing down the physical sample rate. This is required
for MADI's varispeed that allows for ±12.5% speed adjustment from the
"selected" rate (32kHz, 44100kHz, 48kHz and so on).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking
station.
Without this patch, if I plug my headphones into the jack on the
computer, everything works fine. The computer speakers mute and the
audio is played in the headphones. However, if I plug into the docking
station headphone jack the computer speakers are muted but there is no
audio in the headphones.
Addresses https://bugs.launchpad.net/bugs/1060372
Signed-off-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include linux/io.h in ice1712.h since inb() and outb() are used in
inline functions there. Remove the redundant inclusion of that file
in other places at the same time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add chip details for E-mu 1010 PCIe card. It has the same
chip as found in E-mu 1010b but it uses different PCI id.
Signed-off-by: Maxim Kachur <mcdebugger@duganet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch DAPMises headphone and lineout output enable controls.
Earlier these output enable bits were permanently turned on in probe.
In da9055 codec, right outmixer is directly connected with right HP and
Line out. This resulted in two side effects,
(1) When you only want to use lineout, right HP (and connected charge
pump) also gets enabled
(2) When you only want to use stereo HP, lineout also gets enabled
This patch adds three switches to select which output(s) should be
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <david.chen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will greatly reduce the frequency for detection errors in
those cases where the hardware is "flaky", i e, rapidly changing
between plugged and unplugged states even without user interaction.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.
Even when CONFIG_SND_DEBUG is not enabled, we don't want to
return an arbitrary memory location when the channel count is
larger than we expected.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add psc724 subdriver to snd-ice1712 that provides full support for
Philips PSC724 Ultimate Edge sound cards.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Needed by Philips PSC724 subdriver. The code does not contain any
card-specific bits so other ice17xx cards using this codec could be
converted to use this generic code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Needed by Philips PSC724 subdriver. The code does not contain any
card-specific bits so other ice17xx cards using this codec could be
converted to use this generic code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add chip_exit callback to allow card subdrivers to do cleanup work on module
removal.
Needed by Philips PSC724 subdriver to cancel delayed work.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When both an SPDIF and an HDMI device are created on the same card
instance, multiple IEC958 controls are created with indices=0, 1, ...
But the alsa-lib configuration can't know which index corresponds
actually to which PCM device, and both the SPDIF and the HDMI
configurations point to the first IEC958 control wrongly.
This patch introduces a (hackish and ugly) workaround: the IEC958
controls for the SPDIF device are re-labeled with device=1 when HDMI
coexists. The device=1 corresponds to the actual PCM device for
SPDIF, so it's anyway a better representation. In future, HDMI
controls should be moved with the corresponding PCM device number,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If LPIB reports a pretty bad value, we can't trust such hardware for
calculating the PCM delay. Automatically turn off the delay counting
when such a problem is encountered.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=48911
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ux500_msp_i2s patches clashed with:
b18e93a493
ASoC: ux500_msp_i2s: better use devm functions and fix error return code
... leaving the driver uncompilable. This patch fixes the
issues encountered.
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When of_parse_phandle() is used to find a device node, its
reference count is incremented by the helper. Once we're
finished with them, it's our responsibly to ensure they
are freed in the correct manor.
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drivers should not use cpu_is_omap or cpu_class_is_omap macros,
they should be private to the platform init code. And we'll be
removing plat/cpu.h and only have a private soc.h for the
arch/arm/*omap* code.
This patch is intended as preparation for the core omap changes
and removes the need to include plat/cpu.h from several drivers.
This is needed for the ARM common zImage support.
These changes are OK to do because:
- omap-rng.c does not need plat/cpu.h
- omap-aes.c and omap-sham.c get the proper platform_data
passed to them so they don't need extra checks in the driver
- omap-dma.c and omap-pcm.c can test the arch locally as
omap1 and omap2 cannot be compiled together because of
conflicting compiler flags
Cc: Deepak Saxena <dsaxena@plexity.net>
Cc: Matt Mackall <mpm@selenic.com>
Cc: Herbert Xu <herbert@gondor.apana.org.au>
Cc: David S. Miller <davem@davemloft.net>
Cc: Venkatraman S <svenkatr@ti.com>
Cc: Chris Ball <cjb@laptop.org>
Cc: Vinod Koul <vinod.koul@intel.com>
Cc: Dan Williams <djbw@fb.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: linux-crypto@vger.kernel.org
Cc: linux-mmc@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
[tony@atomide.com: mmc changes folded in to an earlier patch]
Signed-off-by: Tony Lindgren <tony@atomide.com>
Now that we have a generic unsol mechanism, we can implement a generic
poll loop, which can be used for debugging, or if a codec's unsol
mechanism is broken.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec,
this patch modify patch_conexant.c to add Lemote specific code.
Both A1004 and A1205 use the same pin configurations, but A1004 need
to increase the default boost of internal mic.
Signed-off-by: Jie Chen <chenj@lemote.com>
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Signed-off-by: Hongliang Tao <taohl@lemote.com>
Signed-off-by: Hua Yan <yanh@lemote.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Delay the registration of VGA switcheroo client to the end of the
probing. Otherwise a too quick switching may result in Oops during
probing.
Also add the check of the return value from snd_hda_lock_devices().
Reported-and-tested-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The white-list entries of position_fix for ASUS laptops have been
added just as a workaround for broken COMBO mode. Now the combo mode
itself is disabled, we can safely remove these entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of updates to the regmap range support, the most important ones
being to rename "n_ranges" to "num_ranges" for consistency and to allow
block writes to cross page boundaries, making things more transparent.
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Merge tag 'regmap/range' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap into asoc-wm2200
regmap: Range API changes
A bunch of updates to the regmap range support, the most important ones
being to rename "n_ranges" to "num_ranges" for consistency and to allow
block writes to cross page boundaries, making things more transparent.
As of commit 99c2aa (firmware loader: fix creation failure of fw loader
device) we can have more than one firmware request outstanding at once so
there is no need to daisychain our requests any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This can be used to provide some additional settling time to ensure that
we don't start microphone detection while the microphone pin is connected
to one of the headphone pins.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is more idiomatic and ensures we don't try to do the ASoC card setup
until we've got all the required resources.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the device is now idle_bias_off these never have any useful effect,
the device will be brought to _OFF when idle, and will at best leave it
powered for longer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This commit add a sound codec driver for Silicon Laboratories 476x
series of AM/FM radio chips.
Signed-off-by: Andrey Smirnov <andrey.smirnov@convergeddevices.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all CODEC devices have three audio interfaces and the clock rates
which support these things vary. Support this by using driver data to
supply the clock rates and by only completing the parts of system setup
which are required for the system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Bells system can take a WM0010 as well as a CODEC - assume there's
one present by default. While we're at it stop using magic numbers for
the DAIs for readability.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare
calls as required by common clock framework.
Signed-off-by: Thomas Abraham <thomas.abraham@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare
calls as required by common clock framework.
Signed-off-by: Thomas Abraham <thomas.abraham@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare
calls as required by common clock framework.
Signed-off-by: Thomas Abraham <thomas.abraham@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare
calls as required by common clock framework.
Signed-off-by: Thomas Abraham <thomas.abraham@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses devm_request_irq() instead of request_irq()
and removed free_irq() from driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSI can support 1 - 8 channel input/output,
but current driver is supporting 2 channel format only.
This patch fixes channel mismatch
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Mike Looijmans <mike.looijmans@topic.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The FSDUR flag configures whether the frame clock uses a high phase of
only one bit or a full word. This has to be set depending on the DAI
format.
For other modes than DSP_B, the FSXDLY/FSRDLY fields have to be set to
1.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sachin Kamat
Daniel Mack
Olof Johansson
Takashi Iwai
Tony Lindgren
Mark Brown
David Henningsson
Eldad Zack
Padmavathi Venna
Peter Ujfalusi
Lothar Waßmann
Tushar Behera
Lee Jones
Scott Ling
Dimitris Papastamos
Kuninori Morimoto
Liam Girdwood
Russell King
Bill Pemberton
Dylan Reid
Adam Buchbinder
Masanari Iida
Clemens Ladisch
Dan Carpenter
Joe Perches
Bo Shen
Eric Millbrandt
Arnd Bergmann
Xi Wang
Charles Keepax
Martin Schwenke
Ondrej Zary
Mukund Navada
Misael Lopez Cruz
Kailang Yang
Adrian Knoth
Wei Yongjun
Scott Ling
Stephen Warren
Sergiu Giurgiu
Daniel J Blueman
Alexander Stein
Linus Walleij
Lars R. Damerow
Laurence Darby
Fabio Estevam
Javier Martin
Lars-Peter Clausen
Gustavo Maciel Dias Vieira
Kees Cook
Randy Dunlap
Didier Villevalois
Pierre-Louis Bossart
Timur Tabi
Denis Kirjanov
Matt Porter
Ulf Hansson
Julia Lawall
Pete Leigh
Stefán Freyr
Maxim Kachur
Ashish Chavan
Huacai Chen
Andrey Smirnov
Thomas Abraham